blob: 83f10e20cfcadaf73ae88a1a8f9371c0a246c814 [file] [log] [blame]
danilchapce251812017-09-11 12:24:41 -07001/*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#include "modules/rtp_rtcp/source/rtp_packet_received.h"
danilchapce251812017-09-11 12:24:41 -070012
13#include <vector>
14
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
16#include "rtc_base/safe_conversions.h"
danilchapce251812017-09-11 12:24:41 -070017
18namespace webrtc {
19
Dino Radaković21360eb2017-10-24 15:40:40 +020020RtpPacketReceived::RtpPacketReceived() = default;
21RtpPacketReceived::RtpPacketReceived(const ExtensionManager* extensions)
22 : RtpPacket(extensions) {}
23
24RtpPacketReceived::~RtpPacketReceived() {}
25
danilchapce251812017-09-11 12:24:41 -070026void RtpPacketReceived::GetHeader(RTPHeader* header) const {
27 header->markerBit = Marker();
28 header->payloadType = PayloadType();
29 header->sequenceNumber = SequenceNumber();
30 header->timestamp = Timestamp();
31 header->ssrc = Ssrc();
32 std::vector<uint32_t> csrcs = Csrcs();
danilchap772bd8b2017-09-13 03:24:28 -070033 header->numCSRCs = rtc::dchecked_cast<uint8_t>(csrcs.size());
danilchapce251812017-09-11 12:24:41 -070034 for (size_t i = 0; i < csrcs.size(); ++i) {
35 header->arrOfCSRCs[i] = csrcs[i];
36 }
37 header->paddingLength = padding_size();
38 header->headerLength = headers_size();
39 header->payload_type_frequency = payload_type_frequency();
40 header->extension.hasTransmissionTimeOffset =
41 GetExtension<TransmissionOffset>(
42 &header->extension.transmissionTimeOffset);
43 header->extension.hasAbsoluteSendTime =
44 GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
45 header->extension.hasTransportSequenceNumber =
46 GetExtension<TransportSequenceNumber>(
47 &header->extension.transportSequenceNumber);
48 header->extension.hasAudioLevel = GetExtension<AudioLevel>(
49 &header->extension.voiceActivity, &header->extension.audioLevel);
50 header->extension.hasVideoRotation =
51 GetExtension<VideoOrientation>(&header->extension.videoRotation);
52 header->extension.hasVideoContentType =
53 GetExtension<VideoContentTypeExtension>(
54 &header->extension.videoContentType);
55 header->extension.has_video_timing =
56 GetExtension<VideoTimingExtension>(&header->extension.video_timing);
57 GetExtension<RtpStreamId>(&header->extension.stream_id);
58 GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
59 GetExtension<RtpMid>(&header->extension.mid);
60 GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
61}
62
63} // namespace webrtc