| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "modules/audio_processing/audio_processing_impl.h" |
| |
| #include <algorithm> |
| #include <array> |
| #include <memory> |
| #include <tuple> |
| |
| #include "absl/types/optional.h" |
| #include "api/make_ref_counted.h" |
| #include "api/scoped_refptr.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/optionally_built_submodule_creators.h" |
| #include "modules/audio_processing/test/audio_processing_builder_for_testing.h" |
| #include "modules/audio_processing/test/echo_canceller_test_tools.h" |
| #include "modules/audio_processing/test/echo_control_mock.h" |
| #include "modules/audio_processing/test/test_utils.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/random.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "test/field_trial.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| using ::testing::Invoke; |
| using ::testing::NotNull; |
| |
| class MockInitialize : public AudioProcessingImpl { |
| public: |
| MockInitialize() : AudioProcessingImpl() {} |
| |
| MOCK_METHOD(void, InitializeLocked, (), (override)); |
| void RealInitializeLocked() { |
| AssertLockedForTest(); |
| AudioProcessingImpl::InitializeLocked(); |
| } |
| |
| MOCK_METHOD(void, AddRef, (), (const, override)); |
| MOCK_METHOD(RefCountReleaseStatus, Release, (), (const, override)); |
| }; |
| |
| // Creates MockEchoControl instances and provides a raw pointer access to |
| // the next created one. The raw pointer is meant to be used with gmock. |
| // Returning a pointer of the next created MockEchoControl instance is necessary |
| // for the following reasons: (i) gmock expectations must be set before any call |
| // occurs, (ii) APM is initialized the first time that |
| // AudioProcessingImpl::ProcessStream() is called and the initialization leads |
| // to the creation of a new EchoControl object. |
| class MockEchoControlFactory : public EchoControlFactory { |
| public: |
| MockEchoControlFactory() : next_mock_(std::make_unique<MockEchoControl>()) {} |
| // Returns a pointer to the next MockEchoControl that this factory creates. |
| MockEchoControl* GetNext() const { return next_mock_.get(); } |
| std::unique_ptr<EchoControl> Create(int sample_rate_hz, |
| int num_render_channels, |
| int num_capture_channels) override { |
| std::unique_ptr<EchoControl> mock = std::move(next_mock_); |
| next_mock_ = std::make_unique<MockEchoControl>(); |
| return mock; |
| } |
| |
| private: |
| std::unique_ptr<MockEchoControl> next_mock_; |
| }; |
| |
| // Mocks EchoDetector and records the first samples of the last analyzed render |
| // stream frame. Used to check what data is read by an EchoDetector |
| // implementation injected into an APM. |
| class TestEchoDetector : public EchoDetector { |
| public: |
| TestEchoDetector() |
| : analyze_render_audio_called_(false), |
| last_render_audio_first_sample_(0.f) {} |
| ~TestEchoDetector() override = default; |
| void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override { |
| last_render_audio_first_sample_ = render_audio[0]; |
| analyze_render_audio_called_ = true; |
| } |
| void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override { |
| } |
| void Initialize(int capture_sample_rate_hz, |
| int num_capture_channels, |
| int render_sample_rate_hz, |
| int num_render_channels) override {} |
| EchoDetector::Metrics GetMetrics() const override { return {}; } |
| // Returns true if AnalyzeRenderAudio() has been called at least once. |
| bool analyze_render_audio_called() const { |
| return analyze_render_audio_called_; |
| } |
| // Returns the first sample of the last analyzed render frame. |
| float last_render_audio_first_sample() const { |
| return last_render_audio_first_sample_; |
| } |
| |
| private: |
| bool analyze_render_audio_called_; |
| float last_render_audio_first_sample_; |
| }; |
| |
| // Mocks CustomProcessing and applies ProcessSample() to all the samples. |
| // Meant to be injected into an APM to modify samples in a known and detectable |
| // way. |
| class TestRenderPreProcessor : public CustomProcessing { |
| public: |
| TestRenderPreProcessor() = default; |
| ~TestRenderPreProcessor() = default; |
| void Initialize(int sample_rate_hz, int num_channels) override {} |
| void Process(AudioBuffer* audio) override { |
| for (size_t k = 0; k < audio->num_channels(); ++k) { |
| rtc::ArrayView<float> channel_view(audio->channels()[k], |
| audio->num_frames()); |
| std::transform(channel_view.begin(), channel_view.end(), |
| channel_view.begin(), ProcessSample); |
| } |
| } |
| std::string ToString() const override { return "TestRenderPreProcessor"; } |
| void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {} |
| // Modifies a sample. This member is used in Process() to modify a frame and |
| // it is publicly visible to enable tests. |
| static constexpr float ProcessSample(float x) { return 2.f * x; } |
| }; |
| |
| // Runs `apm` input processing for volume adjustments for `num_frames` random |
| // frames starting from the volume `initial_volume`. This includes three steps: |
| // 1) Set the input volume 2) Process the stream 3) Set the new recommended |
| // input volume. Returns the new recommended input volume. |
| int ProcessInputVolume(AudioProcessing& apm, |
| int num_frames, |
| int initial_volume) { |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz, |
| /*num_channels=*/kNumChannels); |
| int recommended_input_volume = initial_volume; |
| for (int i = 0; i < num_frames; ++i) { |
| Random random_generator(2341U); |
| RandomizeSampleVector(&random_generator, buffer); |
| |
| apm.set_stream_analog_level(recommended_input_volume); |
| apm.ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers); |
| recommended_input_volume = apm.recommended_stream_analog_level(); |
| } |
| return recommended_input_volume; |
| } |
| |
| } // namespace |
| |
| TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { |
| MockInitialize mock; |
| ON_CALL(mock, InitializeLocked) |
| .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); |
| |
| EXPECT_CALL(mock, InitializeLocked).Times(1); |
