chromium/third_party/webrtc/pc/rtp_transport_internal.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_INTERNAL_H_
#define PC_RTP_TRANSPORT_INTERNAL_H_

#include <string>
#include <utility>

#include "call/rtp_demuxer.h"
#include "p2p/base/ice_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/callback_list.h"
#include "rtc_base/network_route.h"
#include "rtc_base/ssl_stream_adapter.h"

namespace rtc {
class CopyOnWriteBuffer;
struct PacketOptions;
}  // namespace rtc

namespace webrtc {

// This class is an internal interface; it is not accessible to API consumers
// but is accessible to internal classes in order to send and receive RTP and
// RTCP packets belonging to a single RTP session. Additional convenience and
// configuration methods are also provided.
class RtpTransportInternal : public sigslot::has_slots<> { … };

}  // namespace webrtc

#endif  // PC_RTP_TRANSPORT_INTERNAL_H_