import FIREFOX_52_6_0esr_RELEASE from mozilla-esr52 hg repo

dom/media/gtest/TestAudioPacketizer.cpp

@@ -0,0 +1,167 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* 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 <stdint.h>
+#include <math.h>
+#include "../AudioPacketizer.h"
+#include "gtest/gtest.h"
+
+using namespace mozilla;
+
+template<typename T>
+class AutoBuffer
+{
+public:
+  explicit AutoBuffer(size_t aLength)
+  {
+    mStorage = new T[aLength];
+  }
+  ~AutoBuffer() {
+    delete [] mStorage;
+  }
+  T* Get() {
+    return mStorage;
+  }
+private:
+  T* mStorage;
+};
+
+int16_t Sequence(int16_t* aBuffer, uint32_t aSize, uint32_t aStart = 0)
+{
+  uint32_t i;
+  for (i = 0; i < aSize; i++) {
+    aBuffer[i] = aStart + i;
+  }
+  return aStart + i;
+}
+
+void IsSequence(int16_t* aBuffer, uint32_t aSize, uint32_t aStart = 0)
+{
+  for (uint32_t i = 0; i < aSize; i++) {
+    ASSERT_TRUE(aBuffer[i] == static_cast<int64_t>(aStart + i)) <<
+      "Buffer is not a sequence at offset " << i << std::endl;
+  }
+  // Buffer is a sequence.
+}
+
+void Zero(int16_t* aBuffer, uint32_t aSize)
+{
+  for (uint32_t i = 0; i < aSize; i++) {
+    ASSERT_TRUE(aBuffer[i] == 0) <<
+      "Buffer is not null at offset " << i << std::endl;
+  }
+}
+
+double sine(uint32_t aPhase) {
+  return sin(aPhase * 2 * M_PI * 440 / 44100);
+}
+
+TEST(AudioPacketizer, Test)
+{
+  for (int16_t channels = 1; channels < 2; channels++) {
+    // Test that the packetizer returns zero on underrun
+    {
+      AudioPacketizer<int16_t, int16_t> ap(441, channels);
+      for (int16_t i = 0; i < 10; i++) {
+        int16_t* out = ap.Output();
+        Zero(out, 441);
+        delete[] out;
+      }
+    }
+    // Simple test, with input/output buffer size aligned on the packet size,
+    // alternating Input and Output calls.
+    {
+      AudioPacketizer<int16_t, int16_t> ap(441, channels);
+      int16_t seqEnd = 0;
+      for (int16_t i = 0; i < 10; i++) {
+        AutoBuffer<int16_t> b(441 * channels);
+        int16_t prevEnd = seqEnd;
+        seqEnd = Sequence(b.Get(), channels * 441, prevEnd);
+        ap.Input(b.Get(), 441);
+        int16_t* out = ap.Output();
+        IsSequence(out, 441 * channels, prevEnd);
+        delete[] out;
+      }
+    }
+    // Simple test, with input/output buffer size aligned on the packet size,
+    // alternating two Input and Output calls.
+    {
+      AudioPacketizer<int16_t, int16_t> ap(441, channels);
+      int16_t seqEnd = 0;
+      for (int16_t i = 0; i < 10; i++) {
+        AutoBuffer<int16_t> b(441 * channels);
+        AutoBuffer<int16_t> b1(441 * channels);
+        int16_t prevEnd0 = seqEnd;
+        seqEnd = Sequence(b.Get(), 441 * channels, prevEnd0);
+        int16_t prevEnd1 = seqEnd;
+        seqEnd = Sequence(b1.Get(), 441 * channels, seqEnd);
+        ap.Input(b.Get(), 441);
+        ap.Input(b1.Get(), 441);
+        int16_t* out = ap.Output();
+        int16_t* out2 = ap.Output();
+        IsSequence(out, 441 * channels, prevEnd0);
+        IsSequence(out2, 441 * channels, prevEnd1);
+        delete[] out;
+        delete[] out2;
+      }
+    }
+    // Input/output buffer size not aligned on the packet size,
+    // alternating two Input and Output calls.
+    {
+      AudioPacketizer<int16_t, int16_t> ap(441, channels);
+      int16_t prevEnd = 0;
+      int16_t prevSeq = 0;
+      for (int16_t i = 0; i < 10; i++) {
+        AutoBuffer<int16_t> b(480 * channels);
+        AutoBuffer<int16_t> b1(480 * channels);
+        prevSeq = Sequence(b.Get(), 480 * channels, prevSeq);
+        prevSeq = Sequence(b1.Get(), 480 * channels, prevSeq);
+        ap.Input(b.Get(), 480);
+        ap.Input(b1.Get(), 480);
+        int16_t* out = ap.Output();
+        int16_t* out2 = ap.Output();
+        IsSequence(out, 441 * channels, prevEnd);
+        prevEnd += 441 * channels;
+        IsSequence(out2, 441 * channels, prevEnd);
+        prevEnd += 441 * channels;
+        delete[] out;
+        delete[] out2;
+      }
+      printf("Available: %d\n", ap.PacketsAvailable());
+    }
+
+    // "Real-life" test case: streaming a sine wave through a packetizer, and
+    // checking that we have the right output.
+    // 128 is, for example, the size of a Web Audio API block, and 441 is the
+    // size of a webrtc.org packet when the sample rate is 44100 (10ms)
+    {
+      AudioPacketizer<int16_t, int16_t> ap(441, channels);
+      AutoBuffer<int16_t> b(128 * channels);
+      uint32_t phase = 0;
+      uint32_t outPhase = 0;
+      for (int16_t i = 0; i < 1000; i++) {
+        for (int32_t j = 0; j < 128; j++) {
+          for (int32_t c = 0; c < channels; c++) {
+            // int16_t sinewave at 440Hz/44100Hz sample rate
+            b.Get()[j * channels + c] = (2 << 14) * sine(phase);
+          }
+          phase++;
+        }
+        ap.Input(b.Get(), 128);
+        while (ap.PacketsAvailable()) {
+          int16_t* packet = ap.Output();
+          for (uint32_t k = 0; k < ap.PacketSize(); k++) {
+            for (int32_t c = 0; c < channels; c++) {
+              ASSERT_TRUE(packet[k * channels + c] ==
+                          static_cast<int16_t>(((2 << 14) * sine(outPhase))));
+            }
+            outPhase++;
+          }
+          delete [] packet;
+        }
+      }
+    }
+  }
+}