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/*
* 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
#include "modules/audio_processing/agc2/noise_level_estimator.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class ApmDataDumper;
class AdaptiveAgc {
public:
explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
AdaptiveAgc(ApmDataDumper* apm_data_dumper, float extra_saturation_margin_db);
~AdaptiveAgc();
void Process(AudioFrameView<float> float_frame, float last_audio_level);
void Reset();
private:
AdaptiveModeLevelEstimator speech_level_estimator_;
VadWithLevel vad_;
AdaptiveDigitalGainApplier gain_applier_;
ApmDataDumper* const apm_data_dumper_;
NoiseLevelEstimator noise_level_estimator_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_