| /* |
| * 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 { |
| |
| void DumpDebugData(const AdaptiveDigitalGainApplier::FrameInfo& info, |
| ApmDataDumper& dumper) { |
| dumper.DumpRaw("agc2_vad_probability", info.vad_result.speech_probability); |
| dumper.DumpRaw("agc2_vad_rms_dbfs", info.vad_result.rms_dbfs); |
| dumper.DumpRaw("agc2_vad_peak_dbfs", info.vad_result.peak_dbfs); |
| dumper.DumpRaw("agc2_noise_estimate_dbfs", info.input_noise_level_dbfs); |
| dumper.DumpRaw("agc2_last_limiter_audio_level", info.limiter_envelope_dbfs); |
| } |
| |
| constexpr int kGainApplierAdjacentSpeechFramesThreshold = 1; |
| constexpr float kMaxGainChangePerSecondDb = 3.0f; |
| constexpr float kMaxOutputNoiseLevelDbfs = -50.0f; |
| |
| // Detects the available CPU features and applies any kill-switches. |
| AvailableCpuFeatures GetAllowedCpuFeatures( |
| const AudioProcessing::Config::GainController2::AdaptiveDigital& 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; |
| } |
| |
| } // namespace |
| |
| AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper) |
| : speech_level_estimator_(apm_data_dumper), |
| gain_applier_(apm_data_dumper, |
| kGainApplierAdjacentSpeechFramesThreshold, |
| kMaxGainChangePerSecondDb, |
| kMaxOutputNoiseLevelDbfs), |
| apm_data_dumper_(apm_data_dumper), |
| noise_level_estimator_(apm_data_dumper) { |
| RTC_DCHECK(apm_data_dumper); |
| } |
| |
| AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper, |
| const AudioProcessing::Config::GainController2& config) |
| : speech_level_estimator_( |
| apm_data_dumper, |
| config.adaptive_digital.level_estimator, |
| config.adaptive_digital |
| .level_estimator_adjacent_speech_frames_threshold, |
| config.adaptive_digital.initial_saturation_margin_db, |
| config.adaptive_digital.extra_saturation_margin_db), |
| vad_(config.adaptive_digital.vad_reset_period_ms, |
| config.adaptive_digital.vad_probability_attack, |
| GetAllowedCpuFeatures(config.adaptive_digital)), |
| gain_applier_( |
| apm_data_dumper, |
| config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold, |
| config.adaptive_digital.max_gain_change_db_per_second, |
| config.adaptive_digital.max_output_noise_level_dbfs), |
| apm_data_dumper_(apm_data_dumper), |
| noise_level_estimator_(apm_data_dumper) { |
| RTC_DCHECK(apm_data_dumper); |
| if (!config.adaptive_digital.use_saturation_protector) { |
| RTC_LOG(LS_WARNING) << "The saturation protector cannot be disabled."; |
| } |
| } |
| |
| AdaptiveAgc::~AdaptiveAgc() = default; |
| |
| void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) { |
| AdaptiveDigitalGainApplier::FrameInfo info; |
| info.vad_result = vad_.AnalyzeFrame(frame); |
| speech_level_estimator_.Update(info.vad_result); |
| info.input_level_dbfs = speech_level_estimator_.level_dbfs(); |
| info.input_noise_level_dbfs = noise_level_estimator_.Analyze(frame); |
| info.limiter_envelope_dbfs = |
| limiter_envelope > 0 ? FloatS16ToDbfs(limiter_envelope) : -90.0f; |
| info.estimate_is_confident = speech_level_estimator_.IsConfident(); |
| DumpDebugData(info, *apm_data_dumper_); |
| gain_applier_.Process(info, frame); |
| } |
| |
| void AdaptiveAgc::Reset() { |
| speech_level_estimator_.Reset(); |
| } |
| |
| } // namespace webrtc |