chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

/*
 *  Copyright (c) 2016 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_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_

#include <stddef.h>
#include <stdint.h>

#include <utility>

#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_timing.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"

namespace webrtc {
// Class to hold rtp packet with metadata for sender side.
// The metadata is not send over the wire, but packet sender may use it to
// create rtp header extensions or other data that is sent over the wire.
class RtpPacketToSend : public RtpPacket { … };

}  // namespace webrtc
#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_