import FIREFOX_52_6_0esr_RELEASE from mozilla-esr52 hg repo

dom/media/AudioStream.cpp

@@ -0,0 +1,707 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <stdio.h>
+#include <math.h>
+#include <string.h>
+#include "mozilla/Logging.h"
+#include "prdtoa.h"
+#include "AudioStream.h"
+#include "VideoUtils.h"
+#include "mozilla/Monitor.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/Sprintf.h"
+#include <algorithm>
+#include "mozilla/Telemetry.h"
+#include "CubebUtils.h"
+#include "nsPrintfCString.h"
+#include "gfxPrefs.h"
+#include "AudioConverter.h"
+
+namespace mozilla {
+
+#undef LOG
+#undef LOGW
+
+LazyLogModule gAudioStreamLog("AudioStream");
+// For simple logs
+#define LOG(x, ...) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Debug, ("%p " x, this, ##__VA_ARGS__))
+#define LOGW(x, ...) MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Warning, ("%p " x, this, ##__VA_ARGS__))
+
+/**
+ * Keep a list of frames sent to the audio engine in each DataCallback along
+ * with the playback rate at the moment. Since the playback rate and number of
+ * underrun frames can vary in each callback. We need to keep the whole history
+ * in order to calculate the playback position of the audio engine correctly.
+ */
+class FrameHistory {
+  struct Chunk {
+    uint32_t servicedFrames;
+    uint32_t totalFrames;
+    uint32_t rate;
+  };
+
+  template <typename T>
+  static T FramesToUs(uint32_t frames, int rate) {
+    return static_cast<T>(frames) * USECS_PER_S / rate;
+  }
+public:
+  FrameHistory()
+    : mBaseOffset(0), mBasePosition(0) {}
+
+  void Append(uint32_t aServiced, uint32_t aUnderrun, uint32_t aRate) {
+    /* In most case where playback rate stays the same and we don't underrun
+     * frames, we are able to merge chunks to avoid lose of precision to add up
+     * in compressing chunks into |mBaseOffset| and |mBasePosition|.
+     */
+    if (!mChunks.IsEmpty()) {
+      Chunk& c = mChunks.LastElement();
+      // 2 chunks (c1 and c2) can be merged when rate is the same and
+      // adjacent frames are zero. That is, underrun frames in c1 are zero
+      // or serviced frames in c2 are zero.
+      if (c.rate == aRate &&
+          (c.servicedFrames == c.totalFrames ||
+           aServiced == 0)) {
+        c.servicedFrames += aServiced;
+        c.totalFrames += aServiced + aUnderrun;
+        return;
+      }
+    }
+    Chunk* p = mChunks.AppendElement();
+    p->servicedFrames = aServiced;
+    p->totalFrames = aServiced + aUnderrun;
+    p->rate = aRate;
+  }
+
+  /**
+   * @param frames The playback position in frames of the audio engine.
+   * @return The playback position in microseconds of the audio engine,
+   *         adjusted by playback rate changes and underrun frames.
+   */
+  int64_t GetPosition(int64_t frames) {
+    // playback position should not go backward.
+    MOZ_ASSERT(frames >= mBaseOffset);
+    while (true) {
+      if (mChunks.IsEmpty()) {
+        return mBasePosition;
+      }
+      const Chunk& c = mChunks[0];
+      if (frames <= mBaseOffset + c.totalFrames) {
+        uint32_t delta = frames - mBaseOffset;
+        delta = std::min(delta, c.servicedFrames);
+        return static_cast<int64_t>(mBasePosition) +
+               FramesToUs<int64_t>(delta, c.rate);
+      }
+      // Since the playback position of the audio engine will not go backward,
+      // we are able to compress chunks so that |mChunks| won't grow unlimitedly.
+      // Note that we lose precision in converting integers into floats and
+      // inaccuracy will accumulate over time. However, for a 24hr long,
+      // sample rate = 44.1k file, the error will be less than 1 microsecond
+      // after playing 24 hours. So we are fine with that.
