chromium/third_party/webrtc/modules/audio_processing/agc2/speech_level_estimator.h

/*
 *  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_SPEECH_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_

#include <stddef.h>

#include <type_traits>

#include "api/audio/audio_processing.h"
#include "modules/audio_processing/agc2/agc2_common.h"

namespace webrtc {
class ApmDataDumper;

// Active speech level estimator based on the analysis of the following
// framewise properties: RMS level (dBFS), peak level (dBFS), speech
// probability.
class SpeechLevelEstimator { … };

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_