| /* |
| * Copyright (c) 2017 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/gain_controller2.h" |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <memory> |
| #include <numeric> |
| #include <tuple> |
| |
| #include "api/array_view.h" |
| #include "modules/audio_processing/agc2/agc2_testing_common.h" |
| #include "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/test/audio_buffer_tools.h" |
| #include "modules/audio_processing/test/bitexactness_tools.h" |
| #include "rtc_base/checks.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| using ::testing::Eq; |
| using ::testing::Optional; |
| |
| using Agc2Config = AudioProcessing::Config::GainController2; |
| using InputVolumeControllerConfig = InputVolumeController::Config; |
| |
| // Sets all the samples in `ab` to `value`. |
| void SetAudioBufferSamples(float value, AudioBuffer& ab) { |
| for (size_t k = 0; k < ab.num_channels(); ++k) { |
| std::fill(ab.channels()[k], ab.channels()[k] + ab.num_frames(), value); |
| } |
| } |
| |
| float RunAgc2WithConstantInput(GainController2& agc2, |
| float input_level, |
| int num_frames, |
| int sample_rate_hz, |
| int num_channels = 1, |
| int applied_initial_volume = 0) { |
| const int num_samples = rtc::CheckedDivExact(sample_rate_hz, 100); |
| AudioBuffer ab(sample_rate_hz, num_channels, sample_rate_hz, num_channels, |
| sample_rate_hz, num_channels); |
| |
| // Give time to the level estimator to converge. |
| for (int i = 0; i < num_frames + 1; ++i) { |
| SetAudioBufferSamples(input_level, ab); |
| const auto applied_volume = agc2.recommended_input_volume(); |
| agc2.Analyze(applied_volume.value_or(applied_initial_volume), ab); |
| |
| agc2.Process(/*speech_probability=*/absl::nullopt, |
| /*input_volume_changed=*/false, &ab); |
| } |
| |
| // Return the last sample from the last processed frame. |
| return ab.channels()[0][num_samples - 1]; |
| } |
| |
| std::unique_ptr<GainController2> CreateAgc2FixedDigitalMode( |
| float fixed_gain_db, |
| int sample_rate_hz) { |
| Agc2Config config; |
| config.adaptive_digital.enabled = false; |
| config.fixed_digital.gain_db = fixed_gain_db; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| return std::make_unique<GainController2>( |
| config, InputVolumeControllerConfig{}, sample_rate_hz, |
| /*num_channels=*/1, |
| /*use_internal_vad=*/true); |
| } |
| |
| constexpr InputVolumeControllerConfig kTestInputVolumeControllerConfig{ |
| .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 = 100, |
| .speech_probability_threshold = 0.9, |
| .speech_ratio_threshold = 1, |
| }; |
| |
| } // namespace |
| |
| TEST(GainController2, CheckDefaultConfig) { |
| Agc2Config config; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, CheckFixedDigitalConfig) { |
| Agc2Config config; |
| // Attenuation is not allowed. |
| config.fixed_digital.gain_db = -5.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| // No gain is allowed. |
| config.fixed_digital.gain_db = 0.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| // Positive gain is allowed. |
| config.fixed_digital.gain_db = 15.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, CheckHeadroomDb) { |
| Agc2Config config; |
| config.adaptive_digital.headroom_db = -1.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.headroom_db = 0.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| config.adaptive_digital.headroom_db = 5.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, CheckMaxGainDb) { |
| Agc2Config config; |
| config.adaptive_digital.max_gain_db = -1.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.max_gain_db = 0.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.max_gain_db = 5.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, CheckInitialGainDb) { |
| Agc2Config config; |
| config.adaptive_digital.initial_gain_db = -1.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.initial_gain_db = 0.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| config.adaptive_digital.initial_gain_db = 5.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, CheckAdaptiveDigitalMaxGainChangeSpeedConfig) { |
| Agc2Config config; |
