blob: 6c0917af4d016d1da008ea6ac773e6beed4b72d5 [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_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 {
explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
AdaptiveAgc(ApmDataDumper* apm_data_dumper, float extra_saturation_margin_db);
void Process(AudioFrameView<float> float_frame, float last_audio_level);
void Reset();
AdaptiveModeLevelEstimator speech_level_estimator_;
VadWithLevel vad_;
AdaptiveDigitalGainApplier gain_applier_;
ApmDataDumper* const apm_data_dumper_;
NoiseLevelEstimator noise_level_estimator_;
} // namespace webrtc