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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
57 lines
2.3 KiB
C++
57 lines
2.3 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 <vector>
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "rtc_base/safe_conversions.h"
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namespace webrtc {
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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<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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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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}
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} // namespace webrtc
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