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

/*
 *  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.
 */

#include "modules/audio_processing/agc2/saturation_protector_buffer.h"

#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_compare.h"

namespace webrtc {

SaturationProtectorBuffer::SaturationProtectorBuffer() = default;

SaturationProtectorBuffer::~SaturationProtectorBuffer() = default;

bool SaturationProtectorBuffer::operator==(
    const SaturationProtectorBuffer& b) const { … }

int SaturationProtectorBuffer::Capacity() const { … }

int SaturationProtectorBuffer::Size() const { … }

void SaturationProtectorBuffer::Reset() { … }

void SaturationProtectorBuffer::PushBack(float v) { … }

absl::optional<float> SaturationProtectorBuffer::Front() const { … }

int SaturationProtectorBuffer::FrontIndex() const { … }

}  // namespace webrtc