/* * 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 #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 { public: SaturationProtectorBuffer(); ~SaturationProtectorBuffer(); bool operator==(const SaturationProtectorBuffer& b) const; inline bool operator!=(const SaturationProtectorBuffer& b) const { return !(*this == b); } // Maximum number of values that the buffer can contain. int Capacity() const; // Number of values in the buffer. int Size() const; void Reset(); // Pushes back `v`. If the buffer is full, the oldest value is replaced. void PushBack(float v); // Returns the oldest item in the buffer. Returns an empty value if the // buffer is empty. absl::optional Front() const; private: int FrontIndex() const; // `buffer_` has `size_` elements (up to the size of `buffer_`) and `next_` is // the position where the next new value is written in `buffer_`. std::array buffer_; int next_ = 0; int size_ = 0; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_