| // Copyright 2012 The Chromium Authors |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "media/base/audio_converter.h" |
| |
| #include <stddef.h> |
| |
| #include <memory> |
| #include <tuple> |
| |
| #include "base/strings/string_number_conversions.h" |
| #include "media/base/audio_timestamp_helper.h" |
| #include "media/base/fake_audio_render_callback.h" |
| #include "testing/gmock/include/gmock/gmock.h" |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| namespace media { |
| |
| // Parameters which control the many input case tests. |
| static const int kConvertInputs = 8; |
| static const int kConvertCycles = 3; |
| |
| // Parameters used for testing. |
| static constexpr ChannelLayout kChannelLayout = CHANNEL_LAYOUT_STEREO; |
| static const int kHighLatencyBufferSize = 2048; |
| static const int kLowLatencyBufferSize = 256; |
| static const int kSampleRate = 48000; |
| |
| // Number of full sine wave cycles for each Render() call. |
| static const int kSineCycles = 4; |
| |
| // Tuple of <input rate, output rate, output channel layout config, epsilon>. |
| typedef std::tuple<int, int, ChannelLayoutConfig, double> |
| AudioConverterTestData; |
| class AudioConverterTest |
| : public testing::TestWithParam<AudioConverterTestData> { |
| public: |
| AudioConverterTest() : epsilon_(std::get<3>(GetParam())) { |
| // Create input and output parameters based on test parameters. |
| input_parameters_ = |
| AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, |
| ChannelLayoutConfig::FromLayout<kChannelLayout>(), |
| std::get<0>(GetParam()), kHighLatencyBufferSize); |
| output_parameters_ = AudioParameters( |
| AudioParameters::AUDIO_PCM_LOW_LATENCY, std::get<2>(GetParam()), |
| std::get<1>(GetParam()), kLowLatencyBufferSize); |
| |
| converter_ = std::make_unique<AudioConverter>(input_parameters_, |
| output_parameters_, false); |
| |
| audio_bus_ = AudioBus::Create(output_parameters_); |
| expected_audio_bus_ = AudioBus::Create(output_parameters_); |
| |
| // Allocate one callback for generating expected results. |
| double step = kSineCycles / static_cast<double>( |
| output_parameters_.frames_per_buffer()); |
| expected_callback_ = |
| std::make_unique<FakeAudioRenderCallback>(step, kSampleRate); |
| } |
| |
| AudioConverterTest(const AudioConverterTest&) = delete; |
| AudioConverterTest& operator=(const AudioConverterTest&) = delete; |
| |
| // Creates |count| input callbacks to be used for conversion testing. |
| void InitializeInputs(int count) { |
| // Setup FakeAudioRenderCallback step to compensate for resampling. |
| double scale_factor = input_parameters_.sample_rate() / |
| static_cast<double>(output_parameters_.sample_rate()); |
| double step = kSineCycles / (scale_factor * |
| static_cast<double>(output_parameters_.frames_per_buffer())); |
| |
| for (int i = 0; i < count; ++i) { |
| fake_callbacks_.push_back( |
| std::make_unique<FakeAudioRenderCallback>(step, kSampleRate)); |
| converter_->AddInput(fake_callbacks_[i].get()); |
| } |
| } |
| |
| // Resets all input callbacks to a pristine state. |
| void Reset() { |
| converter_->Reset(); |
| for (size_t i = 0; i < fake_callbacks_.size(); ++i) |
| fake_callbacks_[i]->reset(); |
| expected_callback_->reset(); |
| } |
| |
| // Sets the volume on all input callbacks to |volume|. |
| void SetVolume(float volume) { |
| for (size_t i = 0; i < fake_callbacks_.size(); ++i) |
| fake_callbacks_[i]->set_volume(volume); |
| } |
| |
| // Validates audio data between |audio_bus_| and |expected_audio_bus_| from |
| // |index|..|frames| after |scale| is applied to the expected audio data. |
| bool ValidateAudioData(int index, int frames, float scale) { |
| for (int i = 0; i < audio_bus_->channels(); ++i) { |
| for (int j = index; j < frames; ++j) { |
| double error = fabs(audio_bus_->channel(i)[j] - |
| expected_audio_bus_->channel(i)[j] * scale); |
| if (error > epsilon_) { |
| EXPECT_NEAR(expected_audio_bus_->channel(i)[j] * scale, |
