2017-06-21 01:05:22 -07:00
|
|
|
/*
|
|
|
|
|
* 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_STREAM_RECEIVER_CONTROLLER_H_
|
|
|
|
|
#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
|
|
|
|
|
|
|
|
|
|
#include <memory>
|
|
|
|
|
|
|
|
|
|
#include "webrtc/call/rtp_demuxer.h"
|
|
|
|
|
#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
|
2017-07-06 19:44:34 +02:00
|
|
|
#include "webrtc/rtc_base/criticalsection.h"
|
2017-06-21 01:05:22 -07:00
|
|
|
|
|
|
|
|
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.
|
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
Reason for revert:
Relanding
Original issue's description:
> Revert of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #15 id:280001 of https://codereview.webrtc.org/2968693002/ )
>
> Reason for revert:
> Some internal tests keep failing after this change. Try to fix it by reverting it. Will reland it if this isn't the root cause.
>
> Original issue's description:
> > SSRC and RSID may only refer to one sink each in RtpDemuxer
> >
> > RTP demuxing should only match RTP packets with one sink.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2968693002
> > Cr-Commit-Position: refs/heads/master@{#19233}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/7b7e06fd23ac67d81f378b773bb631abb1d82116
>
> TBR=nisse@webrtc.org,danilchap@webrtc.org,perkj@webrtc.org,stefan@webrtc.org,holmer@google.com,deadbeef@webrtc.org,pthatcher@webrtc.org,steveanton@webrtc.org,eladalon@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2993633002
> Cr-Commit-Position: refs/heads/master@{#19239}
> Committed: https://chromium.googlesource.com/external/webrtc/+/59b603fbed5b069090f9084c8eeb82eff7bca30c
TBR=nisse@webrtc.org,danilchap@webrtc.org,perkj@webrtc.org,stefan@webrtc.org,holmer@google.com,deadbeef@webrtc.org,pthatcher@webrtc.org,steveanton@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2993053002
Cr-Commit-Position: refs/heads/master@{#19248}
2017-08-04 06:34:54 -07:00
|
|
|
bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override;
|
2017-06-21 01:05:22 -07:00
|
|
|
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(nisse): Move to a TaskQueue for synchronization. When used
|
|
|
|
|
// by Call, we expect construction and all methods but OnRtpPacket
|
|
|
|
|
// to be called on the same thread, and OnRtpPacket to be called
|
|
|
|
|
// by a single, but possibly distinct, thread. But applications not
|
|
|
|
|
// using Call may have use threads differently.
|
|
|
|
|
rtc::CriticalSection lock_;
|
2017-09-09 04:17:22 -07:00
|
|
|
RtpDemuxer demuxer_ RTC_GUARDED_BY(&lock_);
|
2017-06-21 01:05:22 -07:00
|
|
|
};
|
|
|
|
|
|
|
|
|
|
} // namespace webrtc
|
|
|
|
|
|
|
|
|
|
#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
|