+      mBaseOffset += c.totalFrames;
+      mBasePosition += FramesToUs<double>(c.servicedFrames, c.rate);
+      mChunks.RemoveElementAt(0);
+    }
+  }
+private:
+  AutoTArray<Chunk, 7> mChunks;
+  int64_t mBaseOffset;
+  double mBasePosition;
+};
+
+AudioStream::AudioStream(DataSource& aSource)
+  : mMonitor("AudioStream")
+  , mChannels(0)
+  , mOutChannels(0)
+  , mTimeStretcher(nullptr)
+  , mDumpFile(nullptr)
+  , mState(INITIALIZED)
+  , mDataSource(aSource)
+{
+}
+
+AudioStream::~AudioStream()
+{
+  LOG("deleted, state %d", mState);
+  MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
+             "Should've called Shutdown() before deleting an AudioStream");
+  if (mDumpFile) {
+    fclose(mDumpFile);
+  }
+  if (mTimeStretcher) {
+    soundtouch::destroySoundTouchObj(mTimeStretcher);
+  }
+}
+
+size_t
+AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = aMallocSizeOf(this);
+
+  // Possibly add in the future:
+  // - mTimeStretcher
+  // - mCubebStream
+
+  return amount;
+}
+
+nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
+{
+  mMonitor.AssertCurrentThreadOwns();
+  if (!mTimeStretcher) {
+    mTimeStretcher = soundtouch::createSoundTouchObj();
+    mTimeStretcher->setSampleRate(mAudioClock.GetInputRate());
+    mTimeStretcher->setChannels(mOutChannels);
+    mTimeStretcher->setPitch(1.0);
+  }
+  return NS_OK;
+}
+
+nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
+{
+  // MUST lock since the rate transposer is used from the cubeb callback,
+  // and rate changes can cause the buffer to be reallocated
+  MonitorAutoLock mon(mMonitor);
+
+  NS_ASSERTION(aPlaybackRate > 0.0,
+               "Can't handle negative or null playbackrate in the AudioStream.");
+  // Avoid instantiating the resampler if we are not changing the playback rate.
+  // GetPreservesPitch/SetPreservesPitch don't need locking before calling
+  if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
+    return NS_OK;
+  }
+
+  if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
+    return NS_ERROR_FAILURE;
+  }
+
+  mAudioClock.SetPlaybackRate(aPlaybackRate);
+
+  if (mAudioClock.GetPreservesPitch()) {
+    mTimeStretcher->setTempo(aPlaybackRate);
+    mTimeStretcher->setRate(1.0f);
+  } else {
+    mTimeStretcher->setTempo(1.0f);
+    mTimeStretcher->setRate(aPlaybackRate);
+  }
+  return NS_OK;
+}
+
+nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch)
+{
+  // MUST lock since the rate transposer is used from the cubeb callback,
+  // and rate changes can cause the buffer to be reallocated
+  MonitorAutoLock mon(mMonitor);
+
+  // Avoid instantiating the timestretcher instance if not needed.
+  if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
+    return NS_OK;
+  }
+
+  if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
+    return NS_ERROR_FAILURE;
+  }
+
+  if (aPreservesPitch == true) {
+    mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
+    mTimeStretcher->setRate(1.0f);
+  } else {
+    mTimeStretcher->setTempo(1.0f);
+    mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
+  }
+
+  mAudioClock.SetPreservesPitch(aPreservesPitch);
+
+  return NS_OK;
+}
+
+static void SetUint16LE(uint8_t* aDest, uint16_t aValue)
+{
+  aDest[0] = aValue & 0xFF;
+  aDest[1] = aValue >> 8;
+}
+
+static void SetUint32LE(uint8_t* aDest, uint32_t aValue)
+{
+  SetUint16LE(aDest, aValue & 0xFFFF);
+  SetUint16LE(aDest + 2, aValue >> 16);
+}
+
+static FILE*
+OpenDumpFile(uint32_t aChannels, uint32_t aRate)
+{
+  /**
+   * When MOZ_DUMP_AUDIO is set in the environment (to anything),
+   * we'll drop a series of files in the current working directory named
+   * dumped-audio-<nnn>.wav, one per AudioStream created, containing
+   * the audio for the stream including any skips due to underruns.
