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/*
* 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/logging.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
int GainController2::instance_count_ = 0;
GainController2::GainController2()
: data_dumper_(rtc::AtomicOps::Increment(&instance_count_)),
gain_applier_(/*hard_clip_samples=*/false,
/*initial_gain_factor=*/0.0f),
limiter_(static_cast<size_t>(48000), &data_dumper_, "Agc2"),
calls_since_last_limiter_log_(0) {
if (config_.adaptive_digital.enabled) {
adaptive_agc_ =
std::make_unique<AdaptiveAgc>(&data_dumper_, config_.adaptive_digital);
}
}
GainController2::~GainController2() = default;
void GainController2::Initialize(int sample_rate_hz, int num_channels) {
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);
if (adaptive_agc_) {
adaptive_agc_->Initialize(sample_rate_hz, num_channels);
}
data_dumper_.InitiateNewSetOfRecordings();
data_dumper_.DumpRaw("sample_rate_hz", sample_rate_hz);
calls_since_last_limiter_log_ = 0;
}
void GainController2::Process(AudioBuffer* audio) {
data_dumper_.DumpRaw("agc2_notified_analog_level", analog_level_);
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);
// Log limiter stats every 30 seconds.
++calls_since_last_limiter_log_;
if (calls_since_last_limiter_log_ == 3000) {
calls_since_last_limiter_log_ = 0;
InterpolatedGainCurve::Stats stats = limiter_.GetGainCurveStats();
RTC_LOG(LS_INFO) << "AGC2 limiter stats"
<< " | identity: " << stats.look_ups_identity_region
<< " | knee: " << stats.look_ups_knee_region
<< " | limiter: " << stats.look_ups_limiter_region
<< " | saturation: " << stats.look_ups_saturation_region;
}
}
void GainController2::NotifyAnalogLevel(int level) {
if (analog_level_ != level && adaptive_agc_) {
adaptive_agc_->HandleInputGainChange();
}
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_ =
std::make_unique<AdaptiveAgc>(&data_dumper_, config_.adaptive_digital);
} 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
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