diff --git a/api/audio_options.h b/api/audio_options.h index d62e1f8e9b..8d2880b0a0 100644 --- a/api/audio_options.h +++ b/api/audio_options.h @@ -43,7 +43,6 @@ struct AudioOptions { SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec); SetFrom(&experimental_ns, change.experimental_ns); SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer); - SetFrom(&level_control, change.level_control); SetFrom(&residual_echo_detector, change.residual_echo_detector); SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov); SetFrom(&tx_agc_digital_compression_gain, @@ -52,8 +51,6 @@ struct AudioOptions { SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe); SetFrom(&audio_network_adaptor, change.audio_network_adaptor); SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config); - SetFrom(&level_control_initial_peak_level_dbfs, - change.level_control_initial_peak_level_dbfs); } bool operator==(const AudioOptions& o) const { @@ -76,7 +73,6 @@ struct AudioOptions { delay_agnostic_aec == o.delay_agnostic_aec && experimental_ns == o.experimental_ns && intelligibility_enhancer == o.intelligibility_enhancer && - level_control == o.level_control && residual_echo_detector == o.residual_echo_detector && tx_agc_target_dbov == o.tx_agc_target_dbov && tx_agc_digital_compression_gain == @@ -84,9 +80,7 @@ struct AudioOptions { tx_agc_limiter == o.tx_agc_limiter && combined_audio_video_bwe == o.combined_audio_video_bwe && audio_network_adaptor == o.audio_network_adaptor && - audio_network_adaptor_config == o.audio_network_adaptor_config && - level_control_initial_peak_level_dbfs == - o.level_control_initial_peak_level_dbfs; + audio_network_adaptor_config == o.audio_network_adaptor_config; } bool operator!=(const AudioOptions& o) const { return !(*this == o); } @@ -113,9 +107,6 @@ struct AudioOptions { ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec); ost << ToStringIfSet("experimental_ns", experimental_ns); ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer); - ost << ToStringIfSet("level_control", level_control); - ost << ToStringIfSet("level_control_initial_peak_level_dbfs", - level_control_initial_peak_level_dbfs); ost << ToStringIfSet("residual_echo_detector", residual_echo_detector); ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov); ost << ToStringIfSet("tx_agc_digital_compression_gain", @@ -161,9 +152,6 @@ struct AudioOptions { rtc::Optional delay_agnostic_aec; rtc::Optional experimental_ns; rtc::Optional intelligibility_enhancer; - rtc::Optional level_control; - // Specifies an optional initialization value for the level controller. - rtc::Optional level_control_initial_peak_level_dbfs; // Note that tx_agc_* only applies to non-experimental AGC. rtc::Optional residual_echo_detector; rtc::Optional tx_agc_target_dbov; diff --git a/api/mediaconstraintsinterface.cc b/api/mediaconstraintsinterface.cc index 5e6b21823b..8358644407 100644 --- a/api/mediaconstraintsinterface.cc +++ b/api/mediaconstraintsinterface.cc @@ -107,9 +107,6 @@ const char MediaConstraintsInterface::kExperimentalNoiseSuppression[] = "googNoiseSuppression2"; const char MediaConstraintsInterface::kIntelligibilityEnhancer[] = "intelligibilityEnhancer"; -const char MediaConstraintsInterface::kLevelControl[] = "levelControl"; -const char MediaConstraintsInterface::kLevelControlInitialPeakLevelDBFS[] = - "levelControlInitialPeakLevelDBFS"; const char MediaConstraintsInterface::kHighpassFilter[] = "googHighpassFilter"; const char MediaConstraintsInterface::kTypingNoiseDetection[] = @@ -247,9 +244,6 @@ void CopyConstraintsIntoAudioOptions( ConstraintToOptional( constraints, MediaConstraintsInterface::kIntelligibilityEnhancer, &options->intelligibility_enhancer); - ConstraintToOptional(constraints, - MediaConstraintsInterface::kLevelControl, - &options->level_control); ConstraintToOptional(constraints, MediaConstraintsInterface::kHighpassFilter, &options->highpass_filter); @@ -259,9 +253,6 @@ void CopyConstraintsIntoAudioOptions( ConstraintToOptional(constraints, MediaConstraintsInterface::kAudioMirroring, &options->stereo_swapping); - ConstraintToOptional( - constraints, MediaConstraintsInterface::kLevelControlInitialPeakLevelDBFS, - &options->level_control_initial_peak_level_dbfs); ConstraintToOptional( constraints, MediaConstraintsInterface::kAudioNetworkAdaptorConfig, &options->audio_network_adaptor_config); diff --git a/api/mediaconstraintsinterface.h b/api/mediaconstraintsinterface.h index 73e4619bca..90661b893a 100644 --- a/api/mediaconstraintsinterface.h +++ b/api/mediaconstraintsinterface.h @@ -74,9 +74,6 @@ class MediaConstraintsInterface { static const char kNoiseSuppression[]; // googNoiseSuppression static const char kExperimentalNoiseSuppression[]; // googNoiseSuppression2 static const char kIntelligibilityEnhancer[]; // intelligibilityEnhancer - static const char kLevelControl[]; // levelControl - static const char - kLevelControlInitialPeakLevelDBFS[]; // levelControlInitialPeakLevelDBFS static const char kHighpassFilter[]; // googHighpassFilter static const char kTypingNoiseDetection[]; // googTypingNoiseDetection static const char kAudioMirroring[]; // googAudioMirroring diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc index 7d889f3178..6cd88054b6 100644 --- a/media/engine/webrtcvoiceengine.cc +++ b/media/engine/webrtcvoiceengine.cc @@ -295,7 +295,6 @@ void WebRtcVoiceEngine::Init() { options.delay_agnostic_aec = false; options.experimental_ns = false; options.intelligibility_enhancer = false; - options.level_control = false; options.residual_echo_detector = true; bool error = ApplyOptions(options); RTC_DCHECK(error); @@ -564,22 +563,8 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { new webrtc::Intelligibility(*intelligibility_enhancer_)); } - if (options.level_control) { - level_control_ = options.level_control; - } - webrtc::AudioProcessing::Config apm_config = apm()->GetConfig(); - RTC_LOG(LS_INFO) << "Level control: " - << (!!level_control_ ? *level_control_ : -1); - if (level_control_) { - apm_config.level_controller.enabled = *level_control_; - if (options.level_control_initial_peak_level_dbfs) { - apm_config.level_controller.initial_peak_level_dbfs = - *options.level_control_initial_peak_level_dbfs; - } - } - if (options.highpass_filter) { apm_config.high_pass_filter.enabled = *options.highpass_filter; } diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h index 0c7baf5970..fbf79533c4 100644 --- a/media/engine/webrtcvoiceengine.h +++ b/media/engine/webrtcvoiceengine.h @@ -120,7 +120,7 @@ class WebRtcVoiceEngine final { webrtc::AgcConfig default_agc_config_; // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns - // level controller, and intelligibility_enhancer values, and apply them + // and intelligibility_enhancer values, and apply them // in case they are missing in the audio options. We need to do this because // SetExtraOptions() will revert to defaults for options which are not // provided. @@ -128,7 +128,6 @@ class WebRtcVoiceEngine final { rtc::Optional delay_agnostic_aec_; rtc::Optional experimental_ns_; rtc::Optional intelligibility_enhancer_; - rtc::Optional level_control_; // Jitter buffer settings for new streams. size_t audio_jitter_buffer_max_packets_ = 50; bool audio_jitter_buffer_fast_accelerate_ = false; diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index 3dcea89d90..93d3ec67e5 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -79,27 +79,6 @@ rtc_static_library("audio_processing") { "include/audio_processing.h", "include/config.cc", "include/config.h", - "level_controller/biquad_filter.cc", - "level_controller/biquad_filter.h", - "level_controller/down_sampler.cc", - "level_controller/down_sampler.h", - "level_controller/gain_applier.cc", - "level_controller/gain_applier.h", - "level_controller/gain_selector.cc", - "level_controller/gain_selector.h", - "level_controller/level_controller.cc", - "level_controller/level_controller.h", - "level_controller/level_controller_constants.h", - "level_controller/noise_level_estimator.cc", - "level_controller/noise_level_estimator.h", - "level_controller/noise_spectrum_estimator.cc", - "level_controller/noise_spectrum_estimator.h", - "level_controller/peak_level_estimator.cc", - "level_controller/peak_level_estimator.h", - "level_controller/saturating_gain_estimator.cc", - "level_controller/saturating_gain_estimator.h", - "level_controller/signal_classifier.cc", - "level_controller/signal_classifier.h", "level_estimator_impl.cc", "level_estimator_impl.h", "low_cut_filter.cc", @@ -610,7 +589,6 @@ if (rtc_include_tests) { "echo_detector/moving_max_unittest.cc", "echo_detector/normalized_covariance_estimator_unittest.cc", "gain_control_unittest.cc", - "level_controller/level_controller_unittest.cc", "level_estimator_unittest.cc", "low_cut_filter_unittest.cc", "noise_suppression_unittest.cc", @@ -638,7 +616,6 @@ if (rtc_include_tests) { sources = [ "audio_processing_performance_unittest.cc", - "level_controller/level_controller_complexity_unittest.cc", ] deps = [ ":audio_processing", diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc index f4b8dee221..0caa1422f0 100644 --- a/modules/audio_processing/audio_processing_impl.cc +++ b/modules/audio_processing/audio_processing_impl.cc @@ -37,7 +37,6 @@ #if WEBRTC_INTELLIGIBILITY_ENHANCER #include "modules/audio_processing/intelligibility/intelligibility_enhancer.h" #endif -#include "modules/audio_processing/level_controller/level_controller.h" #include "modules/audio_processing/level_estimator_impl.h" #include "modules/audio_processing/low_cut_filter.h" #include "modules/audio_processing/noise_suppression_impl.h" @@ -188,7 +187,6 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update( bool beamformer_enabled, bool adaptive_gain_controller_enabled, bool gain_controller2_enabled, - bool level_controller_enabled, bool echo_controller_enabled, bool voice_activity_detector_enabled, bool level_estimator_enabled, @@ -208,7 +206,6 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update( (adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_); changed |= (gain_controller2_enabled != gain_controller2_enabled_); - changed |= (level_controller_enabled != level_controller_enabled_); changed |= (echo_controller_enabled != echo_controller_enabled_); changed |= (level_estimator_enabled != level_estimator_enabled_); changed |= @@ -224,7 +221,6 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update( beamformer_enabled_ = beamformer_enabled; adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled; gain_controller2_enabled_ = gain_controller2_enabled; - level_controller_enabled_ = level_controller_enabled; echo_controller_enabled_ = echo_controller_enabled; level_estimator_enabled_ = level_estimator_enabled; voice_activity_detector_enabled_ = voice_activity_detector_enabled; @@ -256,8 +252,7 @@ bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive() bool AudioProcessingImpl::ApmSubmoduleStates::CaptureFullBandProcessingActive() const { - return level_controller_enabled_ || gain_controller2_enabled_ || - capture_post_processor_enabled_; + return gain_controller2_enabled_ || capture_post_processor_enabled_; } bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive() @@ -314,7 +309,6 @@ struct AudioProcessingImpl::ApmPrivateSubmodules { std::unique_ptr agc_manager; std::unique_ptr gain_controller2; std::unique_ptr