| mock.Initialize(); |
| |
| constexpr size_t kMaxSampleRateHz = 32000; |
| constexpr size_t kMaxNumChannels = 2; |
| std::array<int16_t, kMaxNumChannels * kMaxSampleRateHz / 100> frame; |
| frame.fill(0); |
| StreamConfig config(16000, 1); |
| // Call with the default parameters; there should be an init. |
| EXPECT_CALL(mock, InitializeLocked).Times(0); |
| EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data())); |
| EXPECT_NOERR( |
| mock.ProcessReverseStream(frame.data(), config, config, frame.data())); |
| |
| // New sample rate. (Only impacts ProcessStream). |
| config = StreamConfig(32000, 1); |
| EXPECT_CALL(mock, InitializeLocked).Times(1); |
| EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data())); |
| |
| // New number of channels. |
| config = StreamConfig(32000, 2); |
| EXPECT_CALL(mock, InitializeLocked).Times(2); |
| EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data())); |
| EXPECT_NOERR( |
| mock.ProcessReverseStream(frame.data(), config, config, frame.data())); |
| |
| // A new sample rate passed to ProcessReverseStream should cause an init. |
| config = StreamConfig(16000, 2); |
| EXPECT_CALL(mock, InitializeLocked).Times(1); |
| EXPECT_NOERR( |
| mock.ProcessReverseStream(frame.data(), config, config, frame.data())); |
| } |
| |
| TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) { |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting().Create(); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.pre_amplifier.enabled = true; |
| apm_config.pre_amplifier.fixed_gain_factor = 1.f; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kNumChannels = 2; |
| |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| EXPECT_EQ(frame[100], kAudioLevel) |
| << "With factor 1, frame shouldn't be modified."; |
| |
| constexpr float kGainFactor = 2.f; |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor)); |
| |
| // Process for two frames to have time to ramp up gain. |
| for (int i = 0; i < 2; ++i) { |
| frame.fill(kAudioLevel); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| } |
| EXPECT_EQ(frame[100], kGainFactor * kAudioLevel) |
| << "Frame should be amplified."; |
| } |
| |
| TEST(AudioProcessingImplTest, |
| LevelAdjustmentUpdateCapturePreGainRuntimeSetting) { |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting().Create(); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.capture_level_adjustment.enabled = true; |
| apm_config.capture_level_adjustment.pre_gain_factor = 1.f; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kNumChannels = 2; |
| |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| EXPECT_EQ(frame[100], kAudioLevel) |
| << "With factor 1, frame shouldn't be modified."; |
| |
| constexpr float kGainFactor = 2.f; |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor)); |
| |
| // Process for two frames to have time to ramp up gain. |
| for (int i = 0; i < 2; ++i) { |
| frame.fill(kAudioLevel); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| } |
| EXPECT_EQ(frame[100], kGainFactor * kAudioLevel) |
| << "Frame should be amplified."; |
| } |
| |
| TEST(AudioProcessingImplTest, |
| LevelAdjustmentUpdateCapturePostGainRuntimeSetting) { |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting().Create(); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.capture_level_adjustment.enabled = true; |
| apm_config.capture_level_adjustment.post_gain_factor = 1.f; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kNumChannels = 2; |
| |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| EXPECT_EQ(frame[100], kAudioLevel) |
| << "With factor 1, frame shouldn't be modified."; |
| |
| constexpr float kGainFactor = 2.f; |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePostGain(kGainFactor)); |
| |
| // Process for two frames to have time to ramp up gain. |
| for (int i = 0; i < 2; ++i) { |
| frame.fill(kAudioLevel); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| } |
| EXPECT_EQ(frame[100], kGainFactor * kAudioLevel) |
| << "Frame should be amplified."; |
| } |
| |
| TEST(AudioProcessingImplTest, EchoControllerObservesSetCaptureUsageChange) { |
| // Tests that the echo controller observes that the capture usage has been |
| // updated. |
| auto echo_control_factory = std::make_unique<MockEchoControlFactory>(); |
| const MockEchoControlFactory* echo_control_factory_ptr = |
| echo_control_factory.get(); |
| |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetEchoControlFactory(std::move(echo_control_factory)) |
| .Create(); |
| |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 2; |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| |
| MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext(); |
| |
| // Ensure that SetCaptureOutputUsage is not called when no runtime settings |
| // are passed. |
| EXPECT_CALL(*echo_control_mock, SetCaptureOutputUsage(testing::_)).Times(0); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| |
| // Ensure that SetCaptureOutputUsage is called with the right information when |
| // a runtime setting is passed. |
| EXPECT_CALL(*echo_control_mock, |
| SetCaptureOutputUsage(/*capture_output_used=*/false)) |
| .Times(1); |
| EXPECT_TRUE(apm->PostRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( |
| /*capture_output_used=*/false))); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| |
| EXPECT_CALL(*echo_control_mock, |
| SetCaptureOutputUsage(/*capture_output_used=*/true)) |
| .Times(1); |
| EXPECT_TRUE(apm->PostRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( |
| /*capture_output_used=*/true))); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| |
| // The number of positions to place items in the queue is equal to the queue |
| // size minus 1. |
| constexpr int kNumSlotsInQueue = RuntimeSettingQueueSize(); |
| |
| // Ensure that SetCaptureOutputUsage is called with the right information when |
| // many runtime settings are passed. |
| for (int k = 0; k < kNumSlotsInQueue - 1; ++k) { |
| EXPECT_TRUE(apm->PostRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( |
| /*capture_output_used=*/false))); |
| } |
| EXPECT_CALL(*echo_control_mock, |
