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-rw-r--r--dom/media/gtest/TestAudioSegment.cpp257
1 files changed, 257 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"
+
+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>();
+}
+
+} // namespace audio_segment