low_cut_filter; - std::unique_ptr level_controller; std::unique_ptr echo_detector; std::unique_ptr echo_controller; std::unique_ptr capture_post_processor; @@ -440,10 +434,6 @@ AudioProcessingImpl::AudioProcessingImpl( private_submodules_->echo_detector.reset(new ResidualEchoDetector()); } - // TODO(peah): Move this creation to happen only when the level controller - // is enabled. - private_submodules_->level_controller.reset(new LevelController()); - // TODO(alessiob): Move the injected gain controller once injection is // implemented. private_submodules_->gain_controller2.reset(new GainController2()); @@ -602,7 +592,6 @@ int AudioProcessingImpl::InitializeLocked() { proc_sample_rate_hz()); public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz()); public_submodules_->level_estimator->Initialize(); - InitializeLevelController(); InitializeResidualEchoDetector(); InitializeEchoController(); InitializeGainController2(); @@ -706,40 +695,16 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) { void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) { config_ = config; - bool config_ok = LevelController::Validate(config_.level_controller); - if (!config_ok) { - RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n" - "level_controller: " - << LevelController::ToString(config_.level_controller) - << "\nReverting to default parameter set"; - config_.level_controller = AudioProcessing::Config::LevelController(); - } - // Run in a single-threaded manner when applying the settings. rtc::CritScope cs_render(&crit_render_); rtc::CritScope cs_capture(&crit_capture_); - // TODO(peah): Replace the use of capture_nonlocked_.level_controller_enabled - // with the value in config_ everywhere in the code. - if (capture_nonlocked_.level_controller_enabled != - config_.level_controller.enabled) { - capture_nonlocked_.level_controller_enabled = - config_.level_controller.enabled; - // TODO(peah): Remove the conditional initialization to always initialize - // the level controller regardless of whether it is enabled or not. - InitializeLevelController(); - } - RTC_LOG(LS_INFO) << "Level controller activated: " - << capture_nonlocked_.level_controller_enabled; - - private_submodules_->level_controller->ApplyConfig(config_.level_controller); - InitializeLowCutFilter(); RTC_LOG(LS_INFO) << "Highpass filter activated: " << config_.high_pass_filter.enabled; - config_ok = GainController2::Validate(config_.gain_controller2); + const bool config_ok = GainController2::Validate(config_.gain_controller2); if (!config_ok) { RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n" "Gain Controller 2: " @@ -1259,13 +1224,11 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() { #if WEBRTC_INTELLIGIBILITY_ENHANCER if (capture_nonlocked_.intelligibility_enabled) { RTC_DCHECK(public_submodules_->noise_suppression->is_enabled()); - int gain_db = public_submodules_->gain_control->is_enabled() ? - public_submodules_->gain_control->compression_gain_db() : - 0; - float gain = DbToRatio(gain_db); - gain *= capture_nonlocked_.level_controller_enabled ? - private_submodules_->level_controller->GetLastGain() : - 1.f; + const int gain_db = + public_submodules_->gain_control->is_enabled() + ? public_submodules_->gain_control->compression_gain_db() + : 0; + const float gain = DbToRatio(gain_db); public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate( public_submodules_->noise_suppression->NoiseEstimate(), gain); } @@ -1335,10 +1298,6 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() { private_submodules_->gain_controller2->Process(capture_buffer); } - if (capture_nonlocked_.level_controller_enabled) { - private_submodules_->level_controller->Process(capture_buffer); - } - if (private_submodules_->capture_post_processor) { private_submodules_->capture_post_processor->Process(capture_buffer); } @@ -1766,7 +1725,6 @@ bool AudioProcessingImpl::UpdateActiveSubmoduleStates() { capture_nonlocked_.beamformer_enabled, public_submodules_->gain_control->is_enabled(), config_.gain_controller2.enabled, - capture_nonlocked_.level_controller_enabled, capture_nonlocked_.echo_controller_enabled, public_submodules_->voice_detection->is_enabled(), public_submodules_->level_estimator->is_enabled(), @@ -1832,10 +1790,6 @@ void AudioProcessingImpl::InitializeGainController2() { } } -void AudioProcessingImpl::InitializeLevelController() { - private_submodules_->level_controller->Initialize(proc_sample_rate_hz()); -} - void AudioProcessingImpl::InitializeResidualEchoDetector() { RTC_DCHECK(private_submodules_->echo_detector); private_submodules_->echo_detector->Initialize(proc_sample_rate_hz(), @@ -1938,9 +1892,6 @@ void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) { public_submodules_->echo_cancellation->GetExperimentsDescription(); // TODO(peah): Add semicolon-separated concatenations of experiment // descriptions for other submodules. - if (capture_nonlocked_.level_controller_enabled) { - experiments_description += "LevelController;"; - } if (constants_.agc_clipped_level_min != kClippedLevelMin) { experiments_description += "AgcClippingLevelExperiment;"; } diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h index e7c6621ae6..55c47ac43e 100644 --- a/modules/audio_processing/audio_processing_impl.h +++ b/modules/audio_processing/audio_processing_impl.h @@ -169,7 +169,6 @@ class AudioProcessingImpl : public AudioProcessing { bool beamformer_enabled, bool adaptive_gain_controller_enabled, bool gain_controller2_enabled, - bool level_controller_enabled, bool echo_controller_enabled, bool voice_activity_detector_enabled, bool level_estimator_enabled, @@ -193,7 +192,6 @@ class AudioProcessingImpl : public AudioProcessing { bool beamformer_enabled_ = false; bool adaptive_gain_controller_enabled_ = false; bool gain_controller2_enabled_ = false; - bool level_controller_enabled_ = false; bool echo_controller_enabled_ = false; bool level_estimator_enabled_ = false; bool voice_activity_detector_enabled_ = false; @@ -233,7 +231,6 @@ class AudioProcessingImpl : public AudioProcessing { RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); int InitializeLocked(const ProcessingConfig& config) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); - void InitializeLevelController() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); void InitializeResidualEchoDetector() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); void InitializeLowCutFilter() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); @@ -386,7 +383,6 @@ class AudioProcessingImpl : public AudioProcessing { int stream_delay_ms; bool beamformer_enabled; bool intelligibility_enabled; - bool level_controller_enabled = false; bool echo_controller_enabled = false; } capture_nonlocked_; diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc index ecaeed3edc..89d6cb9ee3 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -25,7 +25,6 @@ #include "modules/audio_processing/common.h" #include "modules/audio_processing/include/audio_processing.h" #include "modules/audio_processing/include/mock_audio_processing.h" -#include "modules/audio_processing/level_controller/level_controller_constants.h" #include "modules/audio_processing/test/protobuf_utils.h" #include "modules/audio_processing/test/test_utils.h" #include "modules/include/module_common_types.h" @@ -2821,98 +2820,6 @@ INSTANTIATE_TEST_CASE_P( } // namespace -TEST(ApmConfiguration, DefaultBehavior) { - // Verify that the level controller is default off, it can be activated using - // the config, and that the default initial level is maintained after the - // config has been applied. - std::unique_ptr apm( - new rtc::RefCountedObject(webrtc::Config())); - AudioProcessing::Config config; - EXPECT_FALSE(apm->config_.level_controller.enabled); - // TODO(peah): Add test for the existence of the level controller object once - // that is created only when that is specified in the config. - // TODO(peah): Remove the testing for - // apm->capture_nonlocked_.level_controller_enabled once the value in config_ - // is instead used to activate the level controller. - EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); - EXPECT_NEAR(kTargetLcPeakLeveldBFS, - apm->config_.level_controller.initial_peak_level_dbfs, - std::numeric_limits::epsilon()); - config.level_controller.enabled = true; - apm->ApplyConfig(config); - EXPECT_TRUE(apm->config_.level_controller.enabled); - // TODO(peah): Add test for the existence of the level controller object once - // that is created only when the that is specified in the config. - // TODO(peah): Remove the testing for - // apm->capture_nonlocked_.level_controller_enabled once the value in config_ - // is instead used to activate the level controller. - EXPECT_TRUE(apm->capture_nonlocked_.level_controller_enabled); - EXPECT_NEAR(kTargetLcPeakLeveldBFS, - apm->config_.level_controller.initial_peak_level_dbfs, - std::numeric_limits::epsilon()); -} - -TEST(ApmConfiguration, ValidConfigBehavior) { - // Verify that the initial level can be specified and is retained after the - // config has been applied. - std::unique_ptr apm( - new rtc::RefCountedObject(webrtc::Config())); - AudioProcessing::Config config; - config.level_controller.initial_peak_level_dbfs = -50.f; - apm->ApplyConfig(config); - EXPECT_FALSE(apm->config_.level_controller.enabled); - // TODO(peah): Add test for the existence of the level controller object once - // that is created only when the that is specified in the config. - // TODO(peah): Remove the testing for - // apm->capture_nonlocked_.level_controller_enabled once the value in config_ - // is instead used to activate the level controller. - EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); - EXPECT_NEAR(-50.f, apm->config_.level_controller.initial_peak_level_dbfs, - std::numeric_limits::epsilon()); -} - -TEST(ApmConfiguration, InValidConfigBehavior) { - // Verify that the config is properly reset when nonproper values are applied - // for the initial level. - - // Verify that the config is properly reset when the specified initial peak - // level is too low. - std::unique_ptr apm( - new rtc::RefCountedObject(webrtc::Config())); - AudioProcessing::Config config; - config.level_controller.enabled = true; - config.level_controller.initial_peak_level_dbfs = -101.f; - apm->ApplyConfig(config); - EXPECT_FALSE(apm->config_.level_controller.enabled); - // TODO(peah): Add test for the existence of the level controller object once - // that is created only when the that is specified in the config. - // TODO(peah): Remove the testing for - // apm->capture_nonlocked_.level_controller_enabled once the value in config_ - // is instead used to activate the level controller. - EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); - EXPECT_NEAR(kTargetLcPeakLeveldBFS, - apm->config_.level_controller.initial_peak_level_dbfs, - std::numeric_limits::epsilon()); - - // Verify that the config is properly reset when the specified initial peak - // level is too high. - apm.reset(new rtc::RefCountedObject(webrtc::Config())); - config = AudioProcessing::Config(); - config.level_controller.enabled = true; - config.level_controller.initial_peak_level_dbfs = 1.f; - apm->ApplyConfig(config); - EXPECT_FALSE(apm->config_.level_controller.enabled); - // TODO(peah): Add test for the existence of the level controller object once - // that is created only when that is specified in the config. - // TODO(peah): Remove the testing for - // apm->capture_nonlocked_.level_controller_enabled once the value in config_ - // is instead used to activate the level controller. - EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); - EXPECT_NEAR(kTargetLcPeakLeveldBFS, - apm->config_.level_controller.initial_peak_level_dbfs, - std::numeric_limits::epsilon()); -} - TEST(ApmConfiguration, EnablePostProcessing) { // Verify that apm uses a capture post processing module if one is provided. webrtc::Config webrtc_config; @@ -3007,7 +2914,6 @@ std::unique_ptr CreateApm(bool use_AEC2) { config.residual_echo_detector.enabled = true; config.high_pass_filter.enabled = false; config.gain_controller2.enabled = false; - config.level_controller.enabled = false; apm->ApplyConfig(config); EXPECT_EQ(apm->gain_control()->Enable(false), 0); EXPECT_EQ(apm->level_estimator()->Enable(false), 0); diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h index 7057f2804f..33ecf89340 100644 --- a/modules/audio_processing/include/audio_processing.h +++ b/modules/audio_processing/include/audio_processing.h @@ -211,8 +211,8 @@ struct Intelligibility { // AudioProcessing* apm = AudioProcessingBuilder().Create(); // // AudioProcessing::Config config; -// config.level_controller.enabled = true; // config.high_pass_filter.enabled = true; +// config.gain_controller2.enabled = true; // apm->ApplyConfig(config) // // apm->echo_cancellation()->enable_drift_compensation(false); @@ -262,14 +262,6 @@ class AudioProcessing : public rtc::RefCountInterface { // by changing the default values in the AudioProcessing::Config struct. // The config is applied by passing the struct to the ApplyConfig method. struct Config { - struct LevelController { - bool enabled = false; - - // Sets the initial peak level to use inside the level controller in order - // to compute the signal gain. The unit for the peak level is dBFS and - // the allowed range is [-100, 0]. - float initial_peak_level_dbfs = -6.0206f; - } level_controller; struct ResidualEchoDetector { bool enabled = true; } residual_echo_detector; diff --git a/modules/audio_processing/include/config.h b/modules/audio_processing/include/config.h index 7c34de8ccc..7615f624cf 100644 --- a/modules/audio_processing/include/config.h +++ b/modules/audio_processing/include/config.h @@ -35,7 +35,7 @@ enum class ConfigOptionID { kIntelligibility, kEchoCanceller3, // Deprecated kAecRefinedAdaptiveFilter, - kLevelControl + kLevelControl // Deprecated }; // Class Config is designed to ease passing a set of options across webrtc code. diff --git a/modules/audio_processing/level_controller/biquad_filter.cc b/modules/audio_processing/level_controller/biquad_filter.cc deleted file mode 100644 index 5a4ddc891e..0000000000 --- a/modules/audio_processing/level_controller/biquad_filter.cc +++ /dev/null @@ -1,35 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/biquad_filter.h" - -namespace webrtc { - -// This method applies a biquad filter to an input signal x to produce an -// output signal y. The biquad coefficients are specified at the construction -// of the object. -void BiQuadFilter::Process(rtc::ArrayView x, - rtc::ArrayView y) { - for (size_t k = 0; k < x.size(); ++k) { - // Use temporary variable for x[k] to allow in-place function call - // (that x and y refer to the same array). - const float tmp = x[k]; - y[k] = coefficients_.b[0] * tmp + coefficients_.b[1] * biquad_state_.b[0] + - coefficients_.b[2] * biquad_state_.b[1] - - coefficients_.a[0] * biquad_state_.a[0] - - coefficients_.a[1] * biquad_state_.a[1]; - biquad_state_.b[1] = biquad_state_.b[0]; - biquad_state_.b[0] = tmp; - biquad_state_.a[1] = biquad_state_.a[0]; - biquad_state_.a[0] = y[k]; - } -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/biquad_filter.h b/modules/audio_processing/level_controller/biquad_filter.h deleted file mode 100644 index dad104d43f..0000000000 --- a/modules/audio_processing/level_controller/biquad_filter.h +++ /dev/null @@ -1,58 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_ - -#include - -#include "api/array_view.h" -#include "rtc_base/arraysize.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class BiQuadFilter { - public: - struct BiQuadCoefficients { - float b[3]; - float a[2]; - }; - - BiQuadFilter() = default; - - void Initialize(const BiQuadCoefficients& coefficients) { - coefficients_ = coefficients; - } - - // Produces a filtered output y of the input x. Both x and y need to - // have the same length. - void Process(rtc::ArrayView x, rtc::ArrayView y); - - private: - struct BiQuadState { - BiQuadState() { - std::fill(b, b + arraysize(b), 0.f); - std::fill(a, a + arraysize(a), 0.f); - } - - float b[2]; - float a[2]; - }; - - BiQuadState biquad_state_; - BiQuadCoefficients coefficients_; - - RTC_DISALLOW_COPY_AND_ASSIGN(BiQuadFilter); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_ diff --git a/modules/audio_processing/level_controller/down_sampler.cc b/modules/audio_processing/level_controller/down_sampler.cc deleted file mode 100644 index a1702f432c..0000000000 --- a/modules/audio_processing/level_controller/down_sampler.cc +++ /dev/null @@ -1,100 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/down_sampler.h" - -#include -#include -#include - -#include "modules/audio_processing/include/audio_processing.h" -#include "modules/audio_processing/level_controller/biquad_filter.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" -#include "rtc_base/checks.h" - -namespace webrtc { -namespace { - -// Bandlimiter coefficients computed based on that only -// the first 40 bins of the spectrum for the downsampled -// signal are used. -// [B,A] = butter(2,(41/64*4000)/8000) -const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = { - {0.1455f, 0.2911f, 0.1455f}, - {-0.6698f, 0.2520f}}; - -// [B,A] = butter(2,(41/64*4000)/16000) -const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = { - {0.0462f, 0.0924f, 0.0462f}, - {-1.3066f, 0.4915f}}; - -// [B,A] = butter(2,(41/64*4000)/24000) -const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = { - {0.0226f, 0.0452f, 0.0226f}, - {-1.5320f, 0.6224f}}; - -} // namespace - -DownSampler::DownSampler(ApmDataDumper* data_dumper) - : data_dumper_(data_dumper) { - Initialize(48000); -} -void DownSampler::Initialize(int sample_rate_hz) { - RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || - sample_rate_hz == AudioProcessing::kSampleRate16kHz || - sample_rate_hz == AudioProcessing::kSampleRate32kHz || - sample_rate_hz == AudioProcessing::kSampleRate48kHz); - - sample_rate_hz_ = sample_rate_hz; - down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000); - - /// Note that the down sampling filter is not used if the sample rate is 8 - /// kHz. - if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) { - low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz); - } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) { - low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz); - } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) { - low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz); - } -} - -void DownSampler::DownSample(rtc::ArrayView in, - rtc::ArrayView out) { - data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1); - RTC_DCHECK_EQ(sample_rate_hz_ * AudioProcessing::kChunkSizeMs / 1000, - in.size()); - RTC_DCHECK_EQ( - AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000, - out.size()); - const size_t kMaxNumFrames = - AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000; - float x[kMaxNumFrames]; - - // Band-limit the signal to 4 kHz. - if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) { - low_pass_filter_.Process(in, rtc::ArrayView(x, in.size())); - - // Downsample the signal. - size_t k = 0; - for (size_t j = 0; j < out.size(); ++j) { - RTC_DCHECK_GT(kMaxNumFrames, k); - out[j] = x[k]; - k += down_sampling_factor_; - } - } else { - std::copy(in.data(), in.data() + in.size(), out.data()); - } - - data_dumper_->DumpWav("lc_down_sampler_output", out, - AudioProcessing::kSampleRate8kHz, 1); -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/down_sampler.h b/modules/audio_processing/level_controller/down_sampler.h deleted file mode 100644 index d6502425a1..0000000000 --- a/modules/audio_processing/level_controller/down_sampler.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_ - -#include "api/array_view.h" -#include "modules/audio_processing/level_controller/biquad_filter.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class ApmDataDumper; - -class DownSampler { - public: - explicit DownSampler(ApmDataDumper* data_dumper); - void Initialize(int sample_rate_hz); - - void DownSample(rtc::ArrayView in, rtc::ArrayView out); - - private: - ApmDataDumper* data_dumper_; - int sample_rate_hz_; - int down_sampling_factor_; - BiQuadFilter low_pass_filter_; - - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_ diff --git a/modules/audio_processing/level_controller/gain_applier.cc b/modules/audio_processing/level_controller/gain_applier.cc deleted file mode 100644 index 018f809e01..0000000000 --- a/modules/audio_processing/level_controller/gain_applier.cc +++ /dev/null @@ -1,160 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/gain_applier.h" - -#include - -#include "api/array_view.h" -#include "rtc_base/checks.h" - -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" - -namespace webrtc { -namespace { - -const float kMaxSampleValue = 32767.f; -const float kMinSampleValue = -32767.f; - -int CountSaturations(rtc::ArrayView in) { - return std::count_if(in.begin(), in.end(), [](const float& v) { - return v >= kMaxSampleValue || v <= kMinSampleValue; - }); -} - -int CountSaturations(const AudioBuffer& audio) { - int num_saturations = 0; - for (size_t k = 0; k < audio.num_channels(); ++k) { - num_saturations += CountSaturations(rtc::ArrayView( - audio.channels_const_f()[k], audio.num_frames())); - } - return num_saturations; -} - -void LimitToAllowedRange(rtc::ArrayView x) { - for (auto& v : x) { - v = std::max(kMinSampleValue, v); - v = std::min(kMaxSampleValue, v); - } -} - -void LimitToAllowedRange(AudioBuffer* audio) { - for (size_t k = 0; k < audio->num_channels(); ++k) { - LimitToAllowedRange( - rtc::ArrayView(audio->channels_f()[k], audio->num_frames())); - } -} - -float ApplyIncreasingGain(float new_gain, - float old_gain, - float step_size, - rtc::ArrayView x) { - RTC_DCHECK_LT(0.f, step_size); - float gain = old_gain; - for (auto& v : x) { - gain = std::min(new_gain, gain + step_size); - v *= gain; - } - return gain; -} - -float ApplyDecreasingGain(float new_gain, - float old_gain, - float step_size, - rtc::ArrayView x) { - RTC_DCHECK_GT(0.f, step_size); - float gain = old_gain; - for (auto& v : x) { - gain = std::max(new_gain, gain + step_size); - v *= gain; - } - return gain; -} - -float ApplyConstantGain(float gain, rtc::ArrayView x) { - for (auto& v : x) { - v *= gain; - } - - return gain; -} - -float ApplyGain(float new_gain, - float old_gain, - float increase_step_size, - float decrease_step_size, - rtc::ArrayView x) { - RTC_DCHECK_LT(0.f, increase_step_size); - RTC_DCHECK_GT(0.f, decrease_step_size); - if (new_gain == old_gain) { - return ApplyConstantGain(new_gain, x); - } else if (new_gain > old_gain) { - return