blob: f256be04f4fb3817a13660e2bf1f6c7423e68122 [file] [log] [blame]
/*
* 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<size_t>(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> 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));
config_ = config;
if (gain_applier_.GetGainFactor() != config_.fixed_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_gain_db));
adaptive_digital_mode_ = config_.adaptive_digital_mode;
adaptive_agc_.reset(
new AdaptiveAgc(data_dumper_.get(), config_.extra_saturation_margin_db));
}
bool GainController2::Validate(
const AudioProcessing::Config::GainController2& config) {
return config.fixed_gain_db >= 0.f &&
config.extra_saturation_margin_db >= 0.f &&
config.extra_saturation_margin_db <= 100.f;
}
std::string GainController2::ToString(
const AudioProcessing::Config::GainController2& config) {
rtc::StringBuilder ss;
ss << "{enabled: " << (config.enabled ? "true" : "false") << ", "
<< "fixed_gain_dB: " << config.fixed_gain_db << "}";
return ss.Release();
}
} // namespace webrtc