Nagram/TMessagesProj/jni/voip/webrtc/call/rtp_stream_receiver_controller.h
2021-06-25 03:43:10 +03:00

78 lines
2.9 KiB
C++

/*
* 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"
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<RtpStreamReceiverInterface> 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_