102 lines
3.7 KiB
C++
102 lines
3.7 KiB
C++
/*
|
|
* 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/atomic_ops.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),
|
|
limiter_(static_cast<size_t>(48000), data_dumper_.get(), "Agc2") {
|
|
if (config_.adaptive_digital.enabled) {
|
|
adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get()));
|
|
}
|
|
}
|
|
|
|
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(), audio->num_channels(),
|
|
audio->num_frames());
|
|
// Apply fixed gain first, then the adaptive one.
|
|
gain_applier_.ApplyGain(float_frame);
|
|
if (adaptive_agc_) {
|
|
adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
|
|
}
|
|
limiter_.Process(float_frame);
|
|
}
|
|
|
|
void GainController2::NotifyAnalogLevel(int level) {
|
|
if (analog_level_ != level && adaptive_agc_) {
|
|
adaptive_agc_->Reset();
|
|
}
|
|
analog_level_ = level;
|
|
}
|
|
|
|
void GainController2::ApplyConfig(
|
|
const AudioProcessing::Config::GainController2& config) {
|
|
RTC_DCHECK(Validate(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));
|
|
if (config_.adaptive_digital.enabled) {
|
|
adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
|
|
} else {
|
|
adaptive_agc_.reset();
|
|
}
|
|
}
|
|
|
|
bool GainController2::Validate(
|
|
const AudioProcessing::Config::GainController2& config) {
|
|
const auto& fixed = config.fixed_digital;
|
|
const auto& adaptive = config.adaptive_digital;
|
|
return fixed.gain_db >= 0.f && fixed.gain_db < 50.f &&
|
|
adaptive.vad_probability_attack > 0.f &&
|
|
adaptive.vad_probability_attack <= 1.f &&
|
|
adaptive.level_estimator_adjacent_speech_frames_threshold >= 1 &&
|
|
adaptive.initial_saturation_margin_db >= 0.f &&
|
|
adaptive.initial_saturation_margin_db <= 100.f &&
|
|
adaptive.extra_saturation_margin_db >= 0.f &&
|
|
adaptive.extra_saturation_margin_db <= 100.f &&
|
|
adaptive.gain_applier_adjacent_speech_frames_threshold >= 1 &&
|
|
adaptive.max_gain_change_db_per_second > 0.f &&
|
|
adaptive.max_output_noise_level_dbfs <= 0.f;
|
|
}
|
|
|
|
} // namespace webrtc
|