5daecca41b
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:7b7e06fd23> > 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:59b603fbedTBR=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}
48 lines
1.8 KiB
C++
48 lines
1.8 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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#include <memory>
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#include "webrtc/call/rtp_packet_sink_interface.h"
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namespace webrtc {
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// An RtpStreamReceiver is responsible for the rtp-specific but
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// media-independent state needed for receiving an RTP stream.
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// TODO(nisse): Currently, only owns the association between ssrc and
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// the stream's RtpPacketSinkInterface. Ownership of corresponding
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// objects from modules/rtp_rtcp/ should move to this class (or
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// rather, the corresponding implementation class). We should add
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// methods for getting rtp receive stats, and for sending RTCP
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// messages related to the receive stream.
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class RtpStreamReceiverInterface {
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public:
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virtual ~RtpStreamReceiverInterface() {}
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};
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// This class acts as a factory for RtpStreamReceiver objects.
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class RtpStreamReceiverControllerInterface {
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public:
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virtual ~RtpStreamReceiverControllerInterface() {}
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virtual std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver(
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uint32_t ssrc,
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RtpPacketSinkInterface* sink) = 0;
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// For registering additional sinks, needed for FlexFEC.
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virtual bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0;
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virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
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