1 files changed, 470 insertions, 0 deletions
diff --git a/dom/media/gtest/TestAudioSegment.cpp b/dom/media/gtest/TestAudioSegment.cpp
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+++ b/dom/media/gtest/TestAudioSegment.cpp
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+/* -*- 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 "AudioSegment.h"
+#include <iostream>
+#include "gtest/gtest.h"
+
+#include "AudioGenerator.h"
+
+using namespace mozilla;
+
+namespace audio_segment {
+
+/* Helper function to give us the maximum and minimum value that don't clip,
+ * for a given sample format (integer or floating-point). */
+template <typename T>
+T GetLowValue();
+
+template <typename T>
+T GetHighValue();
+
+template <typename T>
+T GetSilentValue();
+
+template <>
+float GetLowValue<float>() {
+  return -1.0;
+}
+
+template <>
+int16_t GetLowValue<short>() {
+  return -INT16_MAX;
+}
+
+template <>
+float GetHighValue<float>() {
+  return 1.0;
+}
+
+template <>
+int16_t GetHighValue<short>() {
+  return INT16_MAX;
+}
+
+template <>
+float GetSilentValue() {
+  return 0.0;
+}
+
+template <>
+int16_t GetSilentValue() {
+  return 0;
+}
+
+// Get an array of planar audio buffers that has the inverse of the index of the
+// channel (1-indexed) as samples.
+template <typename T>
+const T* const* GetPlanarChannelArray(size_t aChannels, size_t aSize) {
+  T** channels = new T*[aChannels];
+  for (size_t c = 0; c < aChannels; c++) {
+    channels[c] = new T[aSize];
+    for (size_t i = 0; i < aSize; i++) {
+      channels[c][i] = FloatToAudioSample<T>(1. / (c + 1));
+    }
+  }
+  return channels;
+}
+
+template <typename T>
+void DeletePlanarChannelsArray(const T* const* aArrays, size_t aChannels) {
+  for (size_t channel = 0; channel < aChannels; channel++) {
+    delete[] aArrays[channel];
+  }
+  delete[] aArrays;
+}
+
+template <typename T>
+T** GetPlanarArray(size_t aChannels, size_t aSize) {
+  T** channels = new T*[aChannels];
+  for (size_t c = 0; c < aChannels; c++) {
+    channels[c] = new T[aSize];
+    for (size_t i = 0; i < aSize; i++) {
+      channels[c][i] = 0.0f;
+    }
+  }
+  return channels;
+}
+
+template <typename T>
+void DeletePlanarArray(T** aArrays, size_t aChannels) {
+  for (size_t channel = 0; channel < aChannels; channel++) {
+    delete[] aArrays[channel];
+  }
+  delete[] aArrays;
+}
+
+// Get an array of audio samples that have the inverse of the index of the
+// channel (1-indexed) as samples.
+template <typename T>
+const T* GetInterleavedChannelArray(size_t aChannels, size_t aSize) {
+  size_t sampleCount = aChannels * aSize;
+  T* samples = new T[sampleCount];
+  for (size_t i = 0; i < sampleCount; i++) {
+    uint32_t channel = (i % aChannels) + 1;
+    samples[i] = FloatToAudioSample<T>(1. / channel);
+  }
+  return samples;
+}
+
+template <typename T>
+void DeleteInterleavedChannelArray(const T* aArray) {
+  delete[] aArray;
+}
+
+bool FuzzyEqual(float aLhs, float aRhs) { return std::abs(aLhs - aRhs) < 0.01; }
+
+template <typename SrcT, typename DstT>
+void TestInterleaveAndConvert() {
+  size_t arraySize = 1024;
+  size_t maxChannels = 8;  // 7.1
+  for (uint32_t channels = 1; channels < maxChannels; channels++) {
+    const SrcT* const* src = GetPlanarChannelArray<SrcT>(channels, arraySize);
+    DstT* dst = new DstT[channels * arraySize];
+
+    InterleaveAndConvertBuffer(src, arraySize, 1.0, channels, dst);
+
+    uint32_t channelIndex = 0;
+    for (size_t i = 0; i < arraySize * channels; i++) {
+      ASSERT_TRUE(FuzzyEqual(
+          dst[i], FloatToAudioSample<DstT>(1. / (channelIndex + 1))));
+      channelIndex++;
+      channelIndex %= channels;
+    }
+
+    DeletePlanarChannelsArray(src, channels);
+    delete[] dst;
+  }
+}
+
+template <typename SrcT, typename DstT>
+void TestDeinterleaveAndConvert() {
+  size_t arraySize = 1024;
+  size_t maxChannels = 8;  // 7.1