| audio_bus_->channel(i)[j], epsilon_) |
| << " i=" << i << ", j=" << j; |
| return false; |
| } |
| } |
| } |
| return true; |
| } |
| |
| // Runs a single Convert() stage, fills |expected_audio_bus_| appropriately, |
| // and validates equality with |audio_bus_| after |scale| is applied. |
| bool RenderAndValidateAudioData(float scale) { |
| // Render actual audio data. |
| converter_->Convert(audio_bus_.get()); |
| |
| // Render expected audio data. |
| expected_callback_->Render(base::TimeDelta(), base::TimeTicks::Now(), {}, |
| expected_audio_bus_.get()); |
| |
| // Zero out unused channels in the expected AudioBus just as AudioConverter |
| // would during channel mixing. |
| for (int i = input_parameters_.channels(); |
| i < output_parameters_.channels(); ++i) { |
| memset(expected_audio_bus_->channel(i), 0, |
| audio_bus_->frames() * sizeof(*audio_bus_->channel(i))); |
| } |
| |
| return ValidateAudioData(0, audio_bus_->frames(), scale); |
| } |
| |
| // Fills |audio_bus_| fully with |value|. |
| void FillAudioData(float value) { |
| for (int i = 0; i < audio_bus_->channels(); ++i) { |
| std::fill(audio_bus_->channel(i), |
| audio_bus_->channel(i) + audio_bus_->frames(), value); |
| } |
| } |
| |
| // Verifies converter output with a |inputs| number of transform inputs. |
| void RunTest(int inputs) { |
| InitializeInputs(inputs); |
| |
| SetVolume(0); |
| for (int i = 0; i < kConvertCycles; ++i) |
| ASSERT_TRUE(RenderAndValidateAudioData(0)); |
| |
| Reset(); |
| |
| // Set a different volume for each input and verify the results. |
| float total_scale = 0; |
| for (size_t i = 0; i < fake_callbacks_.size(); ++i) { |
| float volume = static_cast<float>(i) / fake_callbacks_.size(); |
| total_scale += volume; |
| fake_callbacks_[i]->set_volume(volume); |
| } |
| for (int i = 0; i < kConvertCycles; ++i) |
| ASSERT_TRUE(RenderAndValidateAudioData(total_scale)); |
| |
| Reset(); |
| |
| // Remove every other input. |
| for (size_t i = 1; i < fake_callbacks_.size(); i += 2) |
| converter_->RemoveInput(fake_callbacks_[i].get()); |
| |
| SetVolume(1); |
| float scale = inputs > 1 ? inputs / 2.0f : inputs; |
| for (int i = 0; i < kConvertCycles; ++i) |
| ASSERT_TRUE(RenderAndValidateAudioData(scale)); |
| } |
| |
| protected: |
| virtual ~AudioConverterTest() = default; |
| |
| // Converter under test. |
| std::unique_ptr<AudioConverter> converter_; |
| |
| // Input and output parameters used for AudioConverter construction. |
| AudioParameters input_parameters_; |
| AudioParameters output_parameters_; |
| |
| // Destination AudioBus for AudioConverter output. |
| std::unique_ptr<AudioBus> audio_bus_; |
| |
| // AudioBus containing expected results for comparison with |audio_bus_|. |
| std::unique_ptr<AudioBus> expected_audio_bus_; |
| |
| // Vector of all input callbacks used to drive AudioConverter::Convert(). |
| std::vector<std::unique_ptr<FakeAudioRenderCallback>> fake_callbacks_; |
| |
| // Parallel input callback which generates the expected output. |
| std::unique_ptr<FakeAudioRenderCallback> expected_callback_; |
| |
| // Epsilon value with which to perform comparisons between |audio_bus_| and |
| // |expected_audio_bus_|. |
| double epsilon_; |
| }; |
| |
| // Ensure the buffer delay provided by AudioConverter is accurate. |
| TEST(AudioConverterTest, AudioDelayAndDiscreteChannelCount) { |
| // Choose input and output parameters such that the transform must make |
| // multiple calls to fill the buffer. |
| AudioParameters input_parameters(AudioParameters::AUDIO_PCM_LINEAR, |
| {CHANNEL_LAYOUT_DISCRETE, 10}, kSampleRate, |
| kLowLatencyBufferSize); |
| AudioParameters output_parameters(AudioParameters::AUDIO_PCM_LINEAR, |
| {CHANNEL_LAYOUT_DISCRETE, 5}, |
| kSampleRate * 2, kHighLatencyBufferSize); |
| |
| AudioConverter converter(input_parameters, output_parameters, false); |
| FakeAudioRenderCallback callback(0.2, kSampleRate); |
| std::unique_ptr<AudioBus> audio_bus = AudioBus::Create(output_parameters); |