+   */
+  static Atomic<int> gDumpedAudioCount(0);
+
+  if (!getenv("MOZ_DUMP_AUDIO"))
+    return nullptr;
+  char buf[100];
+  SprintfLiteral(buf, "dumped-audio-%d.wav", ++gDumpedAudioCount);
+  FILE* f = fopen(buf, "wb");
+  if (!f)
+    return nullptr;
+
+  uint8_t header[] = {
+    // RIFF header
+    0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
+    // fmt chunk. We always write 16-bit samples.
+    0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
+    0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
+    // data chunk
+    0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
+  };
+  static const int CHANNEL_OFFSET = 22;
+  static const int SAMPLE_RATE_OFFSET = 24;
+  static const int BLOCK_ALIGN_OFFSET = 32;
+  SetUint16LE(header + CHANNEL_OFFSET, aChannels);
+  SetUint32LE(header + SAMPLE_RATE_OFFSET, aRate);
+  SetUint16LE(header + BLOCK_ALIGN_OFFSET, aChannels * 2);
+  fwrite(header, sizeof(header), 1, f);
+
+  return f;
+}
+
+template <typename T>
+typename EnableIf<IsSame<T, int16_t>::value, void>::Type
+WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
+  fwrite(aInput, sizeof(T), aSamples, aFile);
+}
+
+template <typename T>
+typename EnableIf<IsSame<T, float>::value, void>::Type
+WriteDumpFileHelper(T* aInput, size_t aSamples, FILE* aFile) {
+  AutoTArray<uint8_t, 1024*2> buf;
+  buf.SetLength(aSamples*2);
+  uint8_t* output = buf.Elements();
+  for (uint32_t i = 0; i < aSamples; ++i) {
+    SetUint16LE(output + i*2, int16_t(aInput[i]*32767.0f));
+  }
+  fwrite(output, 2, aSamples, aFile);
+  fflush(aFile);
+}
+
+static void
+WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
+              void* aBuffer)
+{
+  if (!aDumpFile)
+    return;
+
+  uint32_t samples = aStream->GetOutChannels()*aFrames;
+
+  using SampleT = AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type;
+  WriteDumpFileHelper(reinterpret_cast<SampleT*>(aBuffer), samples, aDumpFile);
+}
+
+template <AudioSampleFormat N>
+struct ToCubebFormat {
+  static const cubeb_sample_format value = CUBEB_SAMPLE_FLOAT32NE;
+};
+
+template <>
+struct ToCubebFormat<AUDIO_FORMAT_S16> {
+  static const cubeb_sample_format value = CUBEB_SAMPLE_S16NE;
+};
+
+template <typename Function, typename... Args>
+int AudioStream::InvokeCubeb(Function aFunction, Args&&... aArgs)
+{
+  MonitorAutoUnlock mon(mMonitor);
+  return aFunction(mCubebStream.get(), Forward<Args>(aArgs)...);
+}
+
+nsresult
+AudioStream::Init(uint32_t aNumChannels, uint32_t aRate,
+                  const dom::AudioChannel aAudioChannel)
+{
+  auto startTime = TimeStamp::Now();
+
+  LOG("%s channels: %d, rate: %d", __FUNCTION__, aNumChannels, aRate);
+  mChannels = aNumChannels;
+  mOutChannels = aNumChannels;
+
+  mDumpFile = OpenDumpFile(aNumChannels, aRate);
+
+  cubeb_stream_params params;
+  params.rate = aRate;
+  params.channels = mOutChannels;
+#if defined(__ANDROID__)
+#if defined(MOZ_B2G)
+  params.stream_type = CubebUtils::ConvertChannelToCubebType(aAudioChannel);
+#else
+  params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
+#endif
+
+  if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
+    return NS_ERROR_INVALID_ARG;
+  }
+#endif
+
+  params.format = ToCubebFormat<AUDIO_OUTPUT_FORMAT>::value;
+  mAudioClock.Init(aRate);
+
+  cubeb* cubebContext = CubebUtils::GetCubebContext();
