chromium/third_party/webrtc/modules/audio_processing/agc2/clipping_predictor_level_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/clipping_predictor_level_buffer.h"

#include <algorithm>
#include <cmath>

#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

namespace webrtc {

bool ClippingPredictorLevelBuffer::Level::operator==(const Level& level) const { … }

ClippingPredictorLevelBuffer::ClippingPredictorLevelBuffer(int capacity)
    : … { … }

void ClippingPredictorLevelBuffer::Reset() { … }

void ClippingPredictorLevelBuffer::Push(Level level) { … }

// TODO(bugs.webrtc.org/12774): Optimize partial computation for long buffers.
absl::optional<ClippingPredictorLevelBuffer::Level>
ClippingPredictorLevelBuffer::ComputePartialMetrics(int delay,
                                                    int num_items) const { … }

}  // namespace webrtc