chromium/third_party/webrtc/modules/audio_coding/neteq/tools/packet.h

/*
 *  Copyright (c) 2014 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_TOOLS_PACKET_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_

#include <list>

#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "rtc_base/copy_on_write_buffer.h"

namespace webrtc {
namespace test {

// Class for handling RTP packets in test applications.
class Packet { … };

}  // namespace test
}  // namespace webrtc
#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_