/* * 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 #include "api/sequence_checker.h" #include "call/rtp_demuxer.h" #include "call/rtp_stream_receiver_controller_interface.h" namespace webrtc { class RtpPacketReceived; // This class represents the RTP receive parsing and demuxing, for a // single RTP session. // TODO(nisse): Add RTCP processing, we should aim to terminate RTCP // and not leave any RTCP processing to individual receive streams. // TODO(nisse): Extract per-packet processing, including parsing and // demuxing, into a separate class. class RtpStreamReceiverController : public RtpStreamReceiverControllerInterface { public: RtpStreamReceiverController(); ~RtpStreamReceiverController() override; // Implements RtpStreamReceiverControllerInterface. std::unique_ptr CreateReceiver( uint32_t ssrc, RtpPacketSinkInterface* sink) override; // Thread-safe wrappers for the corresponding RtpDemuxer methods. bool 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 RtpStreamReceiverInterface { public: Receiver(RtpStreamReceiverController* controller, uint32_t ssrc, RtpPacketSinkInterface* sink); ~Receiver() override; private: RtpStreamReceiverController* const controller_; RtpPacketSinkInterface* const sink_; }; // TODO(bugs.webrtc.org/11993): We expect construction and all methods to be // called on the same thread/tq. Currently this is the worker thread // (including OnRtpPacket) but a more natural fit would be the network thread. // Using a sequence checker to ensure that usage is correct but at the same // time not require a specific thread/tq, an instance of this class + the // associated functionality should be easily moved from one execution context // to another (i.e. when network packets don't hop to the worker thread inside // of Call). SequenceChecker demuxer_sequence_; // At this level the demuxer is only configured to demux by SSRC, so don't // worry about MIDs (MIDs are handled by upper layers). RtpDemuxer demuxer_ RTC_GUARDED_BY(&demuxer_sequence_){false /*use_mid*/}; }; } // namespace webrtc #endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_