danilchap | ce25181 | 2017-09-11 12:24:41 -0700 | [diff] [blame] | 1 | /* |
| 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 Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 11 | #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
danilchap | ce25181 | 2017-09-11 12:24:41 -0700 | [diff] [blame] | 12 | |
| 13 | #include <vector> |
| 14 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 15 | #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| 16 | #include "rtc_base/safe_conversions.h" |
danilchap | ce25181 | 2017-09-11 12:24:41 -0700 | [diff] [blame] | 17 | |
| 18 | namespace webrtc { |
| 19 | |
Dino Radaković | 21360eb | 2017-10-24 15:40:40 +0200 | [diff] [blame] | 20 | RtpPacketReceived::RtpPacketReceived() = default; |
| 21 | RtpPacketReceived::RtpPacketReceived(const ExtensionManager* extensions) |
| 22 | : RtpPacket(extensions) {} |
| 23 | |
| 24 | RtpPacketReceived::~RtpPacketReceived() {} |
| 25 | |
danilchap | ce25181 | 2017-09-11 12:24:41 -0700 | [diff] [blame] | 26 | void 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(); |
danilchap | 772bd8b | 2017-09-13 03:24:28 -0700 | [diff] [blame] | 33 | header->numCSRCs = rtc::dchecked_cast<uint8_t>(csrcs.size()); |
danilchap | ce25181 | 2017-09-11 12:24:41 -0700 | [diff] [blame] | 34 | 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 |