+  if (!cubebContext) {
+    NS_WARNING("Can't get cubeb context!");
+    CubebUtils::ReportCubebStreamInitFailure(true);
+    return NS_ERROR_DOM_MEDIA_CUBEB_INITIALIZATION_ERR;
+  }
+
+  return OpenCubeb(cubebContext, params, startTime, CubebUtils::GetFirstStream());
+}
+
+nsresult
+AudioStream::OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
+                       TimeStamp aStartTime, bool aIsFirst)
+{
+  MOZ_ASSERT(aContext);
+
+  cubeb_stream* stream = nullptr;
+  /* Convert from milliseconds to frames. */
+  uint32_t latency_frames =
+    CubebUtils::GetCubebPlaybackLatencyInMilliseconds() * aParams.rate / 1000;
+  if (cubeb_stream_init(aContext, &stream, "AudioStream",
+                        nullptr, nullptr, nullptr, &aParams,
+                        latency_frames,
+                        DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
+    mCubebStream.reset(stream);
+    CubebUtils::ReportCubebBackendUsed();
+  } else {
+    NS_WARNING(nsPrintfCString("AudioStream::OpenCubeb() %p failed to init cubeb", this).get());
+    CubebUtils::ReportCubebStreamInitFailure(aIsFirst);
+    return NS_ERROR_FAILURE;
+  }
+
+  TimeDuration timeDelta = TimeStamp::Now() - aStartTime;
+  LOG("creation time %sfirst: %u ms", aIsFirst ? "" : "not ",
+      (uint32_t) timeDelta.ToMilliseconds());
+  Telemetry::Accumulate(aIsFirst ? Telemetry::AUDIOSTREAM_FIRST_OPEN_MS :
+      Telemetry::AUDIOSTREAM_LATER_OPEN_MS, timeDelta.ToMilliseconds());
+
+  return NS_OK;
+}
+
+void
+AudioStream::SetVolume(double aVolume)
+{
+  MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
+
+  if (cubeb_stream_set_volume(mCubebStream.get(), aVolume * CubebUtils::GetVolumeScale()) != CUBEB_OK) {
+    NS_WARNING("Could not change volume on cubeb stream.");
+  }
+}
+
+void
+AudioStream::Start()
+{
+  MonitorAutoLock mon(mMonitor);
+  MOZ_ASSERT(mState == INITIALIZED);
+  mState = STARTED;
+  auto r = InvokeCubeb(cubeb_stream_start);
+  if (r != CUBEB_OK) {
+    mState = ERRORED;
+  }
+  LOG("started, state %s", mState == STARTED ? "STARTED" : mState == DRAINED ? "DRAINED" : "ERRORED");
+}
+
+void
+AudioStream::Pause()
+{
+  MonitorAutoLock mon(mMonitor);
+  MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
+  MOZ_ASSERT(mState != STOPPED, "Already Pause()ed.");
+  MOZ_ASSERT(mState != SHUTDOWN, "Already Shutdown()ed.");
+
+  // Do nothing if we are already drained or errored.
+  if (mState == DRAINED || mState == ERRORED) {
+    return;
+  }
+
+  if (InvokeCubeb(cubeb_stream_stop) != CUBEB_OK) {
+    mState = ERRORED;
+  } else if (mState != DRAINED && mState != ERRORED) {
+    // Don't transition to other states if we are already
+    // drained or errored.
+    mState = STOPPED;
+  }
+}
+
+void
+AudioStream::Resume()
+{
+  MonitorAutoLock mon(mMonitor);
+  MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
+  MOZ_ASSERT(mState != STARTED, "Already Start()ed.");
+  MOZ_ASSERT(mState != SHUTDOWN, "Already Shutdown()ed.");
+
+  // Do nothing if we are already drained or errored.
+  if (mState == DRAINED || mState == ERRORED) {
+    return;
+  }
+
+  if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
+    mState = ERRORED;
+  } else if (mState != DRAINED && mState != ERRORED) {
+    // Don't transition to other states if we are already
+    // drained or errored.