| SetCaptureOutputUsage(/*capture_output_used=*/false)) |
| .Times(kNumSlotsInQueue - 1); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| |
| // Ensure that SetCaptureOutputUsage is properly called with the fallback |
| // value when the runtime settings queue becomes full. |
| for (int k = 0; k < kNumSlotsInQueue; ++k) { |
| EXPECT_TRUE(apm->PostRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( |
| /*capture_output_used=*/false))); |
| } |
| EXPECT_FALSE(apm->PostRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( |
| /*capture_output_used=*/false))); |
| EXPECT_FALSE(apm->PostRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting( |
| /*capture_output_used=*/false))); |
| EXPECT_CALL(*echo_control_mock, |
| SetCaptureOutputUsage(/*capture_output_used=*/false)) |
| .Times(kNumSlotsInQueue); |
| EXPECT_CALL(*echo_control_mock, |
| SetCaptureOutputUsage(/*capture_output_used=*/true)) |
| .Times(1); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| EchoControllerObservesPreAmplifierEchoPathGainChange) { |
| // Tests that the echo controller observes an echo path gain change when the |
| // pre-amplifier submodule changes the gain. |
| auto echo_control_factory = std::make_unique<MockEchoControlFactory>(); |
| const auto* echo_control_factory_ptr = echo_control_factory.get(); |
| |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetEchoControlFactory(std::move(echo_control_factory)) |
| .Create(); |
| // Disable AGC. |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.gain_controller1.enabled = false; |
| apm_config.gain_controller2.enabled = false; |
| apm_config.pre_amplifier.enabled = true; |
| apm_config.pre_amplifier.fixed_gain_factor = 1.f; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| |
| MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext(); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false)) |
| .Times(1); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f)); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| EchoControllerObservesLevelAdjustmentPreGainEchoPathGainChange) { |
| // Tests that the echo controller observes an echo path gain change when the |
| // pre-amplifier submodule changes the gain. |
| auto echo_control_factory = std::make_unique<MockEchoControlFactory>(); |
| const auto* echo_control_factory_ptr = echo_control_factory.get(); |
| |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetEchoControlFactory(std::move(echo_control_factory)) |
| .Create(); |
| // Disable AGC. |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.gain_controller1.enabled = false; |
| apm_config.gain_controller2.enabled = false; |
| apm_config.capture_level_adjustment.enabled = true; |
| apm_config.capture_level_adjustment.pre_gain_factor = 1.f; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| |
| MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext(); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false)) |
| .Times(1); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f)); |
| apm->ProcessStream(frame.data(), config, config, frame.data()); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| EchoControllerObservesAnalogAgc1EchoPathGainChange) { |
| // Tests that the echo controller observes an echo path gain change when the |
| // AGC1 analog adaptive submodule changes the analog gain. |
| auto echo_control_factory = std::make_unique<MockEchoControlFactory>(); |
| const auto* echo_control_factory_ptr = echo_control_factory.get(); |
| |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetEchoControlFactory(std::move(echo_control_factory)) |
| .Create(); |
| webrtc::AudioProcessing::Config apm_config; |
| // Enable AGC1. |
| apm_config.gain_controller1.enabled = true; |
| apm_config.gain_controller1.analog_gain_controller.enabled = true; |
| apm_config.gain_controller2.enabled = false; |
| apm_config.pre_amplifier.enabled = false; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int16_t kAudioLevel = 1000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| |
| MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext(); |
| |
| constexpr int kInitialStreamAnalogLevel = 123; |
| apm->set_stream_analog_level(kInitialStreamAnalogLevel); |
| |
| // When the first fame is processed, no echo path gain change must be |
| // detected. |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false)) |
| .Times(1); |
| apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data()); |
| |
| // Simulate the application of the recommended analog level. |
| int recommended_analog_level = apm->recommended_stream_analog_level(); |
| if (recommended_analog_level == kInitialStreamAnalogLevel) { |
| // Force an analog gain change if it did not happen. |
| recommended_analog_level++; |
| } |
| apm->set_stream_analog_level(recommended_analog_level); |
| |
| // After the first fame and with a stream analog level change, the echo path |
| // gain change must be detected. |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true)) |
| .Times(1); |
| apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data()); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| ProcessWithAgc2AndTransientSuppressorVadModeDefault) { |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Disabled/"); |
| auto apm = AudioProcessingBuilder() |
| .SetConfig({.gain_controller1{.enabled = false}}) |
| .Create(); |
| ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.gain_controller1.enabled = false; |
| apm_config.gain_controller2.enabled = true; |
| apm_config.gain_controller2.adaptive_digital.enabled = true; |
| apm_config.transient_suppression.enabled = true; |
| apm->ApplyConfig(apm_config); |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz, |
| /*num_channels=*/kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcess = 10; |
| for (int i = 0; i < kFramesToProcess; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| } |
| } |
| |
| TEST(AudioProcessingImplTest, |
| ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) { |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/"); |
| rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create(); |
| ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.gain_controller1.enabled = false; |
| apm_config.gain_controller2.enabled = true; |
| apm_config.gain_controller2.adaptive_digital.enabled = true; |
| apm_config.transient_suppression.enabled = true; |
| apm->ApplyConfig(apm_config); |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz, |
| /*num_channels=*/kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcess = 10; |
| for (int i = 0; i < kFramesToProcess; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| } |
| } |
| |
| TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) { |
| // Tests that the echo controller observes an echo path gain change when a |
| // playout volume change is reported. |
| auto echo_control_factory = std::make_unique<MockEchoControlFactory>(); |
| const auto* echo_control_factory_ptr = echo_control_factory.get(); |
| |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetEchoControlFactory(std::move(echo_control_factory)) |
| .Create(); |
| // Disable AGC. |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.gain_controller1.enabled = false; |
| apm_config.gain_controller2.enabled = false; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| frame.fill(kAudioLevel); |
| |
| MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext(); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false)) |
| .Times(1); |
| apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data()); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50)); |
| apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data()); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50)); |
| apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data()); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(100)); |
| apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data()); |
| } |
| |
| TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) { |
| // Make sure that signal changes caused by a render pre-processing sub-module |
| // take place before any echo detector analysis. |
| auto test_echo_detector = rtc::make_ref_counted<TestEchoDetector>(); |
| std::unique_ptr<CustomProcessing> test_render_pre_processor( |
| new TestRenderPreProcessor()); |
| // Create APM injecting the test echo detector and render pre-processor. |
| rtc::scoped_refptr<AudioProcessing> apm = |
| AudioProcessingBuilderForTesting() |
| .SetEchoDetector(test_echo_detector) |
| .SetRenderPreProcessing(std::move(test_render_pre_processor)) |
| .Create(); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.pre_amplifier.enabled = true; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int16_t kAudioLevel = 1000; |
| constexpr int kSampleRateHz = 16000; |
| constexpr size_t kNumChannels = 1; |
| // Explicitly initialize APM to ensure no render frames are discarded. |
| const ProcessingConfig processing_config = {{ |
| {kSampleRateHz, kNumChannels}, |
| {kSampleRateHz, kNumChannels}, |
| {kSampleRateHz, kNumChannels}, |
| {kSampleRateHz, kNumChannels}, |
| }}; |
| apm->Initialize(processing_config); |
| |
| std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| |
| constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel); |
| constexpr float kExpectedPreprocessedAudioLevel = |
| TestRenderPreProcessor::ProcessSample(kAudioLevelFloat); |
| ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel); |
| |
| // Analyze a render stream frame. |
| frame.fill(kAudioLevel); |
| ASSERT_EQ(AudioProcessing::Error::kNoError, |
| apm->ProcessReverseStream(frame.data(), stream_config, |
| stream_config, frame.data())); |
| // Trigger a call to in EchoDetector::AnalyzeRenderAudio() via |
| // ProcessStream(). |
| frame.fill(kAudioLevel); |
| ASSERT_EQ(AudioProcessing::Error::kNoError, |
| apm->ProcessStream(frame.data(), stream_config, stream_config, |
| frame.data())); |
| // Regardless of how the call to in EchoDetector::AnalyzeRenderAudio() is |
| // triggered, the line below checks that the call has occurred. If not, the |
| // APM implementation may have changed and this test might need to be adapted. |
| ASSERT_TRUE(test_echo_detector->analyze_render_audio_called()); |
| // Check that the data read in EchoDetector::AnalyzeRenderAudio() is that |
| // produced by the render pre-processor. |
| EXPECT_EQ(kExpectedPreprocessedAudioLevel, |
| test_echo_detector->last_render_audio_first_sample()); |
| } |
| |
| // Disabling build-optional submodules and trying to enable them via the APM |
| // config should be bit-exact with running APM with said submodules disabled. |
| // This mainly tests that SetCreateOptionalSubmodulesForTesting has an effect. |
| TEST(ApmWithSubmodulesExcludedTest, BitexactWithDisabledModules) { |
| auto apm = rtc::make_ref_counted<AudioProcessingImpl>(); |
| ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError); |
| |
| ApmSubmoduleCreationOverrides overrides; |
| overrides.transient_suppression = true; |
| apm->OverrideSubmoduleCreationForTesting(overrides); |
| |
| AudioProcessing::Config apm_config = apm->GetConfig(); |
| apm_config.transient_suppression.enabled = true; |
| apm->ApplyConfig(apm_config); |
| |
| rtc::scoped_refptr<AudioProcessing> apm_reference = |
| AudioProcessingBuilder().Create(); |
| apm_config = apm_reference->GetConfig(); |
| apm_config.transient_suppression.enabled = false; |
| apm_reference->ApplyConfig(apm_config); |
| |
| constexpr int kSampleRateHz = 16000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| std::array<float, kSampleRateHz / 100> buffer_reference; |
| float* channel_pointers[] = {buffer.data()}; |
| float* channel_pointers_reference[] = {buffer_reference.data()}; |
| StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz, |
| /*num_channels=*/kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcessPerConfiguration = 10; |
| |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| std::copy(buffer.begin(), buffer.end(), buffer_reference.begin()); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| ASSERT_EQ( |
| apm_reference->ProcessStream(channel_pointers_reference, stream_config, |
| stream_config, channel_pointers_reference), |
| kNoErr); |
