/* * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #include "modules/audio_processing/gain_controller2.h" #include "common_audio/include/audio_util.h" #include "modules/audio_processing/audio_buffer.h" #include "modules/audio_processing/include/audio_frame_view.h" #include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/atomicops.h" #include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" namespace webrtc { int GainController2::instance_count_ = 0; GainController2::GainController2() : data_dumper_( new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))), gain_applier_(/*hard_clip_samples=*/false, /*initial_gain_factor=*/0.f), adaptive_agc_(new AdaptiveAgc(data_dumper_.get())), limiter_(static_cast(48000), data_dumper_.get(), "Agc2") {} GainController2::~GainController2() = default; void GainController2::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); limiter_.SetSampleRate(sample_rate_hz); data_dumper_->InitiateNewSetOfRecordings(); data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz); } void GainController2::Process(AudioBuffer* audio) { AudioFrameView float_frame(audio->channels_f(), audio->num_channels(), audio->num_frames()); // Apply fixed gain first, then the adaptive one. gain_applier_.ApplyGain(float_frame); if (adaptive_digital_mode_) { adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel()); } limiter_.Process(float_frame); } void GainController2::NotifyAnalogLevel(int level) { if (analog_level_ != level && adaptive_digital_mode_) { adaptive_agc_->Reset(); } analog_level_ = level; } void GainController2::ApplyConfig( const AudioProcessing::Config::GainController2& config) { RTC_DCHECK(Validate(config)) << " the invalid config was " << ToString(config); config_ = config; if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) { // Reset the limiter to quickly react on abrupt level changes caused by // large changes of the fixed gain. limiter_.Reset(); } gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db)); adaptive_digital_mode_ = config_.adaptive_digital.enabled; adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_)); } bool GainController2::Validate( const AudioProcessing::Config::GainController2& config) { return config.fixed_digital.gain_db >= 0.f && config.fixed_digital.gain_db < 50.f && config.adaptive_digital.extra_saturation_margin_db >= 0.f && config.adaptive_digital.extra_saturation_margin_db <= 100.f; } std::string GainController2::ToString( const AudioProcessing::Config::GainController2& config) { rtc::StringBuilder ss; std::string adaptive_digital_level_estimator; using LevelEstimatorType = AudioProcessing::Config::GainController2::LevelEstimator; switch (config.adaptive_digital.level_estimator) { case LevelEstimatorType::kRms: adaptive_digital_level_estimator = "RMS"; break; case LevelEstimatorType::kPeak: adaptive_digital_level_estimator = "peak"; break; } // clang-format off // clang formatting doesn't respect custom nested style. ss << "{" << "enabled: " << (config.enabled ? "true" : "false") << ", " << "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, " << "adaptive_digital: {" << "enabled: " << (config.adaptive_digital.enabled ? "true" : "false") << ", " << "level_estimator: " << adaptive_digital_level_estimator << ", " << "extra_saturation_margin_db:" << config.adaptive_digital.extra_saturation_margin_db << "}" << "}"; // clang-format on return ss.Release(); } } // namespace webrtc