| /* |
| * 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 <memory> |
| |
| #include "api/scoped_refptr.h" |
| #include "modules/audio_processing/include/audio_processing.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/ref_counted_object.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| using ::testing::Invoke; |
| using ::testing::NotNull; |
| |
| class MockInitialize : public AudioProcessingImpl { |
| public: |
| explicit MockInitialize(const webrtc::Config& config) |
| : AudioProcessingImpl(config) {} |
| |
| MOCK_METHOD0(InitializeLocked, int()); |
| int RealInitializeLocked() RTC_NO_THREAD_SAFETY_ANALYSIS { |
| return AudioProcessingImpl::InitializeLocked(); |
| } |
| |
| MOCK_CONST_METHOD0(AddRef, void()); |
| MOCK_CONST_METHOD0(Release, rtc::RefCountReleaseStatus()); |
| }; |
| |
| // 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) 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_; |
| }; |
| |
| void InitializeAudioFrame(size_t input_rate, |
| size_t num_channels, |
| AudioFrame* frame) { |
| const size_t samples_per_input_channel = rtc::CheckedDivExact( |
| input_rate, static_cast<size_t>(rtc::CheckedDivExact( |
| 1000, AudioProcessing::kChunkSizeMs))); |
| RTC_DCHECK_LE(samples_per_input_channel * num_channels, |
| AudioFrame::kMaxDataSizeSamples); |
| frame->samples_per_channel_ = samples_per_input_channel; |
| frame->sample_rate_hz_ = input_rate; |
| frame->num_channels_ = num_channels; |
| } |
| |
| void FillFixedFrame(int16_t audio_level, AudioFrame* frame) { |
| const size_t num_samples = frame->samples_per_channel_ * frame->num_channels_; |
| for (size_t i = 0; i < num_samples; ++i) { |
| frame->mutable_data()[i] = audio_level; |
| } |
| } |
| |
| // 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; } |
| }; |
| |
| } // namespace |
| |
| TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { |
| webrtc::Config config; |
| MockInitialize mock(config); |
| ON_CALL(mock, InitializeLocked()) |
| .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked)); |
| |
| EXPECT_CALL(mock, InitializeLocked()).Times(1); |
| mock.Initialize(); |
| |
| AudioFrame frame; |
| // Call with the default parameters; there should be an init. |
| frame.num_channels_ = 1; |
| SetFrameSampleRate(&frame, 16000); |
| EXPECT_CALL(mock, InitializeLocked()).Times(0); |
| EXPECT_NOERR(mock.ProcessStream(&frame)); |
| EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
| |
| // New sample rate. (Only impacts ProcessStream). |
| SetFrameSampleRate(&frame, 32000); |
| EXPECT_CALL(mock, InitializeLocked()).Times(1); |
| EXPECT_NOERR(mock.ProcessStream(&frame)); |
| |
| // New number of channels. |
| // TODO(peah): Investigate why this causes 2 inits. |
| frame.num_channels_ = 2; |
| EXPECT_CALL(mock, InitializeLocked()).Times(2); |
| EXPECT_NOERR(mock.ProcessStream(&frame)); |
| // ProcessStream sets num_channels_ == num_output_channels. |
| frame.num_channels_ = 2; |
| EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
| |
| // A new sample rate passed to ProcessReverseStream should cause an init. |
| SetFrameSampleRate(&frame, 16000); |
| EXPECT_CALL(mock, InitializeLocked()).Times(1); |
| EXPECT_NOERR(mock.ProcessReverseStream(&frame)); |
| } |
| |
| TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) { |
| std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().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); |
| |
| AudioFrame frame; |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame); |
| |
| FillFixedFrame(kAudioLevel, &frame); |
| apm->ProcessStream(&frame); |
| EXPECT_EQ(frame.data()[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) { |
| FillFixedFrame(kAudioLevel, &frame); |
| apm->ProcessStream(&frame); |
| } |
| EXPECT_EQ(frame.data()[100], kGainFactor * kAudioLevel) |
| << "Frame should be amplified."; |
| } |
| |
| 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(); |
| |
| std::unique_ptr<AudioProcessing> apm( |
| AudioProcessingBuilder() |
| .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); |
| |
| AudioFrame frame; |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame); |
| FillFixedFrame(kAudioLevel, &frame); |
| |
| 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(), /*echo_path_change=*/false)) |
| .Times(1); |
| apm->ProcessStream(&frame); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), /*echo_path_change=*/true)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f)); |
