* 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 {
class SaturationProtectorBuffer {
public:
SaturationProtectorBuffer();
~SaturationProtectorBuffer();
bool operator==(const SaturationProtectorBuffer& b) const;
inline bool operator!=(const SaturationProtectorBuffer& b) const {
return !(*this == b);
}
int Capacity() const;
int Size() const;
void Reset();
void PushBack(float v);
absl::optional<float> Front() const;
private:
int FrontIndex() const;
std::array<float, kSaturationProtectorBufferSize> buffer_;
int next_ = 0;
int size_ = 0;
};
}
#endif