chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.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/sync_buffer.h"

#include <algorithm>  // Access to min.

#include "rtc_base/checks.h"

namespace webrtc {

size_t SyncBuffer::FutureLength() const { … }

void SyncBuffer::PushBack(const AudioMultiVector& append_this) { … }

void SyncBuffer::PushBackInterleaved(const rtc::BufferT<int16_t>& append_this) { … }

void SyncBuffer::PushFrontZeros(size_t length) { … }

void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) { … }

void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
                                size_t length,
                                size_t position) { … }

void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
                                size_t position) { … }

void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
                                         AudioFrame* output) { … }

void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) { … }

void SyncBuffer::Flush() { … }

void SyncBuffer::set_next_index(size_t value) { … }

void SyncBuffer::set_dtmf_index(size_t value) { … }

}  // namespace webrtc