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54047bea1b
This reverts commit109b5fb5f5. Reason for revert: The failing libfuzzer was fixed in commitd6c6f16063Original change's description: > Revert "Extend TransportSequenceNumber RTP header extension" > > This reverts commit28c7362bc4. > > Reason for revert: It breaks Linux64 Release (libfuzzer): > https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket.appspot.com/8921003137877469920/+/steps/compile/0/stdout > > Original change's description: > > Extend TransportSequenceNumber RTP header extension > > > > Extend TransportSequenceNumber RTP header extension to support > > feedback on sender request. > > > > Bug: webrtc:10262 > > Change-Id: Ibc1cf18162d15cd102e951c9dc697d8ea536ebb6 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123233 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26766} > > TBR=danilchap@webrtc.org,aleloi@webrtc.org,kron@webrtc.org > > Change-Id: Ie8a73f5fdffd99919ceaa1ae8911a1645f2077e9 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10262 > Reviewed-on: https://webrtc-review.googlesource.com/c/123522 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26767} TBR=danilchap@webrtc.org,mbonadei@webrtc.org,aleloi@webrtc.org,kron@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10262 Change-Id: I0f854299a46c042cfbdf8b8cc8cd965a228142c8 Reviewed-on: https://webrtc-review.googlesource.com/c/123764 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26798}
78 lines
3.2 KiB
C++
78 lines
3.2 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 "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include <stddef.h>
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#include <cstdint>
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#include <vector>
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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RtpPacketReceived::RtpPacketReceived() = default;
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RtpPacketReceived::RtpPacketReceived(const ExtensionManager* extensions)
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: RtpPacket(extensions) {}
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RtpPacketReceived::RtpPacketReceived(const RtpPacketReceived& packet) = default;
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RtpPacketReceived::RtpPacketReceived(RtpPacketReceived&& packet) = default;
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RtpPacketReceived& RtpPacketReceived::operator=(
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const RtpPacketReceived& packet) = default;
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RtpPacketReceived& RtpPacketReceived::operator=(RtpPacketReceived&& packet) =
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default;
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RtpPacketReceived::~RtpPacketReceived() {}
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void RtpPacketReceived::GetHeader(RTPHeader* header) const {
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header->markerBit = Marker();
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header->payloadType = PayloadType();
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header->sequenceNumber = SequenceNumber();
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header->timestamp = Timestamp();
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header->ssrc = Ssrc();
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std::vector<uint32_t> csrcs = Csrcs();
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header->numCSRCs = rtc::dchecked_cast<uint8_t>(csrcs.size());
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for (size_t i = 0; i < csrcs.size(); ++i) {
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header->arrOfCSRCs[i] = csrcs[i];
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}
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header->paddingLength = padding_size();
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header->headerLength = headers_size();
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header->payload_type_frequency = payload_type_frequency();
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header->extension.hasTransmissionTimeOffset =
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GetExtension<TransmissionOffset>(
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&header->extension.transmissionTimeOffset);
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header->extension.hasAbsoluteSendTime =
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GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
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header->extension.hasTransportSequenceNumber =
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GetExtension<TransportSequenceNumberV2>(
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&header->extension.transportSequenceNumber,
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&header->extension.feedback_request) ||
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GetExtension<TransportSequenceNumber>(
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&header->extension.transportSequenceNumber);
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header->extension.hasAudioLevel = GetExtension<AudioLevel>(
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&header->extension.voiceActivity, &header->extension.audioLevel);
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header->extension.hasVideoRotation =
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GetExtension<VideoOrientation>(&header->extension.videoRotation);
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header->extension.hasVideoContentType =
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GetExtension<VideoContentTypeExtension>(
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&header->extension.videoContentType);
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header->extension.has_video_timing =
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GetExtension<VideoTimingExtension>(&header->extension.video_timing);
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header->extension.has_frame_marking =
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GetExtension<FrameMarkingExtension>(&header->extension.frame_marking);
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GetExtension<RtpStreamId>(&header->extension.stream_id);
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GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
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GetExtension<RtpMid>(&header->extension.mid);
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GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
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header->extension.color_space = GetExtension<ColorSpaceExtension>();
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}
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} // namespace webrtc
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