ApplyIncreasingGain(new_gain, old_gain, increase_step_size, x); - } else { - return ApplyDecreasingGain(new_gain, old_gain, decrease_step_size, x); - } -} - -} // namespace - -GainApplier::GainApplier(ApmDataDumper* data_dumper) - : data_dumper_(data_dumper) {} - -void GainApplier::Initialize(int sample_rate_hz) { - RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || - sample_rate_hz == AudioProcessing::kSampleRate16kHz || - sample_rate_hz == AudioProcessing::kSampleRate32kHz || - sample_rate_hz == AudioProcessing::kSampleRate48kHz); - const float kGainIncreaseStepSize48kHz = 0.0001f; - const float kGainDecreaseStepSize48kHz = -0.01f; - const float kGainSaturatedDecreaseStepSize48kHz = -0.05f; - - last_frame_was_saturated_ = false; - old_gain_ = 1.f; - gain_increase_step_size_ = - kGainIncreaseStepSize48kHz * - (static_cast(AudioProcessing::kSampleRate48kHz) / sample_rate_hz); - gain_normal_decrease_step_size_ = - kGainDecreaseStepSize48kHz * - (static_cast(AudioProcessing::kSampleRate48kHz) / sample_rate_hz); - gain_saturated_decrease_step_size_ = - kGainSaturatedDecreaseStepSize48kHz * - (static_cast(AudioProcessing::kSampleRate48kHz) / sample_rate_hz); -} - -int GainApplier::Process(float new_gain, AudioBuffer* audio) { - RTC_CHECK_NE(0.f, gain_increase_step_size_); - RTC_CHECK_NE(0.f, gain_normal_decrease_step_size_); - RTC_CHECK_NE(0.f, gain_saturated_decrease_step_size_); - int num_saturations = 0; - if (new_gain != 1.f) { - float last_applied_gain = 1.f; - float gain_decrease_step_size = last_frame_was_saturated_ - ? gain_saturated_decrease_step_size_ - : gain_normal_decrease_step_size_; - for (size_t k = 0; k < audio->num_channels(); ++k) { - last_applied_gain = ApplyGain( - new_gain, old_gain_, gain_increase_step_size_, - gain_decrease_step_size, - rtc::ArrayView(audio->channels_f()[k], audio->num_frames())); - } - - num_saturations = CountSaturations(*audio); - LimitToAllowedRange(audio); - old_gain_ = last_applied_gain; - } - - data_dumper_->DumpRaw("lc_last_applied_gain", 1, &old_gain_); - - return num_saturations; -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/gain_applier.h b/modules/audio_processing/level_controller/gain_applier.h deleted file mode 100644 index 5669f45bf7..0000000000 --- a/modules/audio_processing/level_controller/gain_applier.h +++ /dev/null @@ -1,42 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ - -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class ApmDataDumper; -class AudioBuffer; - -class GainApplier { - public: - explicit GainApplier(ApmDataDumper* data_dumper); - void Initialize(int sample_rate_hz); - - // Applies the specified gain to the audio frame and returns the resulting - // number of saturated sample values. - int Process(float new_gain, AudioBuffer* audio); - - private: - ApmDataDumper* const data_dumper_; - float old_gain_ = 1.f; - float gain_increase_step_size_ = 0.f; - float gain_normal_decrease_step_size_ = 0.f; - float gain_saturated_decrease_step_size_ = 0.f; - bool last_frame_was_saturated_; - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ diff --git a/modules/audio_processing/level_controller/gain_selector.cc b/modules/audio_processing/level_controller/gain_selector.cc deleted file mode 100644 index 3ab75b1ce6..0000000000 --- a/modules/audio_processing/level_controller/gain_selector.cc +++ /dev/null @@ -1,87 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/gain_selector.h" - -#include -#include - -#include "modules/audio_processing/include/audio_processing.h" -#include "modules/audio_processing/level_controller/level_controller_constants.h" -#include "rtc_base/checks.h" - -namespace webrtc { - -GainSelector::GainSelector() { - Initialize(AudioProcessing::kSampleRate48kHz); -} - -void GainSelector::Initialize(int sample_rate_hz) { - gain_ = 1.f; - frame_length_ = rtc::CheckedDivExact(sample_rate_hz, 100); - highly_nonstationary_signal_hold_counter_ = 0; -} - -// Chooses the gain to apply by the level controller such that -// 1) The level of the stationary noise does not exceed -// a predefined threshold. -// 2) The gain does not exceed the gain that has been found -// to saturate the signal. -// 3) The peak level achieves the target peak level. -// 4) The gain is not below 1. -// 4) The gain is 1 if the signal has been classified as stationary -// for a long time. -// 5) The gain is not above the maximum gain. -float GainSelector::GetNewGain(float peak_level, - float noise_energy, - float saturating_gain, - bool gain_jumpstart, - SignalClassifier::SignalType signal_type) { - RTC_DCHECK_LT(0.f, peak_level); - - if (signal_type == SignalClassifier::SignalType::kHighlyNonStationary || - gain_jumpstart) { - highly_nonstationary_signal_hold_counter_ = 100; - } else { - highly_nonstationary_signal_hold_counter_ = - std::max(0, highly_nonstationary_signal_hold_counter_ - 1); - } - - float desired_gain; - if (highly_nonstationary_signal_hold_counter_ > 0) { - // Compute a desired gain that ensures that the peak level is amplified to - // the target level. - desired_gain = kTargetLcPeakLevel / peak_level; - - // Limit the desired gain so that it does not amplify the noise too much. - float max_noise_energy = kMaxLcNoisePower * frame_length_; - if (noise_energy * desired_gain * desired_gain > max_noise_energy) { - RTC_DCHECK_LE(0.f, noise_energy); - desired_gain = sqrtf(max_noise_energy / noise_energy); - } - } else { - // If the signal has been stationary for a long while, apply a gain of 1 to - // avoid amplifying pure noise. - desired_gain = 1.0f; - } - - // Smootly update the gain towards the desired gain. - gain_ += 0.2f * (desired_gain - gain_); - - // Limit the gain to not exceed the maximum and the saturating gains, and to - // ensure that the lowest possible gain is 1. - gain_ = std::min(gain_, saturating_gain); - gain_ = std::min(gain_, kMaxLcGain); - gain_ = std::max(gain_, 1.f); - - return gain_; -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/gain_selector.h b/modules/audio_processing/level_controller/gain_selector.h deleted file mode 100644 index 7966c438d7..0000000000 --- a/modules/audio_processing/level_controller/gain_selector.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_ - -#include "rtc_base/constructormagic.h" - -#include "modules/audio_processing/level_controller/signal_classifier.h" - -namespace webrtc { - -class GainSelector { - public: - GainSelector(); - void Initialize(int sample_rate_hz); - float GetNewGain(float peak_level, - float noise_energy, - float saturating_gain, - bool gain_jumpstart, - SignalClassifier::SignalType signal_type); - - private: - float gain_; - size_t frame_length_; - int highly_nonstationary_signal_hold_counter_; - - RTC_DISALLOW_COPY_AND_ASSIGN(GainSelector); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_ diff --git a/modules/audio_processing/level_controller/level_controller.cc b/modules/audio_processing/level_controller/level_controller.cc deleted file mode 100644 index b7854a0c9d..0000000000 --- a/modules/audio_processing/level_controller/level_controller.cc +++ /dev/null @@ -1,295 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/level_controller.h" - -#include -#include -#include - -#include "api/array_view.h" -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/level_controller/gain_applier.h" -#include "modules/audio_processing/level_controller/gain_selector.h" -#include "modules/audio_processing/level_controller/noise_level_estimator.h" -#include "modules/audio_processing/level_controller/peak_level_estimator.h" -#include "modules/audio_processing/level_controller/saturating_gain_estimator.h" -#include "modules/audio_processing/level_controller/signal_classifier.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" -#include "rtc_base/arraysize.h" -#include "rtc_base/checks.h" -#include "rtc_base/logging.h" -#include "system_wrappers/include/metrics.h" - -namespace webrtc { -namespace { - -void UpdateAndRemoveDcLevel(float forgetting_factor, - float* dc_level, - rtc::ArrayView x) { - RTC_DCHECK(!x.empty()); - float mean = - std::accumulate(x.begin(), x.end(), 0.0f) / static_cast(x.size()); - *dc_level += forgetting_factor * (mean - *dc_level); - - for (float& v : x) { - v -= *dc_level; - } -} - -float FrameEnergy(const AudioBuffer& audio) { - float energy = 0.f; - for (size_t k = 0; k < audio.num_channels(); ++k) { - float channel_energy = - std::accumulate(audio.channels_const_f()[k], - audio.channels_const_f()[k] + audio.num_frames(), 0.f, - [](float a, float b) -> float { return a + b * b; }); - energy = std::max(channel_energy, energy); - } - return energy; -} - -float PeakLevel(const AudioBuffer& audio) { - float peak_level = 0.f; - for (size_t k = 0; k < audio.num_channels(); ++k) { - auto* channel_peak_level = std::max_element( - audio.channels_const_f()[k], - audio.channels_const_f()[k] + audio.num_frames(), - [](float a, float b) { return std::abs(a) < std::abs(b); }); - peak_level = std::max(*channel_peak_level, peak_level); - } - return peak_level; -} - -const int kMetricsFrameInterval = 1000; - -} // namespace - -int LevelController::instance_count_ = 0; - -void LevelController::Metrics::Initialize(int sample_rate_hz) { - RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || - sample_rate_hz == AudioProcessing::kSampleRate16kHz || - sample_rate_hz == AudioProcessing::kSampleRate32kHz || - sample_rate_hz == AudioProcessing::kSampleRate48kHz); - - Reset(); - frame_length_ = rtc::CheckedDivExact(sample_rate_hz, 100); -} - -void LevelController::Metrics::Reset() { - metrics_frame_counter_ = 0; - gain_sum_ = 0.f; - peak_level_sum_ = 0.f; - noise_energy_sum_ = 0.f; - max_gain_ = 0.f; - max_peak_level_ = 0.f; - max_noise_energy_ = 0.f; -} - -void LevelController::Metrics::Update(float long_term_peak_level, - float noise_energy, - float gain, - float frame_peak_level) { - const float kdBFSOffset = 90.3090f; - gain_sum_ += gain; - peak_level_sum_ += long_term_peak_level; - noise_energy_sum_ += noise_energy; - max_gain_ = std::max(max_gain_, gain); - max_peak_level_ = std::max(max_peak_level_, long_term_peak_level); - max_noise_energy_ = std::max(max_noise_energy_, noise_energy); - - ++metrics_frame_counter_; - if (metrics_frame_counter_ == kMetricsFrameInterval) { - RTC_DCHECK_LT(0, frame_length_); - RTC_DCHECK_LT(0, kMetricsFrameInterval); - - const int max_noise_power_dbfs = static_cast( - 10 * log10(max_noise_energy_ / frame_length_ + 1e-10f) - kdBFSOffset); - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxNoisePower", - max_noise_power_dbfs, -90, 0, 50); - - const int average_noise_power_dbfs = static_cast( - 10 * log10(noise_energy_sum_ / (frame_length_ * kMetricsFrameInterval) + - 1e-10f) - - kdBFSOffset); - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageNoisePower", - average_noise_power_dbfs, -90, 0, 50); - - const int max_peak_level_dbfs = static_cast( - 10 * log10(max_peak_level_ * max_peak_level_ + 1e-10f) - kdBFSOffset); - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxPeakLevel", - max_peak_level_dbfs, -90, 0, 50); - - const int average_peak_level_dbfs = static_cast( - 10 * log10(peak_level_sum_ * peak_level_sum_ / - (kMetricsFrameInterval * kMetricsFrameInterval) + - 1e-10f) - - kdBFSOffset); - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AveragePeakLevel", - average_peak_level_dbfs, -90, 0, 50); - - RTC_DCHECK_LE(1.f, max_gain_); - RTC_DCHECK_LE(1.f, gain_sum_ / kMetricsFrameInterval); - - const int max_gain_db = static_cast(10 * log10(max_gain_ * max_gain_)); - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxGain", max_gain_db, 0, - 33, 30); - - const int average_gain_db = static_cast( - 10 * log10(gain_sum_ * gain_sum_ / - (kMetricsFrameInterval * kMetricsFrameInterval))); - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageGain", - average_gain_db, 0, 33, 30); - - const int long_term_peak_level_dbfs = static_cast( - 10 * log10(long_term_peak_level * long_term_peak_level + 1e-10f) - - kdBFSOffset); - - const int frame_peak_level_dbfs = static_cast( - 10 * log10(frame_peak_level * frame_peak_level + 1e-10f) - kdBFSOffset); - - RTC_LOG(LS_INFO) << "Level Controller metrics: {Max noise power: " - << max_noise_power_dbfs - << " dBFS, Average noise power: " - << average_noise_power_dbfs - << " dBFS, Max long term peak level: " - << max_peak_level_dbfs - << " dBFS, Average long term peak level: " - << average_peak_level_dbfs - << " dBFS, Max gain: " - << max_gain_db - << " dB, Average gain: " - << average_gain_db - << " dB, Long term peak level: " - << long_term_peak_level_dbfs - << " dBFS, Last frame peak level: " - << frame_peak_level_dbfs - << " dBFS}"; - - Reset(); - } -} - -LevelController::LevelController() - : data_dumper_(new ApmDataDumper(instance_count_)), - gain_applier_(data_dumper_.get()), - signal_classifier_(data_dumper_.get()), - peak_level_estimator_(kTargetLcPeakLeveldBFS) { - Initialize(AudioProcessing::kSampleRate48kHz); - ++instance_count_; -} - -LevelController::~LevelController() {} - -void LevelController::Initialize(int sample_rate_hz) { - RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || - sample_rate_hz == AudioProcessing::kSampleRate16kHz || - sample_rate_hz == AudioProcessing::kSampleRate32kHz || - sample_rate_hz == AudioProcessing::kSampleRate48kHz); - data_dumper_->InitiateNewSetOfRecordings(); - gain_selector_.Initialize(sample_rate_hz); - gain_applier_.Initialize(sample_rate_hz); - signal_classifier_.Initialize(sample_rate_hz); - noise_level_estimator_.Initialize(sample_rate_hz); - peak_level_estimator_.Initialize(config_.initial_peak_level_dbfs); - saturating_gain_estimator_.Initialize(); - metrics_.Initialize(sample_rate_hz); - - last_gain_ = 1.0f; - sample_rate_hz_ = sample_rate_hz; - dc_forgetting_factor_ = 0.01f * sample_rate_hz / 48000.f; - std::fill(dc_level_, dc_level_ + arraysize(dc_level_), 0.f); -} - -void LevelController::Process(AudioBuffer* audio) { - RTC_DCHECK_LT(0, audio->num_channels()); - RTC_DCHECK_GE(2, audio->num_channels()); - RTC_DCHECK_NE(0.f, dc_forgetting_factor_); - RTC_DCHECK(sample_rate_hz_); - data_dumper_->DumpWav("lc_input", audio->num_frames(), - audio->channels_const_f()[0], *sample_rate_hz_, 1); - - // Remove DC level. - for (size_t k = 0; k < audio->num_channels(); ++k) { - UpdateAndRemoveDcLevel( - dc_forgetting_factor_, &dc_level_[k], - rtc::ArrayView(audio->channels_f()[k], audio->num_frames())); - } - - SignalClassifier::SignalType signal_type; - signal_classifier_.Analyze(*audio, &signal_type); - int tmp = static_cast(signal_type); - data_dumper_->DumpRaw("lc_signal_type", 1, &tmp); - - // Estimate the noise energy. - float noise_energy = - noise_level_estimator_.Analyze(signal_type, FrameEnergy(*audio)); - - // Estimate the overall signal peak level. - const float frame_peak_level = PeakLevel(*audio); - const float long_term_peak_level = - peak_level_estimator_.Analyze(signal_type, frame_peak_level); - - float saturating_gain = saturating_gain_estimator_.GetGain(); - - // Compute the new gain to apply. - last_gain_ = - gain_selector_.GetNewGain(long_term_peak_level, noise_energy, - saturating_gain, gain_jumpstart_, signal_type); - - // Unflag the jumpstart of the gain as it should only happen once. - gain_jumpstart_ = false; - - // Apply the gain to the signal. - int num_saturations = gain_applier_.Process(last_gain_, audio); - - // Estimate the gain that saturates the overall signal. - saturating_gain_estimator_.Update(last_gain_, num_saturations); - - // Update the metrics. - metrics_.Update(long_term_peak_level, noise_energy, last_gain_, - frame_peak_level); - - data_dumper_->DumpRaw("lc_selected_gain", 1, &last_gain_); - data_dumper_->DumpRaw("lc_noise_energy", 1, &noise_energy); - data_dumper_->DumpRaw("lc_peak_level", 1, &long_term_peak_level); - data_dumper_->DumpRaw("lc_saturating_gain", 1, &saturating_gain); - - data_dumper_->DumpWav("lc_output", audio->num_frames(), - audio->channels_f()[0], *sample_rate_hz_, 1); -} - -void LevelController::ApplyConfig( - const AudioProcessing::Config::LevelController& config) { - RTC_DCHECK(Validate(config)); - config_ = config; - peak_level_estimator_.Initialize(config_.initial_peak_level_dbfs); - gain_jumpstart_ = true; -} - -std::string LevelController::ToString( - const AudioProcessing::Config::LevelController& config) { - std::stringstream ss; - ss << "{" - << "enabled: " << (config.enabled ? "true" : "false") << ", " - << "initial_peak_level_dbfs: " << config.initial_peak_level_dbfs << "}"; - return ss.str(); -} - -bool LevelController::Validate( - const AudioProcessing::Config::LevelController& config) { - return (config.initial_peak_level_dbfs < - std::numeric_limits::epsilon() && - config.initial_peak_level_dbfs > - -(100.f + std::numeric_limits::epsilon())); -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/level_controller.h b/modules/audio_processing/level_controller/level_controller.h deleted file mode 100644 index 224b886abd..0000000000 --- a/modules/audio_processing/level_controller/level_controller.h +++ /dev/null @@ -1,95 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_ - -#include -#include - -#include "api/optional.h" -#include "modules/audio_processing/include/audio_processing.h" -#include "modules/audio_processing/level_controller/gain_applier.h" -#include "modules/audio_processing/level_controller/gain_selector.h" -#include "modules/audio_processing/level_controller/noise_level_estimator.h" -#include "modules/audio_processing/level_controller/peak_level_estimator.h" -#include "modules/audio_processing/level_controller/saturating_gain_estimator.h" -#include "modules/audio_processing/level_controller/signal_classifier.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class ApmDataDumper; -class AudioBuffer; - -class LevelController { - public: - LevelController(); - ~LevelController(); - - void Initialize(int sample_rate_hz); - void Process(AudioBuffer* audio); - float GetLastGain() { return last_gain_; } - - // TODO(peah): This method is a temporary solution as the the aim is to - // instead apply the config inside the constructor. Therefore this is likely - // to change. - void ApplyConfig(const AudioProcessing::Config::LevelController& config); - // Validates a config. - static bool Validate(const AudioProcessing::Config::LevelController& config); - // Dumps a config to a string. - static std::string ToString( - const AudioProcessing::Config::LevelController& config); - - private: - class Metrics { - public: - Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); } - void Initialize(int sample_rate_hz); - void Update(float long_term_peak_level, - float noise_level, - float gain, - float frame_peak_level); - - private: - void Reset(); - - size_t metrics_frame_counter_; - float gain_sum_; - float peak_level_sum_; - float noise_energy_sum_; - float max_gain_; - float max_peak_level_; - float max_noise_energy_; - float frame_length_; - }; - - std::unique_ptr data_dumper_; - GainSelector gain_selector_; - GainApplier gain_applier_; - SignalClassifier signal_classifier_; - NoiseLevelEstimator noise_level_estimator_; - PeakLevelEstimator peak_level_estimator_; - SaturatingGainEstimator saturating_gain_estimator_; - Metrics metrics_; - rtc::Optional sample_rate_hz_; - static int instance_count_; - float dc_level_[2]; - float dc_forgetting_factor_; - float last_gain_; - bool gain_jumpstart_ = false; - AudioProcessing::Config::LevelController config_; - - RTC_DISALLOW_COPY_AND_ASSIGN(LevelController); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_ diff --git a/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc b/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc deleted file mode 100644 index 83f6725a0f..0000000000 --- a/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc +++ /dev/null @@ -1,240 +0,0 @@ -/* - * Copyright (c) 2016 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 -#include - -#include "api/array_view.h" -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/include/audio_processing.h" -#include "modules/audio_processing/level_controller/level_controller.h" -#include "modules/audio_processing/test/audio_buffer_tools.h" -#include "modules/audio_processing/test/bitexactness_tools.h" -#include "modules/audio_processing/test/performance_timer.h" -#include "modules/audio_processing/test/simulator_buffers.h" -#include "rtc_base/random.h" -#include "system_wrappers/include/clock.h" -#include "test/gtest.h" -#include "test/testsupport/perf_test.h" - -namespace webrtc { -namespace { - -const size_t kNumFramesToProcess = 300; -const size_t kNumFramesToProcessAtWarmup = 300; -const size_t kToTalNumFrames = - kNumFramesToProcess + kNumFramesToProcessAtWarmup; - -void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) { - test::SimulatorBuffers buffers(sample_rate_hz, sample_rate_hz, sample_rate_hz, - sample_rate_hz, num_channels, num_channels, - num_channels, num_channels); - test::PerformanceTimer timer(kNumFramesToProcess); - - LevelController level_controller; - level_controller.Initialize(sample_rate_hz); - - for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) { - buffers.UpdateInputBuffers(); - - if (frame_no >= kNumFramesToProcessAtWarmup) { - timer.StartTimer(); - } - level_controller.Process(buffers.capture_input_buffer.get()); - if (frame_no >= kNumFramesToProcessAtWarmup) { - timer.StopTimer(); - } - } - webrtc::test::PrintResultMeanAndError( - "level_controller_call_durations", - "_" + std::to_string(sample_rate_hz) + "Hz_" + - std::to_string(num_channels) + "_channels", - "StandaloneLevelControl", timer.GetDurationAverage(), - timer.GetDurationStandardDeviation(), "us", false); -} - -void RunTogetherWithApm(const std::string& test_description, - int render_input_sample_rate_hz, - int render_output_sample_rate_hz, - int capture_input_sample_rate_hz, - int capture_output_sample_rate_hz, - size_t num_channels, - bool use_mobile_aec, - bool include_default_apm_processing) { - test::SimulatorBuffers buffers( - render_input_sample_rate_hz, capture_input_sample_rate_hz, - render_output_sample_rate_hz, capture_output_sample_rate_hz, num_channels, - num_channels, num_channels, num_channels); - test::PerformanceTimer render_timer(kNumFramesToProcess); - test::PerformanceTimer capture_timer(kNumFramesToProcess); - test::PerformanceTimer total_timer(kNumFramesToProcess); - - webrtc::Config config; - AudioProcessing::Config apm_config; - if (include_default_apm_processing) { - config.Set(new DelayAgnostic(true)); - config.Set(new ExtendedFilter(true)); - } - apm_config.level_controller.enabled = true; - apm_config.residual_echo_detector.enabled = include_default_apm_processing; - - std::unique_ptr