| config.adaptive_digital.max_gain_change_db_per_second = -1.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.max_gain_change_db_per_second = 0.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.max_gain_change_db_per_second = 5.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, CheckAdaptiveDigitalMaxOutputNoiseLevelConfig) { |
| Agc2Config config; |
| config.adaptive_digital.max_output_noise_level_dbfs = 5.0f; |
| EXPECT_FALSE(GainController2::Validate(config)); |
| config.adaptive_digital.max_output_noise_level_dbfs = 0.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| config.adaptive_digital.max_output_noise_level_dbfs = -5.0f; |
| EXPECT_TRUE(GainController2::Validate(config)); |
| } |
| |
| TEST(GainController2, |
| CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabled) { |
| constexpr float kHighInputLevel = 32767.0f; |
| constexpr float kLowInputLevel = 1000.0f; |
| constexpr int kInitialInputVolume = 100; |
| constexpr int kNumChannels = 2; |
| constexpr int kNumFrames = 5; |
| constexpr int kSampleRateHz = 16000; |
| |
| Agc2Config config; |
| config.input_volume_controller.enabled = false; |
| |
| auto gain_controller = std::make_unique<GainController2>( |
| config, InputVolumeControllerConfig{}, kSampleRateHz, kNumChannels, |
| /*use_internal_vad=*/true); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with no clipping or detected speech. |
| RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with clipping. |
| RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| } |
| |
| TEST( |
| GainController2, |
| CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabledAndSpecificConfigUsed) { |
| constexpr float kHighInputLevel = 32767.0f; |
| constexpr float kLowInputLevel = 1000.0f; |
| constexpr int kInitialInputVolume = 100; |
| constexpr int kNumChannels = 2; |
| constexpr int kNumFrames = 5; |
| constexpr int kSampleRateHz = 16000; |
| |
| Agc2Config config; |
| config.input_volume_controller.enabled = false; |
| |
| auto gain_controller = std::make_unique<GainController2>( |
| config, kTestInputVolumeControllerConfig, kSampleRateHz, kNumChannels, |
| /*use_internal_vad=*/true); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with no clipping or detected speech. |
| RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with clipping. |
| RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| } |
| |
| TEST(GainController2, |
| CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabled) { |
| constexpr float kHighInputLevel = 32767.0f; |
| constexpr float kLowInputLevel = 1000.0f; |
| constexpr int kInitialInputVolume = 100; |
| constexpr int kNumChannels = 2; |
| constexpr int kNumFrames = 5; |
| constexpr int kSampleRateHz = 16000; |
| |
| Agc2Config config; |
| config.input_volume_controller.enabled = true; |
| config.adaptive_digital.enabled = true; |
| |
| auto gain_controller = std::make_unique<GainController2>( |
| config, InputVolumeControllerConfig{}, kSampleRateHz, kNumChannels, |
| /*use_internal_vad=*/true); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with no clipping or detected speech. |
| RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with clipping. |
| RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); |
| } |
| |
| TEST( |
| GainController2, |
| CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabledAndSpecificConfigUsed) { |
| constexpr float kHighInputLevel = 32767.0f; |
| constexpr float kLowInputLevel = 1000.0f; |
| constexpr int kInitialInputVolume = 100; |
| constexpr int kNumChannels = 2; |
| constexpr int kNumFrames = 5; |
| constexpr int kSampleRateHz = 16000; |
| |
| Agc2Config config; |
| config.input_volume_controller.enabled = true; |
| config.adaptive_digital.enabled = true; |
| |
| auto gain_controller = std::make_unique<GainController2>( |
| config, kTestInputVolumeControllerConfig, kSampleRateHz, kNumChannels, |
| /*use_internal_vad=*/true); |
| |
| EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with no clipping or detected speech. |
| RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); |
| |
| // Run AGC for a signal with clipping. |
| RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, |
| kSampleRateHz, kNumChannels, kInitialInputVolume); |
| |
| EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); |
