chromium/third_party/webrtc/modules/audio_processing/agc2/saturation_protector.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_SATURATION_PROTECTOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_

#include <memory>

namespace webrtc {
class ApmDataDumper;

// Saturation protector. Analyzes peak levels and recommends a headroom to
// reduce the chances of clipping.
class SaturationProtector { … };

// Creates a saturation protector that starts at `initial_headroom_db`.
std::unique_ptr<SaturationProtector> CreateSaturationProtector(
    float initial_headroom_db,
    int adjacent_speech_frames_threshold,
    ApmDataDumper* apm_data_dumper);

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_