| apm->ProcessStream(&frame); |
| } |
| |
| 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(); |
| |
| std::unique_ptr<AudioProcessing> apm( |
| AudioProcessingBuilder() |
| .SetEchoControlFactory(std::move(echo_control_factory)) |
| .Create()); |
| webrtc::AudioProcessing::Config apm_config; |
| // Enable AGC1. |
| apm_config.gain_controller1.enabled = true; |
| apm_config.gain_controller1.mode = |
| AudioProcessing::Config::GainController1::kAdaptiveAnalog; |
| apm_config.gain_controller2.enabled = false; |
| apm_config.pre_amplifier.enabled = false; |
| apm->ApplyConfig(apm_config); |
| |
| AudioFrame frame; |
| constexpr int16_t kAudioLevel = 1000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame); |
| FillFixedFrame(kAudioLevel, &frame); |
| |
| MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext(); |
| |
| const int initial_analog_gain = apm->recommended_stream_analog_level(); |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), false)).Times(1); |
| apm->ProcessStream(&frame); |
| |
| // Force an analog gain change if it did not happen. |
| if (initial_analog_gain == apm->recommended_stream_analog_level()) { |
| apm->set_stream_analog_level(initial_analog_gain + 1); |
| } |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), true)).Times(1); |
| apm->ProcessStream(&frame); |
| } |
| |
| 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(); |
| |
| std::unique_ptr<AudioProcessing> apm( |
| AudioProcessingBuilder() |
| .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); |
| |
| AudioFrame frame; |
| constexpr int16_t kAudioLevel = 10000; |
| constexpr size_t kSampleRateHz = 48000; |
| constexpr size_t kNumChannels = 2; |
| InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame); |
| FillFixedFrame(kAudioLevel, &frame); |
| |
| 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(), /*echo_path_change=*/false)) |
| .Times(1); |
| apm->ProcessStream(&frame); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), /*echo_path_change=*/false)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50)); |
| apm->ProcessStream(&frame); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), /*echo_path_change=*/false)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50)); |
| apm->ProcessStream(&frame); |
| |
| EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1); |
| EXPECT_CALL(*echo_control_mock, |
| ProcessCapture(NotNull(), /*echo_path_change=*/true)) |
| .Times(1); |
| apm->SetRuntimeSetting( |
| AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(100)); |
| apm->ProcessStream(&frame); |
| } |
| |
| TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) { |
| // Make sure that signal changes caused by a render pre-processing sub-module |
| // take place before any echo detector analysis. |
| rtc::scoped_refptr<TestEchoDetector> test_echo_detector( |
| new rtc::RefCountedObject<TestEchoDetector>()); |
| std::unique_ptr<CustomProcessing> test_render_pre_processor( |
| new TestRenderPreProcessor()); |
| // Create APM injecting the test echo detector and render pre-processor. |
| std::unique_ptr<AudioProcessing> apm( |
| AudioProcessingBuilder() |
| .SetEchoDetector(test_echo_detector) |
| .SetRenderPreProcessing(std::move(test_render_pre_processor)) |
| .Create()); |
| webrtc::AudioProcessing::Config apm_config; |
| apm_config.pre_amplifier.enabled = true; |
| apm_config.residual_echo_detector.enabled = true; |
| apm->ApplyConfig(apm_config); |
| |
| constexpr int16_t kAudioLevel = 1000; |
| constexpr int kSampleRateHz = 16000; |
| constexpr size_t kNumChannels = 1; |
| AudioFrame frame; |
| InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame); |
| |
| constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel); |
| constexpr float kExpectedPreprocessedAudioLevel = |
| TestRenderPreProcessor::ProcessSample(kAudioLevelFloat); |
| ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel); |
| |
| // Analyze a render stream frame. |
| FillFixedFrame(kAudioLevel, &frame); |
| ASSERT_EQ(AudioProcessing::Error::kNoError, |
| apm->ProcessReverseStream(&frame)); |
| // Trigger a call to in EchoDetector::AnalyzeRenderAudio() via |
| // ProcessStream(). |
| FillFixedFrame(kAudioLevel, &frame); |
| ASSERT_EQ(AudioProcessing::Error::kNoError, apm->ProcessStream(&frame)); |
| // 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()); |
| } |
| |
| } // namespace webrtc |