+    mState = STARTED;
+  }
+}
+
+void
+AudioStream::Shutdown()
+{
+  MonitorAutoLock mon(mMonitor);
+  LOG("Shutdown, state %d", mState);
+
+  if (mCubebStream) {
+    MonitorAutoUnlock mon(mMonitor);
+    // Force stop to put the cubeb stream in a stable state before deletion.
+    cubeb_stream_stop(mCubebStream.get());
+    // Must not try to shut down cubeb from within the lock!  wasapi may still
+    // call our callback after Pause()/stop()!?! Bug 996162
+    mCubebStream.reset();
+  }
+
+  mState = SHUTDOWN;
+}
+
+int64_t
+AudioStream::GetPosition()
+{
+  MonitorAutoLock mon(mMonitor);
+  int64_t frames = GetPositionInFramesUnlocked();
+  return frames >= 0 ? mAudioClock.GetPosition(frames) : -1;
+}
+
+int64_t
+AudioStream::GetPositionInFrames()
+{
+  MonitorAutoLock mon(mMonitor);
+  int64_t frames = GetPositionInFramesUnlocked();
+  return frames >= 0 ? mAudioClock.GetPositionInFrames(frames) : -1;
+}
+
+int64_t
+AudioStream::GetPositionInFramesUnlocked()
+{
+  mMonitor.AssertCurrentThreadOwns();
+
+  if (mState == ERRORED) {
+    return -1;
+  }
+
+  uint64_t position = 0;
+  if (InvokeCubeb(cubeb_stream_get_position, &position) != CUBEB_OK) {
+    return -1;
+  }
+  return std::min<uint64_t>(position, INT64_MAX);
+}
+
+bool
+AudioStream::IsValidAudioFormat(Chunk* aChunk)
+{
+  if (aChunk->Rate() != mAudioClock.GetInputRate()) {
+    LOGW("mismatched sample %u, mInRate=%u", aChunk->Rate(), mAudioClock.GetInputRate());
+    return false;
+  }
+
+  if (aChunk->Channels() > 8) {
+    return false;
+  }
+
+  return true;
+}
+
+void
+AudioStream::GetUnprocessed(AudioBufferWriter& aWriter)
+{
+  mMonitor.AssertCurrentThreadOwns();
+
+  // Flush the timestretcher pipeline, if we were playing using a playback rate
+  // other than 1.0.
+  if (mTimeStretcher && mTimeStretcher->numSamples()) {
+    auto timeStretcher = mTimeStretcher;
+    aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
+      return timeStretcher->receiveSamples(aPtr, aFrames);
+    }, aWriter.Available());
+
+    // TODO: There might be still unprocessed samples in the stretcher.
+    // We should either remove or flush them so they won't be in the output
+    // next time we switch a playback rate other than 1.0.
+    NS_WARNING_ASSERTION(
+      mTimeStretcher->numUnprocessedSamples() == 0, "no samples");
+  }
+
+  while (aWriter.Available() > 0) {
+    UniquePtr<Chunk> c = mDataSource.PopFrames(aWriter.Available());
+    if (c->Frames() == 0) {
+      break;
+    }
+    MOZ_ASSERT(c->Frames() <= aWriter.Available());
+    if (IsValidAudioFormat(c.get())) {
+      aWriter.Write(c->Data(), c->Frames());
+    } else {
+      // Write silence if invalid format.
+      aWriter.WriteZeros(c->Frames());
+    }
+  }
+}
+
+void
+AudioStream::GetTimeStretched(AudioBufferWriter& aWriter)
+{
+  mMonitor.AssertCurrentThreadOwns();
+
+  // We need to call the non-locking version, because we already have the lock.
+  if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
+    return;
+  }
+
+  uint32_t toPopFrames =
+    ceil(aWriter.Available() * mAudioClock.GetPlaybackRate());
+
+  while (mTimeStretcher->numSamples() < aWriter.Available()) {
+    UniquePtr<Chunk> c = mDataSource.PopFrames(toPopFrames);
+    if (c->Frames() == 0) {
+      break;
+    }
+    MOZ_ASSERT(c->Frames() <= toPopFrames);
+    if (IsValidAudioFormat(c.get())) {
+      mTimeStretcher->putSamples(c->Data(), c->Frames());
+    } else {
+      // Write silence if invalid format.
