chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h

/*
 *  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.
 */

#ifndef MODULES_AUDIO_CODING_NETEQ_MERGE_H_
#define MODULES_AUDIO_CODING_NETEQ_MERGE_H_

#include "modules/audio_coding/neteq/audio_multi_vector.h"

namespace webrtc {

// Forward declarations.
class Expand;
class SyncBuffer;

// This class handles the transition from expansion to normal operation.
// When a packet is not available for decoding when needed, the expand operation
// is called to generate extrapolation data. If the missing packet arrives,
// i.e., it was just delayed, it can be decoded and appended directly to the
// end of the expanded data (thanks to how the Expand class operates). However,
// if a later packet arrives instead, the loss is a fact, and the new data must
// be stitched together with the end of the expanded data. This stitching is
// what the Merge class does.
class Merge {};

}  // namespace webrtc
#endif  // MODULES_AUDIO_CODING_NETEQ_MERGE_H_