| for (int j = 0; j < kSampleRateHz / 100; ++j) { |
| EXPECT_EQ(buffer[j], buffer_reference[j]); |
| } |
| } |
| } |
| |
| // Disable transient suppressor creation and run APM in ways that should trigger |
| // calls to the transient suppressor API. |
| TEST(ApmWithSubmodulesExcludedTest, ReinitializeTransientSuppressor) { |
| auto apm = rtc::make_ref_counted<AudioProcessingImpl>(); |
| ASSERT_EQ(apm->Initialize(), kNoErr); |
| |
| ApmSubmoduleCreationOverrides overrides; |
| overrides.transient_suppression = true; |
| apm->OverrideSubmoduleCreationForTesting(overrides); |
| |
| AudioProcessing::Config config = apm->GetConfig(); |
| config.transient_suppression.enabled = true; |
| apm->ApplyConfig(config); |
| // 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels. |
| float buffer[960]; |
| float* channel_pointers[] = {&buffer[0], &buffer[480]}; |
| Random random_generator(2341U); |
| constexpr int kFramesToProcessPerConfiguration = 3; |
| |
| StreamConfig initial_stream_config(/*sample_rate_hz=*/16000, |
| /*num_channels=*/1); |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| EXPECT_EQ(apm->ProcessStream(channel_pointers, initial_stream_config, |
| initial_stream_config, channel_pointers), |
| kNoErr); |
| } |
| |
| StreamConfig stereo_stream_config(/*sample_rate_hz=*/16000, |
| /*num_channels=*/2); |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| EXPECT_EQ(apm->ProcessStream(channel_pointers, stereo_stream_config, |
| stereo_stream_config, channel_pointers), |
| kNoErr); |
| } |
| |
| StreamConfig high_sample_rate_stream_config(/*sample_rate_hz=*/48000, |
| /*num_channels=*/2); |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| EXPECT_EQ( |
| apm->ProcessStream(channel_pointers, high_sample_rate_stream_config, |
| high_sample_rate_stream_config, channel_pointers), |
| kNoErr); |
| } |
| } |
| |
| // Disable transient suppressor creation and run APM in ways that should trigger |
| // calls to the transient suppressor API. |
| TEST(ApmWithSubmodulesExcludedTest, ToggleTransientSuppressor) { |
| auto apm = rtc::make_ref_counted<AudioProcessingImpl>(); |
| ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError); |
| |
| ApmSubmoduleCreationOverrides overrides; |
| overrides.transient_suppression = true; |
| apm->OverrideSubmoduleCreationForTesting(overrides); |
| |
| // 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels. |
| float buffer[960]; |
| float* channel_pointers[] = {&buffer[0], &buffer[480]}; |
| Random random_generator(2341U); |
| constexpr int kFramesToProcessPerConfiguration = 3; |
| StreamConfig stream_config(/*sample_rate_hz=*/16000, |
| /*num_channels=*/1); |
| |
| AudioProcessing::Config config = apm->GetConfig(); |
| config.transient_suppression.enabled = true; |
| apm->ApplyConfig(config); |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| } |
| |
| config = apm->GetConfig(); |
| config.transient_suppression.enabled = false; |
| apm->ApplyConfig(config); |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| } |
| |
| config = apm->GetConfig(); |
| config.transient_suppression.enabled = true; |
| apm->ApplyConfig(config); |
| for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) { |
| RandomizeSampleVector(&random_generator, buffer); |
| EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| } |
| } |
| |
| class StartupInputVolumeParameterizedTest |
| : public ::testing::TestWithParam<int> {}; |
| |
| // Tests that, when no input volume controller is used, the startup input volume |
| // is never modified. |
| TEST_P(StartupInputVolumeParameterizedTest, |
| WithNoInputVolumeControllerStartupVolumeNotModified) { |
| webrtc::AudioProcessing::Config config; |
| config.gain_controller1.enabled = false; |
| config.gain_controller2.enabled = false; |
| auto apm = AudioProcessingBuilder().SetConfig(config).Create(); |
| |
| int startup_volume = GetParam(); |
| int recommended_volume = ProcessInputVolume( |
| *apm, /*num_frames=*/1, /*initial_volume=*/startup_volume); |
| EXPECT_EQ(recommended_volume, startup_volume); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest, |
| StartupInputVolumeParameterizedTest, |
| ::testing::Values(0, 5, 15, 50, 100)); |
| |
| // Tests that, when no input volume controller is used, the recommended input |
| // volume always matches the applied one. |
| TEST(AudioProcessingImplTest, |
| WithNoInputVolumeControllerAppliedAndRecommendedVolumesMatch) { |
| webrtc::AudioProcessing::Config config; |
| config.gain_controller1.enabled = false; |
| config.gain_controller2.enabled = false; |
| auto apm = AudioProcessingBuilder().SetConfig(config).Create(); |
| |
| Random rand_gen(42); |
| for (int i = 0; i < 32; ++i) { |
| SCOPED_TRACE(i); |
| int32_t applied_volume = rand_gen.Rand(/*low=*/0, /*high=*/255); |
| int recommended_volume = |
| ProcessInputVolume(*apm, /*num_frames=*/1, applied_volume); |
| EXPECT_EQ(recommended_volume, applied_volume); |
| } |
| } |
| |
| class ApmInputVolumeControllerParametrizedTest |
| : public ::testing::TestWithParam< |
| std::tuple<int, int, AudioProcessing::Config>> { |
| protected: |
| ApmInputVolumeControllerParametrizedTest() |
| : sample_rate_hz_(std::get<0>(GetParam())), |
| num_channels_(std::get<1>(GetParam())), |
| channels_(num_channels_), |
| channel_pointers_(num_channels_) { |
| const int frame_size = sample_rate_hz_ / 100; |
| for (int c = 0; c < num_channels_; ++c) { |
| channels_[c].resize(frame_size); |
| channel_pointers_[c] = channels_[c].data(); |
| std::fill(channels_[c].begin(), channels_[c].end(), 0.0f); |
| } |
| } |
| |
| int sample_rate_hz() const { return sample_rate_hz_; } |
| int num_channels() const { return num_channels_; } |
| AudioProcessing::Config GetConfig() const { return std::get<2>(GetParam()); } |
| |
| float* const* channel_pointers() { return channel_pointers_.data(); } |
| |
| private: |
| const int sample_rate_hz_; |
| const int num_channels_; |
| std::vector<std::vector<float>> channels_; |
| std::vector<float*> channel_pointers_; |
| }; |
| |
| TEST_P(ApmInputVolumeControllerParametrizedTest, |
| EnforceMinInputVolumeAtStartupWithZeroVolume) { |
