chromium/third_party/webrtc/call/rtp_stream_receiver_controller.h

/*
 *  Copyright (c) 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 CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
#define CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_

#include <memory>

#include "api/sequence_checker.h"
#include "call/rtp_demuxer.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/rtp_rtcp/include/recovered_packet_receiver.h"

namespace webrtc {

class RtpPacketReceived;

// This class represents the RTP receive parsing and demuxing, for a
// single RTP session.
// TODO(bugs.webrtc.org/7135): Add RTCP processing, we should aim to terminate
// RTCP and not leave any RTCP processing to individual receive streams.
class RtpStreamReceiverController : public RtpStreamReceiverControllerInterface,
                                    public RecoveredPacketReceiver { … };

}  // namespace webrtc

#endif  // CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_