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

/*
 *  Copyright (c) 2021 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_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_

#include <array>

#include "absl/types/optional.h"
#include "modules/audio_processing/agc2/agc2_common.h"

namespace webrtc {

// Ring buffer for the saturation protector which only supports (i) push back
// and (ii) read oldest item.
class SaturationProtectorBuffer { … };

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_