Chromium Code Reviews| Index: webrtc/call/rtp_transport_controller_receive.h |
| diff --git a/webrtc/call/rtp_transport_controller_receive.h b/webrtc/call/rtp_transport_controller_receive.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..e1f15ed8e5728d083bb2d5b96f8b79ba2883dad6 |
| --- /dev/null |
| +++ b/webrtc/call/rtp_transport_controller_receive.h |
| @@ -0,0 +1,67 @@ |
| +/* |
| + * 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 WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |
| +#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |
| + |
| +#include <memory> |
| + |
| +#include "webrtc/base/array_view.h" |
| +#include "webrtc/base/criticalsection.h" |
| + |
| +#include "webrtc/call/rtp_demuxer.h" |
| +#include "webrtc/call/rtp_transport_controller_receive_interface.h" |
| + |
| +namespace webrtc { |
| + |
| +class ReceiveSideCongestionController; |
| +class RtpPacketReceived; |
| + |
| +// This class represents the RTP receive parsing and demuxing, for a |
|
danilchap
2017/05/23 12:53:09
What benefits this class has over RtpDemuxer? only
|
| +// single RTP session. It isn't thread aware, leaving responsibility |
| +// of multithreading issues to the user of this class. |
| +// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP |
| +// and not leave any RTCP processing to individual receive streams. |
| +// TODO(nisse): Extract packet processing, including parsing and |
| +// demuxing, into a separate RtpSessionReceiver classes. |
| +class RtpTransportControllerReceive |
| + : public RtpTransportControllerReceiveInterface { |
| + public: |
| + // Implements RtpTransportControllerReceiveInterface. |
| + std::unique_ptr<RtpTransportReceiver> CreateReceiver( |
| + uint32_t ssrc, |
| + RtpPacketSinkInterface* sink) override; |
| + |
| + // Thread-safe wrappers for the corresponding RtpDemuxer methods. |
| + void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override; |
| + size_t RemoveSink(const RtpPacketSinkInterface* sink) override; |
| + |
| + // TODO(nisse): Not yet responsible for parsing. |
| + bool OnRtpPacket(const RtpPacketReceived& packet); |
| + |
| + private: |
| + class Receiver : public RtpTransportReceiver { |
| + public: |
| + Receiver(RtpTransportControllerReceive* transport, |
| + uint32_t ssrc, RtpPacketSinkInterface* sink); |
| + |
| + ~Receiver(); |
| + |
| + private: |
| + RtpTransportControllerReceive *transport_; |
| + RtpPacketSinkInterface* sink_; |
| + }; |
| + |
| + rtc::CriticalSection lock_; |
| + RtpDemuxer demuxer_ GUARDED_BY(&lock_); |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |