35713eaf56
Reason for revert: Intend to fix perf failures and reland. Original issue's description: > Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) > > Reason for revert: > A few perf tests broken, including > > RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx > RampUpTest.UpDownUpTransportSequenceNumberRtx > RampUpTest.UpDownUpTransportSequenceNumberPacketLoss > > > Original issue's description: > > Use RtxReceiveStream. > > > > This also has the beneficial side-effect that when a media stream > > which is protected by FlexFEC receives an RTX retransmission, the > > retransmitted media packet is passed into the FlexFEC machinery, > > which should improve its ability to recover packets via FEC. > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/3008773002 > > Cr-Commit-Position: refs/heads/master@{#19649} > > Committed:5c0f6c62ea> > TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@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/3010983002 > Cr-Commit-Position: refs/heads/master@{#19653} > Committed:3c39c0137aTBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/3006063002 Cr-Commit-Position: refs/heads/master@{#19715}
51 lines
1.8 KiB
C++
51 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_RTX_RECEIVE_STREAM_H_
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#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
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#include <map>
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#include "webrtc/call/rtp_packet_sink_interface.h"
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namespace webrtc {
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class ReceiveStatistics;
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// This class is responsible for RTX decapsulation. The resulting media packets
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// are passed on to a sink representing the associated media stream.
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class RtxReceiveStream : public RtpPacketSinkInterface {
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public:
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RtxReceiveStream(RtpPacketSinkInterface* media_sink,
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std::map<int, int> associated_payload_types,
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uint32_t media_ssrc,
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// TODO(nisse): Delete this argument, and
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// corresponding member variable, by moving the
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// responsibility for rtcp feedback to
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// RtpStreamReceiverController.
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ReceiveStatistics* rtp_receive_statistics = nullptr);
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~RtxReceiveStream() override;
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// RtpPacketSinkInterface.
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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private:
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RtpPacketSinkInterface* const media_sink_;
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// Map from rtx payload type -> media payload type.
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const std::map<int, int> associated_payload_types_;
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// TODO(nisse): Ultimately, the media receive stream shouldn't care about the
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// ssrc, and we should delete this.
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const uint32_t media_ssrc_;
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ReceiveStatistics* const rtp_receive_statistics_;
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};
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} // namespace webrtc
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#endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
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