+  for (uint32_t channels = 1; channels < maxChannels; channels++) {
+    const SrcT* src = GetInterleavedChannelArray<SrcT>(channels, arraySize);
+    DstT** dst = GetPlanarArray<DstT>(channels, arraySize);
+
+    DeinterleaveAndConvertBuffer(src, arraySize, channels, dst);
+
+    for (size_t channel = 0; channel < channels; channel++) {
+      for (size_t i = 0; i < arraySize; i++) {
+        ASSERT_TRUE(FuzzyEqual(dst[channel][i],
+                               FloatToAudioSample<DstT>(1. / (channel + 1))));
+      }
+    }
+
+    DeleteInterleavedChannelArray(src);
+    DeletePlanarArray(dst, channels);
+  }
+}
+
+uint8_t gSilence[4096] = {0};
+
+template <typename T>
+T* SilentChannel() {
+  return reinterpret_cast<T*>(gSilence);
+}
+
+template <typename T>
+void TestUpmixStereo() {
+  size_t arraySize = 1024;
+  nsTArray<T*> channels;
+  nsTArray<const T*> channelsptr;
+
+  channels.SetLength(1);
+  channelsptr.SetLength(1);
+
+  channels[0] = new T[arraySize];
+
+  for (size_t i = 0; i < arraySize; i++) {
+    channels[0][i] = GetHighValue<T>();
+  }
+  channelsptr[0] = channels[0];
+
+  AudioChannelsUpMix(&channelsptr, 2, SilentChannel<T>());
+
+  for (size_t channel = 0; channel < 2; channel++) {
+    for (size_t i = 0; i < arraySize; i++) {
+      ASSERT_TRUE(channelsptr[channel][i] == GetHighValue<T>());
+    }
+  }
+  delete[] channels[0];
+}
+
+template <typename T>
+void TestDownmixStereo() {
+  const size_t arraySize = 1024;
+  nsTArray<const T*> inputptr;
+  nsTArray<T*> input;
+  T** output;
+
+  output = new T*[1];
+  output[0] = new T[arraySize];
+
+  input.SetLength(2);
+  inputptr.SetLength(2);
+
+  for (size_t channel = 0; channel < input.Length(); channel++) {
+    input[channel] = new T[arraySize];
+    for (size_t i = 0; i < arraySize; i++) {
+      input[channel][i] = channel == 0 ? GetLowValue<T>() : GetHighValue<T>();
+    }
+    inputptr[channel] = input[channel];
+  }
+
+  AudioChannelsDownMix(inputptr, output, 1, arraySize);
+
+  for (size_t i = 0; i < arraySize; i++) {
+    ASSERT_TRUE(output[0][i] == GetSilentValue<T>());
+    ASSERT_TRUE(output[0][i] == GetSilentValue<T>());
+  }
+
+  delete[] output[0];
+  delete[] output;
+}
+
+TEST(AudioSegment, Test)
+{
+  TestInterleaveAndConvert<float, float>();
+  TestInterleaveAndConvert<float, int16_t>();
+  TestInterleaveAndConvert<int16_t, float>();
+  TestInterleaveAndConvert<int16_t, int16_t>();
+  TestDeinterleaveAndConvert<float, float>();
+  TestDeinterleaveAndConvert<float, int16_t>();
+  TestDeinterleaveAndConvert<int16_t, float>();
+  TestDeinterleaveAndConvert<int16_t, int16_t>();
+  TestUpmixStereo<float>();
+  TestUpmixStereo<int16_t>();
+  TestDownmixStereo<float>();
+  TestDownmixStereo<int16_t>();
+}
+
+template <class T, uint32_t Channels>
+void fillChunk(AudioChunk* aChunk, int aDuration) {
+  static_assert(Channels != 0, "Filling 0 channels is a no-op");
+
+  aChunk->mDuration = aDuration;
+
+  AutoTArray<nsTArray<T>, Channels> buffer;
+  buffer.SetLength(Channels);
+  aChunk->mChannelData.ClearAndRetainStorage();
+  aChunk->mChannelData.SetCapacity(Channels);
+  for (nsTArray<T>& channel : buffer) {
+    T* ch = channel.AppendElements(aDuration);
+    for (int i = 0; i < aDuration; ++i) {
+      ch[i] = GetHighValue<T>();
+    }
+    aChunk->mChannelData.AppendElement(ch);
+  }
+
+  aChunk->mBuffer = new mozilla::SharedChannelArrayBuffer<T>(std::move(buffer));
+  aChunk->mBufferFormat = AudioSampleTypeToFormat<T>::Format;
+}
+
+TEST(AudioSegment, FlushAfter_ZeroDuration)
+{
+  AudioChunk c;
+  fillChunk<float, 2>(&c, 10);
+
+  AudioSegment s;
+  s.AppendAndConsumeChunk(std::move(c));
+  s.FlushAfter(0);
+  EXPECT_EQ(s.GetDuration(), 0);
+}
+
+TEST(AudioSegment, FlushAfter_SmallerDuration)
+{
+  // It was crashing when the first chunk was silence (null) and FlushAfter
+  // was called for a duration, smaller or equal to the duration of the
+  // first chunk.
