chromium/third_party/webrtc/pc/rtp_transport.h

/*
 *  Copyright 2017 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 PC_RTP_TRANSPORT_H_
#define PC_RTP_TRANSPORT_H_

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

#include <string>

#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/units/timestamp.h"
#include "call/rtp_demuxer.h"
#include "call/video_receive_stream.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/received_packet.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"

namespace rtc {

class CopyOnWriteBuffer;
struct PacketOptions;
class PacketTransportInternal;

}  // namespace rtc

namespace webrtc {

class RtpTransport : public RtpTransportInternal { … };

}  // namespace webrtc

#endif  // PC_RTP_TRANSPORT_H_