| const StreamConfig stream_config(sample_rate_hz(), num_channels()); |
| auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create(); |
| |
| apm->set_stream_analog_level(0); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| EXPECT_GT(apm->recommended_stream_analog_level(), 0); |
| } |
| |
| TEST_P(ApmInputVolumeControllerParametrizedTest, |
| EnforceMinInputVolumeAtStartupWithNonZeroVolume) { |
| const StreamConfig stream_config(sample_rate_hz(), num_channels()); |
| auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create(); |
| |
| constexpr int kStartupVolume = 3; |
| apm->set_stream_analog_level(kStartupVolume); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| EXPECT_GT(apm->recommended_stream_analog_level(), kStartupVolume); |
| } |
| |
| TEST_P(ApmInputVolumeControllerParametrizedTest, |
| EnforceMinInputVolumeAfterManualVolumeAdjustment) { |
| const auto config = GetConfig(); |
| if (config.gain_controller1.enabled) { |
| // After a downward manual adjustment, AGC1 slowly converges to the minimum |
| // input volume. |
| GTEST_SKIP() << "Does not apply to AGC1"; |
| } |
| const StreamConfig stream_config(sample_rate_hz(), num_channels()); |
| auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create(); |
| |
| apm->set_stream_analog_level(20); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| constexpr int kManuallyAdjustedVolume = 3; |
| apm->set_stream_analog_level(kManuallyAdjustedVolume); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| EXPECT_GT(apm->recommended_stream_analog_level(), kManuallyAdjustedVolume); |
| } |
| |
| TEST_P(ApmInputVolumeControllerParametrizedTest, |
| DoNotEnforceMinInputVolumeAtStartupWithHighVolume) { |
| const StreamConfig stream_config(sample_rate_hz(), num_channels()); |
| auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create(); |
| |
| constexpr int kStartupVolume = 200; |
| apm->set_stream_analog_level(kStartupVolume); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| EXPECT_EQ(apm->recommended_stream_analog_level(), kStartupVolume); |
| } |
| |
| TEST_P(ApmInputVolumeControllerParametrizedTest, |
| DoNotEnforceMinInputVolumeAfterManualVolumeAdjustmentToZero) { |
| const StreamConfig stream_config(sample_rate_hz(), num_channels()); |
| auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create(); |
| |
| apm->set_stream_analog_level(100); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| apm->set_stream_analog_level(0); |
| apm->ProcessStream(channel_pointers(), stream_config, stream_config, |
| channel_pointers()); |
| EXPECT_EQ(apm->recommended_stream_analog_level(), 0); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P( |
| AudioProcessingImplTest, |
| ApmInputVolumeControllerParametrizedTest, |
| ::testing::Combine( |
| ::testing::Values(8000, 16000, 32000, 48000), // Sample rates. |
| ::testing::Values(1, 2), // Number of channels. |
| ::testing::Values( |
| // Full AGC1. |
| AudioProcessing::Config{ |
| .gain_controller1 = {.enabled = true, |
| .analog_gain_controller = |
| {.enabled = true, |
| .enable_digital_adaptive = true}}, |
| .gain_controller2 = {.enabled = false}}, |
| // Hybrid AGC. |
| AudioProcessing::Config{ |
| .gain_controller1 = {.enabled = true, |
| .analog_gain_controller = |
| {.enabled = true, |
| .enable_digital_adaptive = false}}, |
| .gain_controller2 = {.enabled = true, |
| .adaptive_digital = {.enabled = true}}}))); |
| |
| // When the input volume is not emulated and no input volume controller is |
| // active, the recommended volume must always be the applied volume. |
| TEST(AudioProcessingImplTest, |
| RecommendAppliedInputVolumeWithNoAgcWithNoEmulation) { |
| auto apm = AudioProcessingBuilder() |
| .SetConfig({.capture_level_adjustment = {.enabled = false}, |
| .gain_controller1 = {.enabled = false}}) |
| .Create(); |
| |
| constexpr int kOneFrame = 1; |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123); |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59); |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135); |
| } |
| |
| // When the input volume is emulated, the recommended volume must always be the |
| // applied volume and at any time it must not be that set in the input volume |
| // emulator. |
| // TODO(bugs.webrtc.org/14581): Enable when APM fixed to let this test pass. |
| TEST(AudioProcessingImplTest, |
| DISABLED_RecommendAppliedInputVolumeWithNoAgcWithEmulation) { |
| auto apm = |
| AudioProcessingBuilder() |
| .SetConfig({.capture_level_adjustment = {.enabled = true, |
| .analog_mic_gain_emulation{ |
| .enabled = true, |
| .initial_level = 255}}, |
| .gain_controller1 = {.enabled = false}}) |
| .Create(); |
| |
| constexpr int kOneFrame = 1; |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123); |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59); |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135); |
| } |
| |
| // Even if there is an enabled input volume controller, when the input volume is |
| // emulated, the recommended volume is always the applied volume because the |
| // active controller must only adjust the internally emulated volume and leave |
| // the externally applied volume unchanged. |
| // TODO(bugs.webrtc.org/14581): Enable when APM fixed to let this test pass. |
| TEST(AudioProcessingImplTest, |
| DISABLED_RecommendAppliedInputVolumeWithAgcWithEmulation) { |
| auto apm = |
| AudioProcessingBuilder() |
| .SetConfig({.capture_level_adjustment = {.enabled = true, |
| .analog_mic_gain_emulation{ |
| .enabled = true}}, |
| .gain_controller1 = {.enabled = true, |
| .analog_gain_controller{ |
| .enabled = true, |
| }}}) |
| .Create(); |
| |
| constexpr int kOneFrame = 1; |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123); |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59); |
| EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| Agc2FieldTrialDoNotSwitchToFullAgc2WhenNoAgcIsActive) { |
| constexpr AudioProcessing::Config kOriginal{ |
| .gain_controller1{.enabled = false}, |
| .gain_controller2{.enabled = false}, |
| }; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| Agc2FieldTrialDoNotSwitchToFullAgc2WithAgc1Agc2InputVolumeControllers) { |
