blob: 381e454868bcb22011cee6e2e6b5f342564e4eae [file] [log] [blame]
/*
* Copyright (c) 2018 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/agc2/adaptive_digital_gain_controller.h"
#include <algorithm>
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/vad_wrapper.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
// Peak and RMS audio levels in dBFS.
struct AudioLevels {
float peak_dbfs;
float rms_dbfs;
};
// Computes the audio levels for the first channel in `frame`.
AudioLevels ComputeAudioLevels(AudioFrameView<float> frame) {
float peak = 0.0f;
float rms = 0.0f;
for (const auto& x : frame.channel(0)) {
peak = std::max(std::fabs(x), peak);
rms += x * x;
}
return {FloatS16ToDbfs(peak),
FloatS16ToDbfs(std::sqrt(rms / frame.samples_per_channel()))};
}
} // namespace
AdaptiveDigitalGainController::AdaptiveDigitalGainController(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int sample_rate_hz,
int num_channels)
: speech_level_estimator_(apm_data_dumper, config),
gain_controller_(apm_data_dumper, config, sample_rate_hz, num_channels),
apm_data_dumper_(apm_data_dumper),
noise_level_estimator_(CreateNoiseFloorEstimator(apm_data_dumper)),
saturation_protector_(
CreateSaturationProtector(kSaturationProtectorInitialHeadroomDb,
config.adjacent_speech_frames_threshold,
apm_data_dumper)) {
RTC_DCHECK(apm_data_dumper);
RTC_DCHECK(noise_level_estimator_);
RTC_DCHECK(saturation_protector_);
}
AdaptiveDigitalGainController::~AdaptiveDigitalGainController() = default;
void AdaptiveDigitalGainController::Initialize(int sample_rate_hz,
int num_channels) {
gain_controller_.Initialize(sample_rate_hz, num_channels);
}
void AdaptiveDigitalGainController::Process(AudioFrameView<float> frame,
float speech_probability,
float limiter_envelope) {
AudioLevels levels = ComputeAudioLevels(frame);
apm_data_dumper_->DumpRaw("agc2_input_rms_dbfs", levels.rms_dbfs);
apm_data_dumper_->DumpRaw("agc2_input_peak_dbfs", levels.peak_dbfs);
AdaptiveDigitalGainApplier::FrameInfo info;
info.speech_probability = speech_probability;
speech_level_estimator_.Update(levels.rms_dbfs, levels.peak_dbfs,
info.speech_probability);
info.speech_level_dbfs = speech_level_estimator_.level_dbfs();
info.speech_level_reliable = speech_level_estimator_.IsConfident();
apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", info.speech_level_dbfs);
apm_data_dumper_->DumpRaw("agc2_speech_level_reliable",
info.speech_level_reliable);
info.noise_rms_dbfs = noise_level_estimator_->Analyze(frame);
apm_data_dumper_->DumpRaw("agc2_noise_rms_dbfs", info.noise_rms_dbfs);
saturation_protector_->Analyze(info.speech_probability, levels.peak_dbfs,
info.speech_level_dbfs);
info.headroom_db = saturation_protector_->HeadroomDb();
apm_data_dumper_->DumpRaw("agc2_headroom_db", info.headroom_db);
info.limiter_envelope_dbfs = FloatS16ToDbfs(limiter_envelope);
apm_data_dumper_->DumpRaw("agc2_limiter_envelope_dbfs",
info.limiter_envelope_dbfs);
gain_controller_.Process(info, frame);
}
void AdaptiveDigitalGainController::HandleInputGainChange() {
speech_level_estimator_.Reset();
saturation_protector_->Reset();
}
} // namespace webrtc