+  TrackTime duration = 10;
+  TrackTime smaller_duration = 8;
+  AudioChunk c1;
+  c1.SetNull(duration);
+  AudioChunk c2;
+  fillChunk<float, 2>(&c2, duration);
+
+  AudioSegment s;
+  s.AppendAndConsumeChunk(std::move(c1));
+  s.AppendAndConsumeChunk(std::move(c2));
+  s.FlushAfter(smaller_duration);
+  EXPECT_EQ(s.GetDuration(), smaller_duration) << "Check new duration";
+
+  TrackTime chunkByChunkDuration = 0;
+  for (AudioSegment::ChunkIterator iter(s); !iter.IsEnded(); iter.Next()) {
+    chunkByChunkDuration += iter->GetDuration();
+  }
+  EXPECT_EQ(s.GetDuration(), chunkByChunkDuration)
+      << "Confirm duration chunk by chunk";
+}
+
+TEST(AudioSegment, MemoizedOutputChannelCount)
+{
+  AudioSegment s;
+  EXPECT_EQ(s.MaxChannelCount(), 0U) << "0 channels on init";
+
+  s.AppendNullData(1);
+  EXPECT_EQ(s.MaxChannelCount(), 0U) << "Null data has 0 channels";
+
+  s.Clear();
+  EXPECT_EQ(s.MaxChannelCount(), 0U) << "Still 0 after clearing";
+
+  AudioChunk c1;
+  fillChunk<float, 1>(&c1, 1);
+  s.AppendAndConsumeChunk(std::move(c1));
+  EXPECT_EQ(s.MaxChannelCount(), 1U) << "A single chunk's channel count";
+
+  AudioChunk c2;
+  fillChunk<float, 2>(&c2, 1);
+  s.AppendAndConsumeChunk(std::move(c2));
+  EXPECT_EQ(s.MaxChannelCount(), 2U) << "The max of two chunks' channel count";
+
+  s.ForgetUpTo(2);
+  EXPECT_EQ(s.MaxChannelCount(), 2U) << "Memoized value with null chunks";
+
+  s.Clear();
+  EXPECT_EQ(s.MaxChannelCount(), 2U) << "Still memoized after clearing";
+
+  AudioChunk c3;
+  fillChunk<float, 1>(&c3, 1);
+  s.AppendAndConsumeChunk(std::move(c3));
+  EXPECT_EQ(s.MaxChannelCount(), 1U) << "Real chunk trumps memoized value";
+
+  s.Clear();
+  EXPECT_EQ(s.MaxChannelCount(), 1U) << "Memoized value was updated";
+}
+
+TEST(AudioSegment, AppendAndConsumeChunk)
+{
+  AudioChunk c;
+  fillChunk<float, 2>(&c, 10);
+  AudioChunk temp(c);
+  EXPECT_TRUE(c.mBuffer->IsShared());
+
+  AudioSegment s;
+  s.AppendAndConsumeChunk(std::move(temp));
+  EXPECT_FALSE(s.IsEmpty());
+  EXPECT_TRUE(c.mBuffer->IsShared());
+
+  s.Clear();
+  EXPECT_FALSE(c.mBuffer->IsShared());
+}
+
+TEST(AudioSegment, AppendAndConsumeEmptyChunk)
+{
+  AudioChunk c;
+  AudioSegment s;
+  s.AppendAndConsumeChunk(std::move(c));
+  EXPECT_TRUE(s.IsEmpty());
+}
+
+TEST(AudioSegment, AppendAndConsumeNonEmptyZeroDurationChunk)
+{
+  AudioChunk c;
+  fillChunk<float, 2>(&c, 0);
+  AudioChunk temp(c);
+  EXPECT_TRUE(c.mBuffer->IsShared());
+
+  AudioSegment s;
+  s.AppendAndConsumeChunk(std::move(temp));
+  EXPECT_TRUE(s.IsEmpty());