| constexpr AudioProcessing::Config kOriginal{ |
| .gain_controller1{.enabled = true, |
| .analog_gain_controller{.enabled = true}}, |
| .gain_controller2{.enabled = true, |
| .input_volume_controller{.enabled = true}}, |
| }; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2); |
| } |
| |
| class Agc2FieldTrialParametrizedTest |
| : public ::testing::TestWithParam<AudioProcessing::Config> {}; |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfDisabled) { |
| const AudioProcessing::Config original = GetParam(); |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Disabled/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(original).Create()->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(original); |
| adjusted = apm->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2); |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfNoOverride) { |
| const AudioProcessing::Config original = GetParam(); |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "switch_to_agc2:false," |
| "disallow_transient_suppressor_usage:false/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(original).Create()->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(original); |
| adjusted = apm->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2); |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, DoNotSwitchToFullAgc2) { |
| const AudioProcessing::Config original = GetParam(); |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:false/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(original).Create()->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(original); |
| adjusted = apm->GetConfig(); |
| EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1); |
| EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2); |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, SwitchToFullAgc2) { |
| const AudioProcessing::Config original = GetParam(); |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(original).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.gain_controller1.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(original); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(adjusted.gain_controller1.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled); |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, |
| SwitchToFullAgc2AndOverrideInputVolumeControllerParameters) { |
| const AudioProcessing::Config original = GetParam(); |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true," |
| "min_input_volume:123," |
| "clipped_level_min:20," |
| "clipped_level_step:30," |
| "clipped_ratio_threshold:0.4," |
| "clipped_wait_frames:50," |
| "enable_clipping_predictor:true," |
| "target_range_max_dbfs:-6," |
| "target_range_min_dbfs:-70," |
| "update_input_volume_wait_frames:80," |
| "speech_probability_threshold:0.9," |
| "speech_ratio_threshold:1.0/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(original).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.gain_controller1.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(original); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(adjusted.gain_controller1.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled); |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, |
| SwitchToFullAgc2AndOverrideAdaptiveDigitalControllerParameters) { |
| const AudioProcessing::Config original = GetParam(); |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true," |
| "headroom_db:10," |
| "max_gain_db:20," |
| "initial_gain_db:7," |
| "max_gain_change_db_per_second:5," |
| "max_output_noise_level_dbfs:-40/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(original).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.gain_controller1.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled); |
| ASSERT_NE(adjusted.gain_controller2.adaptive_digital, |
| original.gain_controller2.adaptive_digital); |
| EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10); |
| EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20); |
| EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7); |
| EXPECT_EQ( |
| adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second, |
| 5); |
| EXPECT_EQ( |
| adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs, |
| -40); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(original); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(adjusted.gain_controller1.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled); |
| EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled); |
| ASSERT_NE(adjusted.gain_controller2.adaptive_digital, |
| original.gain_controller2.adaptive_digital); |
| EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10); |
| EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20); |
| EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7); |
| EXPECT_EQ( |
| adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second, |
| 5); |
| EXPECT_EQ( |
| adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs, |
| -40); |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithTs) { |
| AudioProcessing::Config config = GetParam(); |
| if (!config.transient_suppression.enabled) { |
| GTEST_SKIP() << "TS is disabled, skip."; |
| } |
| |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Disabled/"); |
| auto apm = AudioProcessingBuilder().SetConfig(config).Create(); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcess = 10; |
| int volume = 100; |
| for (int i = 0; i < kFramesToProcess; ++i) { |
| SCOPED_TRACE(i); |
| RandomizeSampleVector(&random_generator, buffer); |
| apm->set_stream_analog_level(volume); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| volume = apm->recommended_stream_analog_level(); |
| } |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithoutTs) { |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "switch_to_agc2:false," |
| "disallow_transient_suppressor_usage:true/"); |
| auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create(); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcess = 10; |
| int volume = 100; |
| for (int i = 0; i < kFramesToProcess; ++i) { |