| } |
| |
| // Checks that the default config is applied. |
| TEST(GainController2, ApplyDefaultConfig) { |
| auto gain_controller2 = std::make_unique<GainController2>( |
| Agc2Config{}, InputVolumeControllerConfig{}, |
| /*sample_rate_hz=*/16000, /*num_channels=*/2, |
| /*use_internal_vad=*/true); |
| EXPECT_TRUE(gain_controller2.get()); |
| } |
| |
| TEST(GainController2FixedDigital, GainShouldChangeOnSetGain) { |
| constexpr float kInputLevel = 1000.0f; |
| constexpr size_t kNumFrames = 5; |
| constexpr size_t kSampleRateHz = 8000; |
| constexpr float kGain0Db = 0.0f; |
| constexpr float kGain20Db = 20.0f; |
| |
| auto agc2_fixed = CreateAgc2FixedDigitalMode(kGain0Db, kSampleRateHz); |
| |
| // Signal level is unchanged with 0 db gain. |
| EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, kNumFrames, |
| kSampleRateHz), |
| kInputLevel); |
| |
| // +20 db should increase signal by a factor of 10. |
| agc2_fixed->SetFixedGainDb(kGain20Db); |
| EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, kNumFrames, |
| kSampleRateHz), |
| kInputLevel * 10); |
| } |
| |
| TEST(GainController2FixedDigital, ChangeFixedGainShouldBeFastAndTimeInvariant) { |
| // Number of frames required for the fixed gain controller to adapt on the |
| // input signal when the gain changes. |
| constexpr size_t kNumFrames = 5; |
| |
| constexpr float kInputLevel = 1000.0f; |
| constexpr size_t kSampleRateHz = 8000; |
| constexpr float kGainDbLow = 0.0f; |
| constexpr float kGainDbHigh = 25.0f; |
| static_assert(kGainDbLow < kGainDbHigh, ""); |
| |
| auto agc2_fixed = CreateAgc2FixedDigitalMode(kGainDbLow, kSampleRateHz); |
| |
| // Start with a lower gain. |
| const float output_level_pre = RunAgc2WithConstantInput( |
| *agc2_fixed, kInputLevel, kNumFrames, kSampleRateHz); |
| |
| // Increase gain. |
| agc2_fixed->SetFixedGainDb(kGainDbHigh); |
| static_cast<void>(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, |
| kNumFrames, kSampleRateHz)); |
| |
| // Back to the lower gain. |
| agc2_fixed->SetFixedGainDb(kGainDbLow); |
| const float output_level_post = RunAgc2WithConstantInput( |
| *agc2_fixed, kInputLevel, kNumFrames, kSampleRateHz); |
| |
| EXPECT_EQ(output_level_pre, output_level_post); |
| } |
| |
| class FixedDigitalTest |
| : public ::testing::TestWithParam<std::tuple<float, float, int, bool>> { |
| protected: |
| float gain_db_min() const { return std::get<0>(GetParam()); } |
| float gain_db_max() const { return std::get<1>(GetParam()); } |
| int sample_rate_hz() const { return std::get<2>(GetParam()); } |
| bool saturation_expected() const { return std::get<3>(GetParam()); } |
| }; |
| |
| TEST_P(FixedDigitalTest, CheckSaturationBehaviorWithLimiter) { |
| for (const float gain_db : test::LinSpace(gain_db_min(), gain_db_max(), 10)) { |
| SCOPED_TRACE(gain_db); |
| auto agc2_fixed = CreateAgc2FixedDigitalMode(gain_db, sample_rate_hz()); |
| const float processed_sample = |
| RunAgc2WithConstantInput(*agc2_fixed, /*input_level=*/32767.0f, |
| /*num_frames=*/5, sample_rate_hz()); |
| if (saturation_expected()) { |
| EXPECT_FLOAT_EQ(processed_sample, 32767.0f); |
| } else { |
| EXPECT_LT(processed_sample, 32767.0f); |
| } |
| } |
| } |
| |
| static_assert(test::kLimiterMaxInputLevelDbFs < 10, ""); |
| INSTANTIATE_TEST_SUITE_P( |
| GainController2, |
| FixedDigitalTest, |
| ::testing::Values( |
| // When gain < `test::kLimiterMaxInputLevelDbFs`, the limiter will not |
| // saturate the signal (at any sample rate). |
| std::make_tuple(0.1f, |
| test::kLimiterMaxInputLevelDbFs - 0.01f, |
| 8000, |
| false), |
| std::make_tuple(0.1, |
| test::kLimiterMaxInputLevelDbFs - 0.01f, |
| 48000, |
| false), |
| // When gain > `test::kLimiterMaxInputLevelDbFs`, the limiter will |
| // saturate the signal (at any sample rate). |
| std::make_tuple(test::kLimiterMaxInputLevelDbFs + 0.01f, |
| 10.0f, |
| 8000, |
| true), |
| std::make_tuple(test::kLimiterMaxInputLevelDbFs + 0.01f, |
| 10.0f, |
| 48000, |
| true))); |
| |
| // Processes a test audio file and checks that the gain applied at the end of |
| // the recording is close to the expected value. |
| TEST(GainController2, CheckFinalGainWithAdaptiveDigitalController) { |
| constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; |
| constexpr int kStereo = 2; |
| |
| // Create AGC2 enabling only the adaptive digital controller. |
| Agc2Config config; |
| config.fixed_digital.gain_db = 0.0f; |
| config.adaptive_digital.enabled = true; |
| GainController2 agc2(config, /*input_volume_controller_config=*/{}, |
| kSampleRateHz, kStereo, |
| /*use_internal_vad=*/true); |
| |
| test::InputAudioFile input_file( |
| test::GetApmCaptureTestVectorFileName(kSampleRateHz), |
| /*loop_at_end=*/true); |
| const StreamConfig stream_config(kSampleRateHz, kStereo); |
| |
| // Init buffers. |
| constexpr int kFrameDurationMs = 10; |
| std::vector<float> frame(kStereo * stream_config.num_frames()); |
| AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, |
| kSampleRateHz, kStereo); |
| |
| // Simulate. |
| constexpr float kGainDb = -6.0f; |
| const float gain = std::pow(10.0f, kGainDb / 20.0f); |
| constexpr int kDurationMs = 10000; |
| constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs; |
| for (int i = 0; i < kNumFramesToProcess; ++i) { |
| ReadFloatSamplesFromStereoFile(stream_config.num_frames(), |
| stream_config.num_channels(), &input_file, |
| frame); |
| // Apply a fixed gain to the input audio. |
| for (float& x : frame) { |
| x *= gain; |
| } |
| test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer); |
| agc2.Process(/*speech_probability=*/absl::nullopt, |
| /*input_volume_changed=*/false, &audio_buffer); |
| } |
| |
| // Estimate the applied gain by processing a probing frame. |
| SetAudioBufferSamples(/*value=*/1.0f, audio_buffer); |
| agc2.Process(/*speech_probability=*/absl::nullopt, |
| /*input_volume_changed=*/false, &audio_buffer); |
| const float applied_gain_db = |
| 20.0f * std::log10(audio_buffer.channels_const()[0][0]); |
| |
| constexpr float kExpectedGainDb = 7.0f; |
| constexpr float kToleranceDb = 0.3f; |
| EXPECT_NEAR(applied_gain_db, kExpectedGainDb, kToleranceDb); |
| } |
| |
| #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
| // Checks that `GainController2` crashes in debug mode if it runs its internal |
| // VAD and the speech probability values are provided by the caller. |
| TEST(GainController2DeathTest, |
| DebugCrashIfUseInternalVadAndSpeechProbabilityGiven) { |
| constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; |
| constexpr int kStereo = 2; |
| AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, |
| kSampleRateHz, kStereo); |
| // Create AGC2 so that the interval VAD is also created. |
| GainController2 agc2(/*config=*/{.adaptive_digital = {.enabled = true}}, |
| /*input_volume_controller_config=*/{}, kSampleRateHz, |
| kStereo, |
| /*use_internal_vad=*/true); |
| |
| EXPECT_DEATH(agc2.Process(/*speech_probability=*/0.123f, |
| /*input_volume_changed=*/false, &audio_buffer), |
| ""); |
| } |
| #endif |
| |
| // Processes a test audio file and checks that the injected speech probability |
| // is not ignored when the internal VAD is not used. |
| TEST(GainController2, |
| CheckInjectedVadProbabilityUsedWithAdaptiveDigitalController) { |
| constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; |
| constexpr int kStereo = 2; |
| |
| // Create AGC2 enabling only the adaptive digital controller. |
| Agc2Config config; |
| config.fixed_digital.gain_db = 0.0f; |
| config.adaptive_digital.enabled = true; |
| GainController2 agc2(config, /*input_volume_controller_config=*/{}, |
| kSampleRateHz, kStereo, |
| /*use_internal_vad=*/false); |
| GainController2 agc2_reference(config, /*input_volume_controller_config=*/{}, |
| kSampleRateHz, kStereo, |
| /*use_internal_vad=*/true); |
| |
| test::InputAudioFile input_file( |
| test::GetApmCaptureTestVectorFileName(kSampleRateHz), |
| /*loop_at_end=*/true); |
| const StreamConfig stream_config(kSampleRateHz, kStereo); |
| |
| // Init buffers. |
| constexpr int kFrameDurationMs = 10; |
| std::vector<float> frame(kStereo * stream_config.num_frames()); |
| AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, |
| kSampleRateHz, kStereo); |
| AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz, |
| kStereo, kSampleRateHz, kStereo); |
| // Simulate. |
| constexpr float kGainDb = -6.0f; |