apm; - apm.reset(AudioProcessingBuilder().Create(config)); - ASSERT_TRUE(apm.get()); - apm->ApplyConfig(apm_config); - - ASSERT_EQ(AudioProcessing::kNoError, - apm->gain_control()->Enable(include_default_apm_processing)); - if (use_mobile_aec) { - ASSERT_EQ(AudioProcessing::kNoError, - apm->echo_cancellation()->Enable(false)); - ASSERT_EQ(AudioProcessing::kNoError, apm->echo_control_mobile()->Enable( - include_default_apm_processing)); - } else { - ASSERT_EQ(AudioProcessing::kNoError, - apm->echo_cancellation()->Enable(include_default_apm_processing)); - ASSERT_EQ(AudioProcessing::kNoError, - apm->echo_control_mobile()->Enable(false)); - } - apm_config.high_pass_filter.enabled = include_default_apm_processing; - ASSERT_EQ(AudioProcessing::kNoError, - apm->noise_suppression()->Enable(include_default_apm_processing)); - ASSERT_EQ(AudioProcessing::kNoError, - apm->voice_detection()->Enable(include_default_apm_processing)); - ASSERT_EQ(AudioProcessing::kNoError, - apm->level_estimator()->Enable(include_default_apm_processing)); - - StreamConfig render_input_config(render_input_sample_rate_hz, num_channels, - false); - StreamConfig render_output_config(render_output_sample_rate_hz, num_channels, - false); - StreamConfig capture_input_config(capture_input_sample_rate_hz, num_channels, - false); - StreamConfig capture_output_config(capture_output_sample_rate_hz, - num_channels, false); - - for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) { - buffers.UpdateInputBuffers(); - - if (frame_no >= kNumFramesToProcessAtWarmup) { - total_timer.StartTimer(); - render_timer.StartTimer(); - } - ASSERT_EQ(AudioProcessing::kNoError, - apm->ProcessReverseStream( - &buffers.render_input[0], render_input_config, - render_output_config, &buffers.render_output[0])); - - if (frame_no >= kNumFramesToProcessAtWarmup) { - render_timer.StopTimer(); - - capture_timer.StartTimer(); - } - - ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0)); - ASSERT_EQ( - AudioProcessing::kNoError, - apm->ProcessStream(&buffers.capture_input[0], capture_input_config, - capture_output_config, &buffers.capture_output[0])); - - if (frame_no >= kNumFramesToProcessAtWarmup) { - capture_timer.StopTimer(); - total_timer.StopTimer(); - } - } - - webrtc::test::PrintResultMeanAndError( - "level_controller_call_durations", - "_" + std::to_string(render_input_sample_rate_hz) + "_" + - std::to_string(render_output_sample_rate_hz) + "_" + - std::to_string(capture_input_sample_rate_hz) + "_" + - std::to_string(capture_output_sample_rate_hz) + "Hz_" + - std::to_string(num_channels) + "_channels" + "_render", - test_description, render_timer.GetDurationAverage(), - render_timer.GetDurationStandardDeviation(), "us", false); - webrtc::test::PrintResultMeanAndError( - "level_controller_call_durations", - "_" + std::to_string(render_input_sample_rate_hz) + "_" + - std::to_string(render_output_sample_rate_hz) + "_" + - std::to_string(capture_input_sample_rate_hz) + "_" + - std::to_string(capture_output_sample_rate_hz) + "Hz_" + - std::to_string(num_channels) + "_channels" + "_capture", - test_description, capture_timer.GetDurationAverage(), - capture_timer.GetDurationStandardDeviation(), "us", false); - webrtc::test::PrintResultMeanAndError( - "level_controller_call_durations", - "_" + std::to_string(render_input_sample_rate_hz) + "_" + - std::to_string(render_output_sample_rate_hz) + "_" + - std::to_string(capture_input_sample_rate_hz) + "_" + - std::to_string(capture_output_sample_rate_hz) + "Hz_" + - std::to_string(num_channels) + "_channels" + "_total", - test_description, total_timer.GetDurationAverage(), - total_timer.GetDurationStandardDeviation(), "us", false); -} - -} // namespace - -// TODO(peah): Reactivate once issue 7712 has been resolved. -TEST(LevelControllerPerformanceTest, DISABLED_StandaloneProcessing) { - int sample_rates_to_test[] = { - AudioProcessing::kSampleRate8kHz, AudioProcessing::kSampleRate16kHz, - AudioProcessing::kSampleRate32kHz, AudioProcessing::kSampleRate48kHz}; - for (auto sample_rate : sample_rates_to_test) { - for (size_t num_channels = 1; num_channels <= 2; ++num_channels) { - RunStandaloneSubmodule(sample_rate, num_channels); - } - } -} - -void TestSomeSampleRatesWithApm(const std::string& test_name, - bool use_mobile_agc, - bool include_default_apm_processing) { - // Test some stereo combinations first. - size_t num_channels = 2; - RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate16kHz, - AudioProcessing::kSampleRate32kHz, num_channels, - use_mobile_agc, include_default_apm_processing); - RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz, - AudioProcessing::kSampleRate8kHz, num_channels, - use_mobile_agc, include_default_apm_processing); - RunTogetherWithApm(test_name, 48000, 48000, 44100, 44100, num_channels, - use_mobile_agc, include_default_apm_processing); - - // Then test mono combinations. - num_channels = 1; - RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz, - AudioProcessing::kSampleRate48kHz, num_channels, - use_mobile_agc, include_default_apm_processing); -} - -// TODO(peah): Reactivate once issue 7712 has been resolved. -#if !defined(WEBRTC_ANDROID) -TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) { -#else -TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) { -#endif - // Run without default APM processing and desktop AGC. - TestSomeSampleRatesWithApm("SimpleLevelControlViaApm", false, false); -} - -// TODO(peah): Reactivate once issue 7712 has been resolved. -#if !defined(WEBRTC_ANDROID) -TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) { -#else -TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) { -#endif - bool include_default_apm_processing = true; - TestSomeSampleRatesWithApm("LevelControlAndDefaultDesktopApm", false, - include_default_apm_processing); - TestSomeSampleRatesWithApm("LevelControlAndDefaultMobileApm", true, - include_default_apm_processing); -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/level_controller_constants.h b/modules/audio_processing/level_controller/level_controller_constants.h deleted file mode 100644 index 6cf2cd4c7e..0000000000 --- a/modules/audio_processing/level_controller/level_controller_constants.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_ - -namespace webrtc { - -const float kMaxLcGain = 10; -const float kMaxLcNoisePower = 100.f * 100.f; -const float kTargetLcPeakLevel = 16384.f; -const float kTargetLcPeakLeveldBFS = -6.0206f; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_ diff --git a/modules/audio_processing/level_controller/level_controller_unittest.cc b/modules/audio_processing/level_controller/level_controller_unittest.cc deleted file mode 100644 index cb36ae08f3..0000000000 --- a/modules/audio_processing/level_controller/level_controller_unittest.cc +++ /dev/null @@ -1,156 +0,0 @@ -/* - * Copyright (c) 2016 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 - -#include "api/array_view.h" -#include "api/optional.h" -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/include/audio_processing.h" -#include "modules/audio_processing/level_controller/level_controller.h" -#include "modules/audio_processing/test/audio_buffer_tools.h" -#include "modules/audio_processing/test/bitexactness_tools.h" -#include "test/gtest.h" - -namespace webrtc { -namespace { - -const int kNumFramesToProcess = 1000; - -// Processes a specified amount of frames, verifies the results and reports -// any errors. -void RunBitexactnessTest(int sample_rate_hz, - size_t num_channels, - rtc::Optional initial_peak_level_dbfs, - rtc::ArrayView output_reference) { - LevelController level_controller; - level_controller.Initialize(sample_rate_hz); - if (initial_peak_level_dbfs) { - AudioProcessing::Config::LevelController config; - config.initial_peak_level_dbfs = *initial_peak_level_dbfs; - level_controller.ApplyConfig(config); - } - - int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); - const StreamConfig capture_config(sample_rate_hz, num_channels, false); - AudioBuffer capture_buffer( - capture_config.num_frames(), capture_config.num_channels(), - capture_config.num_frames(), capture_config.num_channels(), - capture_config.num_frames()); - test::InputAudioFile capture_file( - test::GetApmCaptureTestVectorFileName(sample_rate_hz)); - std::vector capture_input(samples_per_channel * num_channels); - for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) { - ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels, - &capture_file, capture_input); - - test::CopyVectorToAudioBuffer(capture_config, capture_input, - &capture_buffer); - - level_controller.Process(&capture_buffer); - } - - // Extract test results. - std::vector capture_output; - test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer, - &capture_output); - - // Compare the output with the reference. Only the first values of the output - // from last frame processed are compared in order not having to specify all - // preceding frames as testvectors. As the algorithm being tested has a - // memory, testing only the last frame implicitly also tests the preceeding - // frames. - const float kVectorElementErrorBound = 1.0f / 32768.0f; - EXPECT_TRUE(test::VerifyDeinterleavedArray( - capture_config.num_frames(), capture_config.num_channels(), - output_reference, capture_output, kVectorElementErrorBound)); -} - -} // namespace - -TEST(LevelControllerConfig, ToString) { - AudioProcessing::Config config; - config.level_controller.enabled = true; - config.level_controller.initial_peak_level_dbfs = -6.0206f; - EXPECT_EQ("{enabled: true, initial_peak_level_dbfs: -6.0206}", - LevelController::ToString(config.level_controller)); - - config.level_controller.enabled = false; - config.level_controller.initial_peak_level_dbfs = -50.f; - EXPECT_EQ("{enabled: false, initial_peak_level_dbfs: -50}", - LevelController::ToString(config.level_controller)); -} - -TEST(LevelControlBitExactnessTest, Mono8kHz) { - const float kOutputReference[] = {-0.013939f, -0.012154f, -0.009054f}; - RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, Mono16kHz) { - const float kOutputReference[] = {-0.013706f, -0.013215f, -0.013018f}; - RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, Mono32kHz) { - const float kOutputReference[] = {-0.014495f, -0.016425f, -0.016085f}; - RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1, rtc::nullopt, - kOutputReference); -} - -// TODO(peah): Investigate why this particular testcase differ between Android -// and the rest of the platforms. -TEST(LevelControlBitExactnessTest, Mono48kHz) { -#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \ - defined(WEBRTC_ANDROID)) - const float kOutputReference[] = {-0.014277f, -0.015180f, -0.017437f}; -#else - const float kOutputReference[] = {-0.014306f, -0.015209f, -0.017466f}; -#endif - RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, Stereo8kHz) { - const float kOutputReference[] = {-0.014063f, -0.008450f, -0.012159f, - -0.051967f, -0.023202f, -0.047858f}; - RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, Stereo16kHz) { - const float kOutputReference[] = {-0.012714f, -0.005896f, -0.012220f, - -0.053306f, -0.024549f, -0.051527f}; - RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, Stereo32kHz) { - const float kOutputReference[] = {-0.011764f, -0.007044f, -0.013472f, - -0.053537f, -0.026322f, -0.056253f}; - RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, Stereo48kHz) { - const float