| /* |
| * Copyright (c) 2018 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/agc2/adaptive_agc.h" |
| |
| #include "common_audio/include/audio_util.h" |
| #include "modules/audio_processing/agc2/cpu_features.h" |
| #include "modules/audio_processing/agc2/vad_with_level.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| using AdaptiveDigitalConfig = |
| AudioProcessing::Config::GainController2::AdaptiveDigital; |
| using NoiseEstimatorType = |
| AudioProcessing::Config::GainController2::NoiseEstimator; |
| |
| // Detects the available CPU features and applies any kill-switches. |
| AvailableCpuFeatures GetAllowedCpuFeatures( |
| const AdaptiveDigitalConfig& config) { |
| AvailableCpuFeatures features = GetAvailableCpuFeatures(); |
| if (!config.sse2_allowed) { |
| features.sse2 = false; |
| } |
| if (!config.avx2_allowed) { |
| features.avx2 = false; |
| } |
| if (!config.neon_allowed) { |
| features.neon = false; |
| } |
| return features; |
| } |
| |
| std::unique_ptr<NoiseLevelEstimator> CreateNoiseLevelEstimator( |
| NoiseEstimatorType estimator_type, |
| ApmDataDumper* apm_data_dumper) { |
| switch (estimator_type) { |
| case NoiseEstimatorType::kStationaryNoise: |
| return CreateStationaryNoiseEstimator(apm_data_dumper); |
| case NoiseEstimatorType::kNoiseFloor: |
| return CreateNoiseFloorEstimator(apm_data_dumper); |
| } |
| } |
| |
| } // namespace |
| |
| AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper, |
| const AdaptiveDigitalConfig& config) |
| : speech_level_estimator_(apm_data_dumper, |
| config.adjacent_speech_frames_threshold), |
| vad_(config.vad_reset_period_ms, GetAllowedCpuFeatures(config)), |
| gain_controller_(apm_data_dumper, |
| config.adjacent_speech_frames_threshold, |
| config.max_gain_change_db_per_second, |
| config.max_output_noise_level_dbfs, |
| config.dry_run), |
| apm_data_dumper_(apm_data_dumper), |
| noise_level_estimator_( |
| CreateNoiseLevelEstimator(config.noise_estimator, apm_data_dumper)), |
| saturation_protector_( |
| CreateSaturationProtector(kSaturationProtectorInitialHeadroomDb, |
| kSaturationProtectorExtraHeadroomDb, |
| config.adjacent_speech_frames_threshold, |
| apm_data_dumper)) { |
| RTC_DCHECK(apm_data_dumper); |
| RTC_DCHECK(noise_level_estimator_); |
| RTC_DCHECK(saturation_protector_); |
| if (!config.use_saturation_protector) { |
| RTC_LOG(LS_WARNING) << "The saturation protector cannot be disabled."; |
| } |
| } |
| |
| AdaptiveAgc::~AdaptiveAgc() = default; |
| |
| void AdaptiveAgc::Initialize(int sample_rate_hz, int num_channels) { |
| gain_controller_.Initialize(sample_rate_hz, num_channels); |
| } |
| |
| void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) { |
| AdaptiveDigitalGainApplier::FrameInfo info; |
| |
| VadLevelAnalyzer::Result vad_result = vad_.AnalyzeFrame(frame); |
| info.speech_probability = vad_result.speech_probability; |
| apm_data_dumper_->DumpRaw("agc2_speech_probability", |
| vad_result.speech_probability); |
| apm_data_dumper_->DumpRaw("agc2_input_rms_dbfs", vad_result.rms_dbfs); |
| apm_data_dumper_->DumpRaw("agc2_input_peak_dbfs", vad_result.peak_dbfs); |
| |
| speech_level_estimator_.Update(vad_result); |
| info.speech_level_dbfs = speech_level_estimator_.level_dbfs(); |
| info.speech_level_reliable = speech_level_estimator_.IsConfident(); |
| apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", info.speech_level_dbfs); |
| apm_data_dumper_->DumpRaw("agc2_speech_level_reliable", |
| info.speech_level_reliable); |
| |
| info.noise_rms_dbfs = noise_level_estimator_->Analyze(frame); |
| apm_data_dumper_->DumpRaw("agc2_noise_rms_dbfs", info.noise_rms_dbfs); |
| |
| saturation_protector_->Analyze(info.speech_probability, vad_result.peak_dbfs, |
| info.speech_level_dbfs); |
| info.headroom_db = saturation_protector_->HeadroomDb(); |
| apm_data_dumper_->DumpRaw("agc2_headroom_db", info.headroom_db); |
| |
| info.limiter_envelope_dbfs = FloatS16ToDbfs(limiter_envelope); |
| apm_data_dumper_->DumpRaw("agc2_limiter_envelope_dbfs", |
| info.limiter_envelope_dbfs); |
| |
| gain_controller_.Process(info, frame); |
| } |
| |
| void AdaptiveAgc::HandleInputGainChange() { |
| speech_level_estimator_.Reset(); |
| saturation_protector_->Reset(); |
| } |
| |
| } // namespace webrtc |