tg2sip/webrtc_dsp/modules/audio_processing/agc2/adaptive_agc.cc

82 lines
3.0 KiB
C++

/*
* 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_agc.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
namespace webrtc {
AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
: speech_level_estimator_(apm_data_dumper),
gain_applier_(apm_data_dumper),
apm_data_dumper_(apm_data_dumper),
noise_level_estimator_(apm_data_dumper) {
RTC_DCHECK(apm_data_dumper);
}
AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2& config)
: speech_level_estimator_(
apm_data_dumper,
config.adaptive_digital.level_estimator,
config.adaptive_digital.use_saturation_protector,
config.adaptive_digital.extra_saturation_margin_db),
gain_applier_(apm_data_dumper),
apm_data_dumper_(apm_data_dumper),
noise_level_estimator_(apm_data_dumper) {
RTC_DCHECK(apm_data_dumper);
}
AdaptiveAgc::~AdaptiveAgc() = default;
void AdaptiveAgc::Process(AudioFrameView<float> float_frame,
float last_audio_level) {
auto signal_with_levels = SignalWithLevels(float_frame);
signal_with_levels.vad_result = vad_.AnalyzeFrame(float_frame);
apm_data_dumper_->DumpRaw("agc2_vad_probability",
signal_with_levels.vad_result.speech_probability);
apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs",
signal_with_levels.vad_result.speech_rms_dbfs);
apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs",
signal_with_levels.vad_result.speech_peak_dbfs);
speech_level_estimator_.UpdateEstimation(signal_with_levels.vad_result);
signal_with_levels.input_level_dbfs =
speech_level_estimator_.LatestLevelEstimate();
signal_with_levels.input_noise_level_dbfs =
noise_level_estimator_.Analyze(float_frame);
apm_data_dumper_->DumpRaw("agc2_noise_estimate_dbfs",
signal_with_levels.input_noise_level_dbfs);
signal_with_levels.limiter_audio_level_dbfs =
last_audio_level > 0 ? FloatS16ToDbfs(last_audio_level) : -90.f;
apm_data_dumper_->DumpRaw("agc2_last_limiter_audio_level",
signal_with_levels.limiter_audio_level_dbfs);
signal_with_levels.estimate_is_confident =
speech_level_estimator_.LevelEstimationIsConfident();
// The gain applier applies the gain.
gain_applier_.Process(signal_with_levels);
}
void AdaptiveAgc::Reset() {
speech_level_estimator_.Reset();
}
} // namespace webrtc