chromium/third_party/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc

/*
 *  Copyright (c) 2012 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_coding/neteq/buffer_level_filter.h"

#include <stdint.h>

#include <algorithm>

#include "rtc_base/numerics/safe_conversions.h"

namespace webrtc {

BufferLevelFilter::BufferLevelFilter() { … }

void BufferLevelFilter::Reset() { … }

void BufferLevelFilter::Update(size_t buffer_size_samples,
                               int time_stretched_samples) { … }

void BufferLevelFilter::SetFilteredBufferLevel(int buffer_size_samples) { … }

void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) { … }

}  // namespace webrtc