+      AutoTArray<AudioDataValue, 1000> buf;
+      buf.SetLength(mOutChannels * c->Frames());
+      memset(buf.Elements(), 0, buf.Length() * sizeof(AudioDataValue));
+      mTimeStretcher->putSamples(buf.Elements(), c->Frames());
+    }
+  }
+
+  auto timeStretcher = mTimeStretcher;
+  aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
+    return timeStretcher->receiveSamples(aPtr, aFrames);
+  }, aWriter.Available());
+}
+
+long
+AudioStream::DataCallback(void* aBuffer, long aFrames)
+{
+  MonitorAutoLock mon(mMonitor);
+  MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
+
+  auto writer = AudioBufferWriter(
+    reinterpret_cast<AudioDataValue*>(aBuffer), mOutChannels, aFrames);
+
+  if (!strcmp(cubeb_get_backend_id(CubebUtils::GetCubebContext()), "winmm")) {
+    // Don't consume audio data until Start() is called.
+    // Expected only with cubeb winmm backend.
+    if (mState == INITIALIZED) {
+      NS_WARNING("data callback fires before cubeb_stream_start() is called");
+      mAudioClock.UpdateFrameHistory(0, aFrames);
+      return writer.WriteZeros(aFrames);
+    }
+  } else {
+    MOZ_ASSERT(mState != INITIALIZED);
+  }
+
+  // NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
+  // Bug 996162
+
+  if (mAudioClock.GetInputRate() == mAudioClock.GetOutputRate()) {
+    GetUnprocessed(writer);
+  } else {
+    GetTimeStretched(writer);
+  }
+
+  // Always send audible frames first, and silent frames later.
+  // Otherwise it will break the assumption of FrameHistory.
+  if (!mDataSource.Ended()) {
+    mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), writer.Available());
+    if (writer.Available() > 0) {
+      LOGW("lost %d frames", writer.Available());
+      writer.WriteZeros(writer.Available());
+    }
+  } else {
+    // No more new data in the data source. Don't send silent frames so the
+    // cubeb stream can start draining.
+    mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), 0);
+  }
+
+  WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
+
+  return aFrames - writer.Available();
+}
+
+void
+AudioStream::StateCallback(cubeb_state aState)
+{
+  MonitorAutoLock mon(mMonitor);
+  MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
+  LOG("StateCallback, mState=%d cubeb_state=%d", mState, aState);
+  if (aState == CUBEB_STATE_DRAINED) {
+    mState = DRAINED;
+    mDataSource.Drained();
+  } else if (aState == CUBEB_STATE_ERROR) {
+    LOG("StateCallback() state %d cubeb error", mState);
+    mState = ERRORED;
+  }
+}
+
+AudioClock::AudioClock()
+: mOutRate(0),
+  mInRate(0),
+  mPreservesPitch(true),
+  mFrameHistory(new FrameHistory())
+{}
+
+void AudioClock::Init(uint32_t aRate)
+{
+  mOutRate = aRate;
+  mInRate = aRate;
+}
+
+void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun)
+{
+  mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
+}
+
+int64_t AudioClock::GetPositionInFrames(int64_t aFrames) const
+{
+  CheckedInt64 v = UsecsToFrames(GetPosition(aFrames), mInRate);
+  return v.isValid() ? v.value() : -1;
+}
+
+int64_t AudioClock::GetPosition(int64_t frames) const
+{
+  return mFrameHistory->GetPosition(frames);
+}
+
+void AudioClock::SetPlaybackRate(double aPlaybackRate)
+{
+  mOutRate = static_cast<uint32_t>(mInRate / aPlaybackRate);
+}
+
+double AudioClock::GetPlaybackRate() const
+{
+  return static_cast<double>(mInRate) / mOutRate;
+}
+
+void AudioClock::SetPreservesPitch(bool aPreservesPitch)
+{
+  mPreservesPitch = aPreservesPitch;
+}
+
+bool AudioClock::GetPreservesPitch() const
+{
+  return mPreservesPitch;
+}
+
+} // namespace mozilla