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Diffstat (limited to 'dom/media/gtest/TestAudioSegment.cpp')
| -rw-r--r-- | dom/media/gtest/TestAudioSegment.cpp | 470 |
1 files changed, 470 insertions, 0 deletions
diff --git a/dom/media/gtest/TestAudioSegment.cpp b/dom/media/gtest/TestAudioSegment.cpp new file mode 100644 index 0000000000..64366045ca --- /dev/null +++ b/dom/media/gtest/TestAudioSegment.cpp @@ -0,0 +1,470 @@ +/* -*- 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 |