118 lines
4.3 KiB
C++
118 lines
4.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/gain_controller2.h"
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/atomicops.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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int GainController2::instance_count_ = 0;
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GainController2::GainController2()
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: data_dumper_(
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new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
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gain_applier_(/*hard_clip_samples=*/false,
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/*initial_gain_factor=*/0.f),
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adaptive_agc_(new AdaptiveAgc(data_dumper_.get())),
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limiter_(static_cast<size_t>(48000), data_dumper_.get(), "Agc2") {}
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GainController2::~GainController2() = default;
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void GainController2::Initialize(int sample_rate_hz) {
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RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate48kHz);
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limiter_.SetSampleRate(sample_rate_hz);
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data_dumper_->InitiateNewSetOfRecordings();
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data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
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}
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void GainController2::Process(AudioBuffer* audio) {
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AudioFrameView<float> float_frame(audio->channels_f(), audio->num_channels(),
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audio->num_frames());
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// Apply fixed gain first, then the adaptive one.
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gain_applier_.ApplyGain(float_frame);
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if (adaptive_digital_mode_) {
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adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
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}
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limiter_.Process(float_frame);
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}
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void GainController2::NotifyAnalogLevel(int level) {
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if (analog_level_ != level && adaptive_digital_mode_) {
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adaptive_agc_->Reset();
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}
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analog_level_ = level;
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}
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void GainController2::ApplyConfig(
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const AudioProcessing::Config::GainController2& config) {
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RTC_DCHECK(Validate(config))
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<< " the invalid config was " << ToString(config);
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config_ = config;
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if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) {
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// Reset the limiter to quickly react on abrupt level changes caused by
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// large changes of the fixed gain.
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limiter_.Reset();
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}
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gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
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adaptive_digital_mode_ = config_.adaptive_digital.enabled;
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adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
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}
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bool GainController2::Validate(
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const AudioProcessing::Config::GainController2& config) {
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return config.fixed_digital.gain_db >= 0.f &&
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config.fixed_digital.gain_db < 50.f &&
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config.adaptive_digital.extra_saturation_margin_db >= 0.f &&
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config.adaptive_digital.extra_saturation_margin_db <= 100.f;
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}
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std::string GainController2::ToString(
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const AudioProcessing::Config::GainController2& config) {
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rtc::StringBuilder ss;
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std::string adaptive_digital_level_estimator;
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using LevelEstimatorType =
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AudioProcessing::Config::GainController2::LevelEstimator;
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switch (config.adaptive_digital.level_estimator) {
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case LevelEstimatorType::kRms:
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adaptive_digital_level_estimator = "RMS";
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break;
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case LevelEstimatorType::kPeak:
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adaptive_digital_level_estimator = "peak";
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break;
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}
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// clang-format off
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// clang formatting doesn't respect custom nested style.
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ss << "{"
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<< "enabled: " << (config.enabled ? "true" : "false") << ", "
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<< "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, "
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<< "adaptive_digital: {"
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<< "enabled: "
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<< (config.adaptive_digital.enabled ? "true" : "false") << ", "
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<< "level_estimator: " << adaptive_digital_level_estimator << ", "
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<< "extra_saturation_margin_db:"
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<< config.adaptive_digital.extra_saturation_margin_db << "}"
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<< "}";
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// clang-format on
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return ss.Release();
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}
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} // namespace webrtc
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