| SCOPED_TRACE(i); |
| RandomizeSampleVector(&random_generator, buffer); |
| apm->set_stream_analog_level(volume); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| volume = apm->recommended_stream_analog_level(); |
| } |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, |
| ProcessSucceedsWhenSwitchToFullAgc2WithTs) { |
| AudioProcessing::Config config = GetParam(); |
| if (!config.transient_suppression.enabled) { |
| GTEST_SKIP() << "TS is disabled, skip."; |
| } |
| |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "switch_to_agc2:true," |
| "disallow_transient_suppressor_usage:false/"); |
| auto apm = AudioProcessingBuilder().SetConfig(config).Create(); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcess = 10; |
| int volume = 100; |
| for (int i = 0; i < kFramesToProcess; ++i) { |
| SCOPED_TRACE(i); |
| RandomizeSampleVector(&random_generator, buffer); |
| apm->set_stream_analog_level(volume); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| volume = apm->recommended_stream_analog_level(); |
| } |
| } |
| |
| TEST_P(Agc2FieldTrialParametrizedTest, |
| ProcessSucceedsWhenSwitchToFullAgc2WithoutTs) { |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "switch_to_agc2:true," |
| "disallow_transient_suppressor_usage:true/"); |
| auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create(); |
| |
| constexpr int kSampleRateHz = 48000; |
| constexpr int kNumChannels = 1; |
| std::array<float, kSampleRateHz / 100> buffer; |
| float* channel_pointers[] = {buffer.data()}; |
| StreamConfig stream_config(kSampleRateHz, kNumChannels); |
| Random random_generator(2341U); |
| constexpr int kFramesToProcess = 10; |
| int volume = 100; |
| for (int i = 0; i < kFramesToProcess; ++i) { |
| SCOPED_TRACE(i); |
| RandomizeSampleVector(&random_generator, buffer); |
| apm->set_stream_analog_level(volume); |
| ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config, |
| channel_pointers), |
| kNoErr); |
| volume = apm->recommended_stream_analog_level(); |
| } |
| } |
| |
| INSTANTIATE_TEST_SUITE_P( |
| AudioProcessingImplTest, |
| Agc2FieldTrialParametrizedTest, |
| ::testing::Values( |
| // Full AGC1, TS disabled. |
| AudioProcessing::Config{ |
| .transient_suppression = {.enabled = false}, |
| .gain_controller1 = |
| {.enabled = true, |
| .analog_gain_controller = {.enabled = true, |
| .enable_digital_adaptive = true}}, |
| .gain_controller2 = {.enabled = false}}, |
| // Full AGC1, TS enabled. |
| AudioProcessing::Config{ |
| .transient_suppression = {.enabled = true}, |
| .gain_controller1 = |
| {.enabled = true, |
| .analog_gain_controller = {.enabled = true, |
| .enable_digital_adaptive = true}}, |
| .gain_controller2 = {.enabled = false}}, |
| // Hybrid AGC, TS disabled. |
| AudioProcessing::Config{ |
| .transient_suppression = {.enabled = false}, |
| .gain_controller1 = |
| {.enabled = true, |
| .analog_gain_controller = {.enabled = true, |
| .enable_digital_adaptive = false}}, |
| .gain_controller2 = {.enabled = true, |
| .adaptive_digital = {.enabled = true}}}, |
| // Hybrid AGC, TS enabled. |
| AudioProcessing::Config{ |
| .transient_suppression = {.enabled = true}, |
| .gain_controller1 = |
| {.enabled = true, |
| .analog_gain_controller = {.enabled = true, |
| .enable_digital_adaptive = false}}, |
| .gain_controller2 = {.enabled = true, |
| .adaptive_digital = {.enabled = true}}})); |
| |
| TEST(AudioProcessingImplTest, CanDisableTransientSuppressor) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = false}}; |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled); |
| } |
| |
| TEST(AudioProcessingImplTest, CanEnableTs) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = true}}; |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_TRUE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_TRUE(adjusted.transient_suppression.enabled); |
| } |
| |
| TEST(AudioProcessingImplTest, CanDisableTsWithAgc2FieldTrialDisabled) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = false}}; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Disabled/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled); |
| } |
| |
| TEST(AudioProcessingImplTest, CanEnableTsWithAgc2FieldTrialDisabled) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = true}}; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Disabled/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_TRUE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_TRUE(adjusted.transient_suppression.enabled); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| CanDisableTsWithAgc2FieldTrialEnabledAndUsageAllowed) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = false}}; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "disallow_transient_suppressor_usage:false/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(adjusted.transient_suppression.enabled); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| CanEnableTsWithAgc2FieldTrialEnabledAndUsageAllowed) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = true}}; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "disallow_transient_suppressor_usage:false/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_TRUE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_TRUE(adjusted.transient_suppression.enabled); |
| } |
| |
| TEST(AudioProcessingImplTest, |
| CannotEnableTsWithAgc2FieldTrialEnabledAndUsageDisallowed) { |
| constexpr AudioProcessing::Config kOriginal = { |
| .transient_suppression = {.enabled = true}}; |
| webrtc::test::ScopedFieldTrials field_trials( |
| "WebRTC-Audio-GainController2/Enabled," |
| "disallow_transient_suppressor_usage:true/"); |
| |
| // Test config application via `AudioProcessing` ctor. |
| auto adjusted = |
| AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig(); |
| EXPECT_FALSE(adjusted.transient_suppression.enabled); |
| |
| // Test config application via `AudioProcessing::ApplyConfig()`. |
| auto apm = AudioProcessingBuilder().Create(); |
| apm->ApplyConfig(kOriginal); |
| adjusted = apm->GetConfig(); |
| EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled); |
| } |
| |
| } // namespace webrtc |