kOutputReference[] = {-0.010643f, -0.006334f, -0.011377f, - -0.049088f, -0.023600f, -0.050465f}; - RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2, rtc::nullopt, - kOutputReference); -} - -TEST(LevelControlBitExactnessTest, MonoInitial48kHz) { - const float kOutputReference[] = {-0.013884f, -0.014761f, -0.016951f}; - RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, -50, - kOutputReference); -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/noise_level_estimator.cc b/modules/audio_processing/level_controller/noise_level_estimator.cc deleted file mode 100644 index abf4ea2cb1..0000000000 --- a/modules/audio_processing/level_controller/noise_level_estimator.cc +++ /dev/null @@ -1,72 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/noise_level_estimator.h" - -#include - -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" - -namespace webrtc { - -NoiseLevelEstimator::NoiseLevelEstimator() { - Initialize(AudioProcessing::kSampleRate48kHz); -} - -NoiseLevelEstimator::~NoiseLevelEstimator() {} - -void NoiseLevelEstimator::Initialize(int sample_rate_hz) { - noise_energy_ = 1.f; - first_update_ = true; - min_noise_energy_ = sample_rate_hz * 2.f * 2.f / 100.f; - noise_energy_hold_counter_ = 0; -} - -float NoiseLevelEstimator::Analyze(SignalClassifier::SignalType signal_type, - float frame_energy) { - if (frame_energy <= 0.f) { - return noise_energy_; - } - - if (first_update_) { - // Initialize the noise energy to the frame energy. - first_update_ = false; - return noise_energy_ = std::max(frame_energy, min_noise_energy_); - } - - // Update the noise estimate in a minimum statistics-type manner. - if (signal_type == SignalClassifier::SignalType::kStationary) { - if (frame_energy > noise_energy_) { - // Leak the estimate upwards towards the frame energy if no recent - // downward update. - noise_energy_hold_counter_ = std::max(noise_energy_hold_counter_ - 1, 0); - - if (noise_energy_hold_counter_ == 0) { - noise_energy_ = std::min(noise_energy_ * 1.01f, frame_energy); - } - } else { - // Update smoothly downwards with a limited maximum update magnitude. - noise_energy_ = - std::max(noise_energy_ * 0.9f, - noise_energy_ + 0.05f * (frame_energy - noise_energy_)); - noise_energy_hold_counter_ = 1000; - } - } else { - // For a non-stationary signal, leak the estimate downwards in order to - // avoid estimate locking due to incorrect signal classification. - noise_energy_ = noise_energy_ * 0.99f; - } - - // Ensure a minimum of the estimate. - return noise_energy_ = std::max(noise_energy_, min_noise_energy_); -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/noise_level_estimator.h b/modules/audio_processing/level_controller/noise_level_estimator.h deleted file mode 100644 index 94ef6737e7..0000000000 --- a/modules/audio_processing/level_controller/noise_level_estimator.h +++ /dev/null @@ -1,37 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_ - -#include "modules/audio_processing/level_controller/signal_classifier.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class NoiseLevelEstimator { - public: - NoiseLevelEstimator(); - ~NoiseLevelEstimator(); - void Initialize(int sample_rate_hz); - float Analyze(SignalClassifier::SignalType signal_type, float frame_energy); - - private: - float min_noise_energy_ = 0.f; - bool first_update_; - float noise_energy_; - int noise_energy_hold_counter_; - - RTC_DISALLOW_COPY_AND_ASSIGN(NoiseLevelEstimator); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_ diff --git a/modules/audio_processing/level_controller/noise_spectrum_estimator.cc b/modules/audio_processing/level_controller/noise_spectrum_estimator.cc deleted file mode 100644 index 6e921c24d1..0000000000 --- a/modules/audio_processing/level_controller/noise_spectrum_estimator.cc +++ /dev/null @@ -1,68 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/noise_spectrum_estimator.h" - -#include -#include - -#include "api/array_view.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" -#include "rtc_base/arraysize.h" - -namespace webrtc { -namespace { -constexpr float kMinNoisePower = 100.f; -} // namespace - -NoiseSpectrumEstimator::NoiseSpectrumEstimator(ApmDataDumper* data_dumper) - : data_dumper_(data_dumper) { - Initialize(); -} - -void NoiseSpectrumEstimator::Initialize() { - std::fill(noise_spectrum_, noise_spectrum_ + arraysize(noise_spectrum_), - kMinNoisePower); -} - -void NoiseSpectrumEstimator::Update(rtc::ArrayView spectrum, - bool first_update) { - RTC_DCHECK_EQ(65, spectrum.size()); - - if (first_update) { - // Initialize the noise spectral estimate with the signal spectrum. - std::copy(spectrum.data(), spectrum.data() + spectrum.size(), - noise_spectrum_); - } else { - // Smoothly update the noise spectral estimate towards the signal spectrum - // such that the magnitude of the updates are limited. - for (size_t k = 0; k < spectrum.size(); ++k) { - if (noise_spectrum_[k] < spectrum[k]) { - noise_spectrum_[k] = std::min( - 1.01f * noise_spectrum_[k], - noise_spectrum_[k] + 0.05f * (spectrum[k] - noise_spectrum_[k])); - } else { - noise_spectrum_[k] = std::max( - 0.99f * noise_spectrum_[k], - noise_spectrum_[k] + 0.05f * (spectrum[k] - noise_spectrum_[k])); - } - } - } - - // Ensure that the noise spectal estimate does not become too low. - for (auto& v : noise_spectrum_) { - v = std::max(v, kMinNoisePower); - } - - data_dumper_->DumpRaw("lc_noise_spectrum", 65, noise_spectrum_); - data_dumper_->DumpRaw("lc_signal_spectrum", spectrum); -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/noise_spectrum_estimator.h b/modules/audio_processing/level_controller/noise_spectrum_estimator.h deleted file mode 100644 index f10933ec96..0000000000 --- a/modules/audio_processing/level_controller/noise_spectrum_estimator.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_ - -#include "api/array_view.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class ApmDataDumper; - -class NoiseSpectrumEstimator { - public: - explicit NoiseSpectrumEstimator(ApmDataDumper* data_dumper); - void Initialize(); - void Update(rtc::ArrayView spectrum, bool first_update); - - rtc::ArrayView GetNoiseSpectrum() const { - return rtc::ArrayView(noise_spectrum_); - } - - private: - ApmDataDumper* data_dumper_; - float noise_spectrum_[65]; - - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSpectrumEstimator); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_ diff --git a/modules/audio_processing/level_controller/peak_level_estimator.cc b/modules/audio_processing/level_controller/peak_level_estimator.cc deleted file mode 100644 index f602892600..0000000000 --- a/modules/audio_processing/level_controller/peak_level_estimator.cc +++ /dev/null @@ -1,74 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/peak_level_estimator.h" - -#include - -#include "common_audio/include/audio_util.h" -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" - -namespace webrtc { -namespace { - -constexpr float kMinLevel = 30.f; - -} // namespace - -PeakLevelEstimator::PeakLevelEstimator(float initial_peak_level_dbfs) { - Initialize(initial_peak_level_dbfs); -} - -PeakLevelEstimator::~PeakLevelEstimator() {} - -void PeakLevelEstimator::Initialize(float initial_peak_level_dbfs) { - RTC_DCHECK_LE(-100.f, initial_peak_level_dbfs); - RTC_DCHECK_GE(0.f, initial_peak_level_dbfs); - - peak_level_ = std::max(DbfsToFloatS16(initial_peak_level_dbfs), kMinLevel); - - hold_counter_ = 0; - initialization_phase_ = true; -} - -float PeakLevelEstimator::Analyze(SignalClassifier::SignalType signal_type, - float frame_peak_level) { - if (frame_peak_level == 0) { - RTC_DCHECK_LE(kMinLevel, peak_level_); - return peak_level_; - } - - if (peak_level_ < frame_peak_level) { - // Smoothly update the estimate upwards when the frame peak level is - // higher than the estimate. - peak_level_ += 0.1f * (frame_peak_level - peak_level_); - hold_counter_ = 100; - initialization_phase_ = false; - } else { - hold_counter_ = std::max(0, hold_counter_ - 1); - - // When the signal is highly non-stationary, update the estimate slowly - // downwards if the estimate is lower than the frame peak level. - if ((signal_type == SignalClassifier::SignalType::kHighlyNonStationary && - hold_counter_ == 0) || - initialization_phase_) { - peak_level_ = - std::max(peak_level_ + 0.01f * (frame_peak_level - peak_level_), - peak_level_ * 0.995f); - } - } - - peak_level_ = std::max(peak_level_, kMinLevel); - - return peak_level_; -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/peak_level_estimator.h b/modules/audio_processing/level_controller/peak_level_estimator.h deleted file mode 100644 index 0aa55d2d55..0000000000 --- a/modules/audio_processing/level_controller/peak_level_estimator.h +++ /dev/null @@ -1,37 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_ - -#include "modules/audio_processing/level_controller/level_controller_constants.h" -#include "modules/audio_processing/level_controller/signal_classifier.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class PeakLevelEstimator { - public: - explicit PeakLevelEstimator(float initial_peak_level_dbfs); - ~PeakLevelEstimator(); - void Initialize(float initial_peak_level_dbfs); - float Analyze(SignalClassifier::SignalType signal_type, - float frame_peak_level); - private: - float peak_level_; - int hold_counter_; - bool initialization_phase_; - - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PeakLevelEstimator); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_ diff --git a/modules/audio_processing/level_controller/saturating_gain_estimator.cc b/modules/audio_processing/level_controller/saturating_gain_estimator.cc deleted file mode 100644 index 60110c684b..0000000000 --- a/modules/audio_processing/level_controller/saturating_gain_estimator.cc +++ /dev/null @@ -1,48 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/saturating_gain_estimator.h" - -#include -#include - -#include "modules/audio_processing/level_controller/level_controller_constants.