| const float gain = std::pow(10.0f, kGainDb / 20.0f); |
| constexpr int kDurationMs = 10000; |
| constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs; |
| constexpr float kSpeechProbabilities[] = {1.0f, 0.3f}; |
| constexpr float kEpsilon = 0.0001f; |
| bool all_samples_zero = true; |
| bool all_samples_equal = true; |
| for (int i = 0, j = 0; i < kNumFramesToProcess; ++i, j = 1 - j) { |
| ReadFloatSamplesFromStereoFile(stream_config.num_frames(), |
| stream_config.num_channels(), &input_file, |
| frame); |
| // Apply a fixed gain to the input audio. |
| for (float& x : frame) { |
| x *= gain; |
| } |
| test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer); |
| agc2.Process(kSpeechProbabilities[j], /*input_volume_changed=*/false, |
| &audio_buffer); |
| test::CopyVectorToAudioBuffer(stream_config, frame, |
| &audio_buffer_reference); |
| agc2_reference.Process(/*speech_probability=*/absl::nullopt, |
| /*input_volume_changed=*/false, |
| &audio_buffer_reference); |
| // Check the output buffers. |
| for (int i = 0; i < kStereo; ++i) { |
| for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) { |
| all_samples_zero &= |
| fabs(audio_buffer.channels_const()[i][j]) < kEpsilon; |
| all_samples_equal &= |
| fabs(audio_buffer.channels_const()[i][j] - |
| audio_buffer_reference.channels_const()[i][j]) < kEpsilon; |
| } |
| } |
| } |
| EXPECT_FALSE(all_samples_zero); |
| EXPECT_FALSE(all_samples_equal); |
| } |
| |
| // Processes a test audio file and checks that the output is equal when |
| // an injected speech probability from `VoiceActivityDetectorWrapper` and |
| // the speech probability computed by the internal VAD are the same. |
| TEST(GainController2, |
| CheckEqualResultFromInjectedVadProbabilityWithAdaptiveDigitalController) { |
| constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; |
| constexpr int kStereo = 2; |
| |
| // Create AGC2 enabling only the adaptive digital controller. |
| Agc2Config config; |
| config.fixed_digital.gain_db = 0.0f; |
| config.adaptive_digital.enabled = true; |
| GainController2 agc2(config, /*input_volume_controller_config=*/{}, |
| kSampleRateHz, kStereo, |
| /*use_internal_vad=*/false); |
| GainController2 agc2_reference(config, /*input_volume_controller_config=*/{}, |
| kSampleRateHz, kStereo, |
| /*use_internal_vad=*/true); |
| VoiceActivityDetectorWrapper vad(GetAvailableCpuFeatures(), kSampleRateHz); |
| test::InputAudioFile input_file( |
| test::GetApmCaptureTestVectorFileName(kSampleRateHz), |
| /*loop_at_end=*/true); |
| const StreamConfig stream_config(kSampleRateHz, kStereo); |
| |
| // Init buffers. |
| constexpr int kFrameDurationMs = 10; |
| std::vector<float> frame(kStereo * stream_config.num_frames()); |
| AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, |
| kSampleRateHz, kStereo); |
| AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz, |
| kStereo, kSampleRateHz, kStereo); |
| |
| // Simulate. |
| constexpr float kGainDb = -6.0f; |
| const float gain = std::pow(10.0f, kGainDb / 20.0f); |
| constexpr int kDurationMs = 10000; |
| constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs; |
| for (int i = 0; i < kNumFramesToProcess; ++i) { |
| ReadFloatSamplesFromStereoFile(stream_config.num_frames(), |
| stream_config.num_channels(), &input_file, |
| frame); |
| // Apply a fixed gain to the input audio. |
| for (float& x : frame) { |
| x *= gain; |
| } |
| test::CopyVectorToAudioBuffer(stream_config, frame, |
| &audio_buffer_reference); |
| agc2_reference.Process(absl::nullopt, /*input_volume_changed=*/false, |
| &audio_buffer_reference); |
| test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer); |
| float speech_probability = vad.Analyze(AudioFrameView<const float>( |
| audio_buffer.channels(), audio_buffer.num_channels(), |
| audio_buffer.num_frames())); |
| agc2.Process(speech_probability, /*input_volume_changed=*/false, |
| &audio_buffer); |
| // Check the output buffer. |
| for (int i = 0; i < kStereo; ++i) { |
| for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) { |
| EXPECT_FLOAT_EQ(audio_buffer.channels_const()[i][j], |
| audio_buffer_reference.channels_const()[i][j]); |
| } |
| } |
| } |
| } |
| |
| } // namespace test |
| } // namespace webrtc |
