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}
78 lines
2.6 KiB
C++
78 lines
2.6 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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#include <utility>
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#include "webrtc/call/rtx_receive_stream.h"
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#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "webrtc/rtc_base/logging.h"
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namespace webrtc {
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RtxReceiveStream::RtxReceiveStream(
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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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ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
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: media_sink_(media_sink),
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associated_payload_types_(std::move(associated_payload_types)),
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media_ssrc_(media_ssrc),
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rtp_receive_statistics_(rtp_receive_statistics) {
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if (associated_payload_types_.empty()) {
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LOG(LS_WARNING)
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<< "RtxReceiveStream created with empty payload type mapping.";
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}
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}
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RtxReceiveStream::~RtxReceiveStream() = default;
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void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
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if (rtp_receive_statistics_) {
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RTPHeader header;
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rtx_packet.GetHeader(&header);
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rtp_receive_statistics_->IncomingPacket(header, rtx_packet.size(),
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false /* retransmitted */);
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}
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rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
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if (payload.size() < kRtxHeaderSize) {
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return;
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}
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auto it = associated_payload_types_.find(rtx_packet.PayloadType());
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if (it == associated_payload_types_.end()) {
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LOG(LS_VERBOSE) << "Unknown payload type "
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<< static_cast<int>(rtx_packet.PayloadType())
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<< " on rtx ssrc " << rtx_packet.Ssrc();
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return;
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}
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RtpPacketReceived media_packet;
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media_packet.CopyHeaderFrom(rtx_packet);
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media_packet.SetSsrc(media_ssrc_);
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media_packet.SetSequenceNumber((payload[0] << 8) + payload[1]);
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media_packet.SetPayloadType(it->second);
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media_packet.set_recovered(true);
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// Skip the RTX header.
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rtc::ArrayView<const uint8_t> rtx_payload =
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payload.subview(kRtxHeaderSize);
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uint8_t* media_payload = media_packet.AllocatePayload(rtx_payload.size());
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RTC_DCHECK(media_payload != nullptr);
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memcpy(media_payload, rtx_payload.data(), rtx_payload.size());
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media_sink_->OnRtpPacket(media_packet);
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}
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} // namespace webrtc
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