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" - -namespace webrtc { - -SaturatingGainEstimator::SaturatingGainEstimator() { - Initialize(); -} - -SaturatingGainEstimator::~SaturatingGainEstimator() {} - -void SaturatingGainEstimator::Initialize() { - saturating_gain_ = kMaxLcGain; - saturating_gain_hold_counter_ = 0; -} - -void SaturatingGainEstimator::Update(float gain, int num_saturations) { - bool too_many_saturations = (num_saturations > 2); - - if (too_many_saturations) { - saturating_gain_ = 0.95f * gain; - saturating_gain_hold_counter_ = 1000; - } else { - saturating_gain_hold_counter_ = - std::max(0, saturating_gain_hold_counter_ - 1); - if (saturating_gain_hold_counter_ == 0) { - saturating_gain_ *= 1.001f; - saturating_gain_ = std::min(kMaxLcGain, saturating_gain_); - } - } -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/saturating_gain_estimator.h b/modules/audio_processing/level_controller/saturating_gain_estimator.h deleted file mode 100644 index 8980f4ef97..0000000000 --- a/modules/audio_processing/level_controller/saturating_gain_estimator.h +++ /dev/null @@ -1,37 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_ - -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class ApmDataDumper; - -class SaturatingGainEstimator { - public: - SaturatingGainEstimator(); - ~SaturatingGainEstimator(); - void Initialize(); - void Update(float gain, int num_saturations); - float GetGain() const { return saturating_gain_; } - - private: - float saturating_gain_; - int saturating_gain_hold_counter_; - - RTC_DISALLOW_COPY_AND_ASSIGN(SaturatingGainEstimator); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_ diff --git a/modules/audio_processing/level_controller/signal_classifier.cc b/modules/audio_processing/level_controller/signal_classifier.cc deleted file mode 100644 index d2d5917387..0000000000 --- a/modules/audio_processing/level_controller/signal_classifier.cc +++ /dev/null @@ -1,171 +0,0 @@ -/* - * Copyright (c) 2016 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/level_controller/signal_classifier.h" - -#include -#include -#include - -#include "api/array_view.h" -#include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/level_controller/down_sampler.h" -#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { -namespace { - -void RemoveDcLevel(rtc::ArrayView x) { - RTC_DCHECK_LT(0, x.size()); - float mean = std::accumulate(x.data(), x.data() + x.size(), 0.f); - mean /= x.size(); - - for (float& v : x) { - v -= mean; - } -} - -void PowerSpectrum(const OouraFft* ooura_fft, - rtc::ArrayView x, - rtc::ArrayView spectrum) { - RTC_DCHECK_EQ(65, spectrum.size()); - RTC_DCHECK_EQ(128, x.size()); - float X[128]; - std::copy(x.data(), x.data() + x.size(), X); - ooura_fft->Fft(X); - - float* X_p = X; - RTC_DCHECK_EQ(X_p, &X[0]); - spectrum[0] = (*X_p) * (*X_p); - ++X_p; - RTC_DCHECK_EQ(X_p, &X[1]); - spectrum[64] = (*X_p) * (*X_p); - for (int k = 1; k < 64; ++k) { - ++X_p; - RTC_DCHECK_EQ(X_p, &X[2 * k]); - spectrum[k] = (*X_p) * (*X_p); - ++X_p; - RTC_DCHECK_EQ(X_p, &X[2 * k + 1]); - spectrum[k] += (*X_p) * (*X_p); - } -} - -webrtc::SignalClassifier::SignalType ClassifySignal( - rtc::ArrayView signal_spectrum, - rtc::ArrayView noise_spectrum, - ApmDataDumper* data_dumper) { - int num_stationary_bands = 0; - int num_highly_nonstationary_bands = 0; - - // Detect stationary and highly nonstationary bands. - for (size_t k = 1; k < 40; k++) { - if (signal_spectrum[k] < 3 * noise_spectrum[k] && - signal_spectrum[k] * 3 > noise_spectrum[k]) { - ++num_stationary_bands; - } else if (signal_spectrum[k] > 9 * noise_spectrum[k]) { - ++num_highly_nonstationary_bands; - } - } - - data_dumper->DumpRaw("lc_num_stationary_bands", 1, &num_stationary_bands); - data_dumper->DumpRaw("lc_num_highly_nonstationary_bands", 1, - &num_highly_nonstationary_bands); - - // Use the detected number of bands to classify the overall signal - // stationarity. - if (num_stationary_bands > 15) { - return SignalClassifier::SignalType::kStationary; - } else if (num_highly_nonstationary_bands > 15) { - return SignalClassifier::SignalType::kHighlyNonStationary; - } else { - return SignalClassifier::SignalType::kNonStationary; - } -} - -} // namespace - -SignalClassifier::FrameExtender::FrameExtender(size_t frame_size, - size_t extended_frame_size) - : x_old_(extended_frame_size - frame_size, 0.f) {} - -SignalClassifier::FrameExtender::~FrameExtender() = default; - -void SignalClassifier::FrameExtender::ExtendFrame( - rtc::ArrayView x, - rtc::ArrayView x_extended) { - RTC_DCHECK_EQ(x_old_.size() + x.size(), x_extended.size()); - std::copy(x_old_.data(), x_old_.data() + x_old_.size(), x_extended.data()); - std::copy(x.data(), x.data() + x.size(), x_extended.data() + x_old_.size()); - std::copy(x_extended.data() + x_extended.size() - x_old_.size(), - x_extended.data() + x_extended.size(), x_old_.data()); -} - -SignalClassifier::SignalClassifier(ApmDataDumper* data_dumper) - : data_dumper_(data_dumper), - down_sampler_(data_dumper_), - noise_spectrum_estimator_(data_dumper_) { - Initialize(AudioProcessing::kSampleRate48kHz); -} -SignalClassifier::~SignalClassifier() {} - -void SignalClassifier::Initialize(int sample_rate_hz) { - down_sampler_.Initialize(sample_rate_hz); - noise_spectrum_estimator_.Initialize(); - frame_extender_.reset(new FrameExtender(80, 128)); - sample_rate_hz_ = sample_rate_hz; - initialization_frames_left_ = 2; - consistent_classification_counter_ = 3; - last_signal_type_ = SignalClassifier::SignalType::kNonStationary; -} - -void SignalClassifier::Analyze(const AudioBuffer& audio, - SignalType* signal_type) { - RTC_DCHECK_EQ(audio.num_frames(), sample_rate_hz_ / 100); - - // Compute the signal power spectrum. - float downsampled_frame[80]; - down_sampler_.DownSample(rtc::ArrayView( - audio.channels_const_f()[0], audio.num_frames()), - downsampled_frame); - float extended_frame[128]; - frame_extender_->ExtendFrame(downsampled_frame, extended_frame); - RemoveDcLevel(extended_frame); - float signal_spectrum[65]; - PowerSpectrum(&ooura_fft_, extended_frame, signal_spectrum); - - // Classify the signal based on the estimate of the noise spectrum and the - // signal spectrum estimate. - *signal_type = ClassifySignal(signal_spectrum, - noise_spectrum_estimator_.GetNoiseSpectrum(), - data_dumper_); - - // Update the noise spectrum based on the signal spectrum. - noise_spectrum_estimator_.Update(signal_spectrum, - initialization_frames_left_ > 0); - - // Update the number of frames until a reliable signal spectrum is achieved. - initialization_frames_left_ = std::max(0, initialization_frames_left_ - 1); - - if (last_signal_type_ == *signal_type) { - consistent_classification_counter_ = - std::max(0, consistent_classification_counter_ - 1); - } else { - last_signal_type_ = *signal_type; - consistent_classification_counter_ = 3; - } - - if (consistent_classification_counter_ > 0) { - *signal_type = SignalClassifier::SignalType::kNonStationary; - } -} - -} // namespace webrtc diff --git a/modules/audio_processing/level_controller/signal_classifier.h b/modules/audio_processing/level_controller/signal_classifier.h deleted file mode 100644 index 2be13fef7a..0000000000 --- a/modules/audio_processing/level_controller/signal_classifier.h +++ /dev/null @@ -1,67 +0,0 @@ -/* - * Copyright (c) 2016 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. - */ - -#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_ -#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_ - -#include -#include - -#include "api/array_view.h" -#include "modules/audio_processing/level_controller/down_sampler.h" -#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h" -#include "modules/audio_processing/utility/ooura_fft.h" -#include "rtc_base/constructormagic.h" - -namespace webrtc { - -class ApmDataDumper; -class AudioBuffer; - -class SignalClassifier { - public: - enum class SignalType { kHighlyNonStationary, kNonStationary, kStationary }; - - explicit SignalClassifier(ApmDataDumper* data_dumper); - ~SignalClassifier(); - - void Initialize(int sample_rate_hz); - void Analyze(const AudioBuffer& audio, SignalType* signal_type); - - private: - class FrameExtender { - public: - FrameExtender(size_t frame_size, size_t extended_frame_size); - ~FrameExtender(); - - void ExtendFrame(rtc::ArrayView x, - rtc::ArrayView x_extended); - - private: - std::vector x_old_; - - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender); - }; - - ApmDataDumper* const data_dumper_; - DownSampler down_sampler_; - std::unique_ptr frame_extender_; - NoiseSpectrumEstimator noise_spectrum_estimator_; - int sample_rate_hz_; - int initialization_frames_left_; - int consistent_classification_counter_; - SignalType last_signal_type_; - const OouraFft ooura_fft_; - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_ diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc index 6d0b07c7ed..83e85314ad 100644 --- a/modules/audio_processing/test/aec_dump_based_simulator.cc +++ b/modules/audio_processing/test/aec_dump_based_simulator.cc @@ -473,10 +473,6 @@ void AecDumpBasedSimulator::HandleMessage( new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter)); } - if (settings_.use_lc) { - apm_config.level_controller.enabled = *settings_.use_lc; - } - if (settings_.use_ed) { apm_config.residual_echo_detector.enabled = *settings_.use_ed; } diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc index 82bffe427f..b4c352589e 100644 --- a/modules/audio_processing/test/audio_processing_simulator.cc +++ b/modules/audio_processing/test/audio_processing_simulator.cc @@ -328,9 +328,6 @@ void AudioProcessingSimulator::CreateAudioProcessor() { if (settings_.use_aec3 && *settings_.use_aec3) { echo_control_factory.reset(new EchoCanceller3Factory()); } - if (settings_.use_lc) { - apm_config.level_controller.enabled = *settings_.use_lc; - } if (settings_.use_hpf) { apm_config.high_pass_filter.enabled = *settings_.use_hpf; } diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h index 41a3f45106..a6bdb9057e 100644 --- a/modules/audio_processing/test/audio_processing_simulator.h +++ b/modules/audio_processing/test/audio_processing_simulator.h @@ -66,7 +66,6 @@ struct SimulationSettings { rtc::Optional use_extended_filter; rtc::Optional use_drift_compensation; rtc::Optional use_aec3; - rtc::Optional use_lc; rtc::Optional use_experimental_agc; rtc::Optional aecm_routing_mode; rtc::Optional use_aecm_comfort_noise; diff --git a/modules/audio_processing/test/audioproc_float.cc b/modules/audio_processing/test/audioproc_float.cc index c5229a4e10..554d6b405e 100644 --- a/modules/audio_processing/test/audioproc_float.cc +++ b/modules/audio_processing/test/audioproc_float.cc @@ -121,9 +121,6 @@ DEFINE_int(drift_compensation, DEFINE_int(aec3, kParameterNotSpecifiedValue, "Activate (1) or deactivate(0) the experimental AEC mode AEC3"); -DEFINE_int(lc, - kParameterNotSpecifiedValue, - "Activate (1) or deactivate(0) the level control"); DEFINE_int(experimental_agc, kParameterNotSpecifiedValue, "Activate (1) or deactivate(0) the experimental AGC"); @@ -261,7 +258,6 @@ SimulationSettings CreateSettings() { &settings.use_refined_adaptive_filter); SetSettingIfFlagSet(FLAG_aec3, &settings.use_aec3); - SetSettingIfFlagSet(FLAG_lc, &settings.use_lc); SetSettingIfFlagSet(FLAG_experimental_agc, &settings.use_experimental_agc); SetSettingIfSpecified(FLAG_aecm_routing_mode, &settings.aecm_routing_mode); SetSettingIfFlagSet(FLAG_aecm_comfort_noise, diff --git a/modules/audio_processing/test/debug_dump_test.cc b/modules/audio_processing/test/debug_dump_test.cc index 56f47b00fa..4d3be48684 100644 --- a/modules/audio_processing/test/debug_dump_test.cc +++ b/modules/audio_processing/test/debug_dump_test.cc @@ -484,31 +484,6 @@ TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) { } } -TEST_F(DebugDumpTest, VerifyLevelControllerExperimentalString) { - Config config; - AudioProcessing::Config apm_config; - apm_config.level_controller.enabled = true; - DebugDumpGenerator generator(config, apm_config); - generator.StartRecording(); - generator.Process(100); - generator.StopRecording(); - - DebugDumpReplayer debug_dump_replayer_; - - ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name())); - - while (const rtc::Optional event = - debug_dump_replayer_.GetNextEvent()) { - debug_dump_replayer_.RunNextEvent(); - if (event->type() == audioproc::Event::CONFIG) { - const audioproc::Config* msg = &event->config(); - ASSERT_TRUE(msg->has_experiments_description()); - EXPECT_PRED_FORMAT2(testing::IsSubstring, "LevelController", - msg->experiments_description().c_str()); - } - } -} - TEST_F(DebugDumpTest, VerifyAgcClippingLevelExperimentalString) { Config config; // Arbitrarily set clipping gain to 17, which will never be the default.