+  EXPECT_FALSE(c.mBuffer->IsShared());
+}
+
+TEST(AudioSegment, CombineChunksInAppendAndConsumeChunk)
+{
+  AudioChunk source;
+  fillChunk<float, 2>(&source, 10);
+
+  auto checkChunks = [&](const AudioSegment& aSegement,
+                         const nsTArray<TrackTime>& aDurations) {
+    size_t i = 0;
+    for (AudioSegment::ConstChunkIterator iter(aSegement); !iter.IsEnded();
+         iter.Next()) {
+      EXPECT_EQ(iter->GetDuration(), aDurations[i++]);
+    }
+    EXPECT_EQ(i, aDurations.Length());
+  };
+
+  // The chunks can be merged if their duration are adjacent.
+  {
+    AudioChunk c1(source);
+    c1.SliceTo(2, 5);
+
+    AudioChunk c2(source);
+    c2.SliceTo(5, 9);
+
+    AudioSegment s;
+    s.AppendAndConsumeChunk(std::move(c1));
+    EXPECT_EQ(s.GetDuration(), 3);
+
+    s.AppendAndConsumeChunk(std::move(c2));
+    EXPECT_EQ(s.GetDuration(), 7);
+
+    checkChunks(s, {7});
+  }
+  // Otherwise, they cannot be merged.
+  {
+    // If durations of chunks are overlapped, they cannot be merged.
+    AudioChunk c1(source);
+    c1.SliceTo(2, 5);
+
+    AudioChunk c2(source);
+    c2.SliceTo(4, 9);
+
+    AudioSegment s;
+    s.AppendAndConsumeChunk(std::move(c1));
+    EXPECT_EQ(s.GetDuration(), 3);
+
+    s.AppendAndConsumeChunk(std::move(c2));
+    EXPECT_EQ(s.GetDuration(), 8);
+
+    checkChunks(s, {3, 5});
+  }
+  {
+    // If durations of chunks are discontinuous, they cannot be merged.
+    AudioChunk c1(source);
+    c1.SliceTo(2, 4);
+
+    AudioChunk c2(source);
+    c2.SliceTo(5, 9);
+
+    AudioSegment s;
+    s.AppendAndConsumeChunk(std::move(c1));
+    EXPECT_EQ(s.GetDuration(), 2);
+
+    s.AppendAndConsumeChunk(std::move(c2));
+    EXPECT_EQ(s.GetDuration(), 6);
+
+    checkChunks(s, {2, 4});
+  }
+}
+
+TEST(AudioSegment, ConvertFromAndToInterleaved)
+{
+  const uint32_t channels = 2;
+  const uint32_t rate = 44100;
+  AudioGenerator<AudioDataValue> generator(channels, rate);
+
+  const size_t frames = 10;
+  const size_t bufferSize = frames * channels;
+  nsTArray<AudioDataValue> buffer(bufferSize);
+  buffer.AppendElements(bufferSize);
+
+  generator.GenerateInterleaved(buffer.Elements(), frames);
+
+  AudioSegment data;
+  data.AppendFromInterleavedBuffer(buffer.Elements(), frames, channels,
+                                   PRINCIPAL_HANDLE_NONE);
+
+  nsTArray<AudioDataValue> interleaved;
+  size_t sampleCount = data.WriteToInterleavedBuffer(interleaved, channels);
+
+  EXPECT_EQ(sampleCount, bufferSize);
+  EXPECT_EQ(interleaved, buffer);
+}
+
+}  // namespace audio_segment