| converter.AddInput(&callback); |
| converter.Convert(audio_bus.get()); |
| |
| // double input_sample_rate = input_parameters.sample_rate(); |
| // int fill_count = |
| // (output_parameters.frames_per_buffer() * input_sample_rate / |
| // output_parameters.sample_rate()) / |
| // input_parameters.frames_per_buffer(); |
| // |
| // This magic number is the accumulated MultiChannelResampler delay after |
| // |fill_count| (4) callbacks to provide input. The number of frames delayed |
| // is an implementation detail of the SincResampler chunk size (448 for the |
| // first two callbacks, 512 for the last two callbacks). See |
| // SincResampler.ChunkSize(). |
| int kExpectedDelay = 960; |
| auto expected_delay = |
| AudioTimestampHelper::FramesToTime(kExpectedDelay, kSampleRate); |
| EXPECT_EQ(expected_delay, callback.last_delay()); |
| EXPECT_EQ(input_parameters.channels(), callback.last_channel_count()); |
| } |
| |
| // Ensure that glitch info is propagated to all callbacks. |
| TEST(AudioConverterTest, PropagatesGlitchInfo) { |
| // Choose input and output parameters such that the transform must make |
| // multiple calls to fill the buffer. |
| AudioParameters input_parameters(AudioParameters::AUDIO_PCM_LINEAR, |
| ChannelLayoutConfig::Stereo(), kSampleRate, |
| kLowLatencyBufferSize); |
| AudioParameters output_parameters(AudioParameters::AUDIO_PCM_LINEAR, |
| ChannelLayoutConfig::Stereo(), |
| kSampleRate * 2, kHighLatencyBufferSize); |
| AudioGlitchInfo glitch_info{.duration = base::Seconds(5), .count = 123}; |
| |
| AudioConverter converter(input_parameters, output_parameters, false); |
| FakeAudioRenderCallback callback1(0.2, kSampleRate); |
| FakeAudioRenderCallback callback2(0.2, kSampleRate); |
| std::unique_ptr<AudioBus> audio_bus = AudioBus::Create(output_parameters); |
| converter.AddInput(&callback1); |
| converter.AddInput(&callback2); |
| |
| // Send no glitches, so the cumulative glitches should remain at 0. |
| converter.ConvertWithInfo(0, {}, audio_bus.get()); |
| EXPECT_EQ(callback1.cumulative_glitch_info(), AudioGlitchInfo()); |
| EXPECT_EQ(callback2.cumulative_glitch_info(), AudioGlitchInfo()); |
| |
| // Send glitches, and expect them to be forwarded to the callbacks. The |
| // callbacks will be called several times due to their differing buffer sizes, |
| // but the glitch info should only be passed on once. |
| converter.ConvertWithInfo(0, glitch_info, audio_bus.get()); |
| EXPECT_EQ(callback1.cumulative_glitch_info(), glitch_info); |
| EXPECT_EQ(callback2.cumulative_glitch_info(), glitch_info); |
| |
| // Send no glitches, so the cumulative glitches should remain unchanged. |
| converter.ConvertWithInfo(0, {}, audio_bus.get()); |
| EXPECT_EQ(callback1.cumulative_glitch_info(), glitch_info); |
| EXPECT_EQ(callback2.cumulative_glitch_info(), glitch_info); |
| } |
| |
| TEST_P(AudioConverterTest, ArbitraryOutputRequestSize) { |
| // Resize output bus to be half of |output_parameters_|'s frames_per_buffer(). |
| audio_bus_ = AudioBus::Create(output_parameters_.channels(), |
| output_parameters_.frames_per_buffer() / 2); |
| RunTest(1); |
| } |
| |
| TEST_P(AudioConverterTest, NoInputs) { |
| FillAudioData(1.0f); |
| EXPECT_TRUE(RenderAndValidateAudioData(0.0f)); |
| } |
| |
| TEST_P(AudioConverterTest, OneInput) { |
| RunTest(1); |
| } |
| |
| TEST_P(AudioConverterTest, ManyInputs) { |
| RunTest(kConvertInputs); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P( |
| AudioConverterTest, |
| AudioConverterTest, |
| testing::Values( |
| // No resampling. No channel mixing. |
| std::make_tuple(44100, |
| 44100, |
| ChannelLayoutConfig::Stereo(), |
| 0.00000048), |
| |
| // Upsampling. Channel upmixing. |
| std::make_tuple(44100, |
| 48000, |
| ChannelLayoutConfig::FromLayout<CHANNEL_LAYOUT_QUAD>(), |
| 0.033), |
| |
| // Downsampling. Channel downmixing. |
| std::make_tuple(48000, 41000, ChannelLayoutConfig::Mono(), 0.042))); |
| |
| } // namespace media |