| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| |
| #include <stddef.h> |
| #include <cstdint> |
| #include <vector> |
| |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| |
| namespace webrtc { |
| |
| RtpPacketReceived::RtpPacketReceived() = default; |
| RtpPacketReceived::RtpPacketReceived(const ExtensionManager* extensions) |
| : RtpPacket(extensions) {} |
| RtpPacketReceived::RtpPacketReceived(const RtpPacketReceived& packet) = default; |
| RtpPacketReceived::RtpPacketReceived(RtpPacketReceived&& packet) = default; |
| |
| RtpPacketReceived& RtpPacketReceived::operator=( |
| const RtpPacketReceived& packet) = default; |
| RtpPacketReceived& RtpPacketReceived::operator=(RtpPacketReceived&& packet) = |
| default; |
| |
| RtpPacketReceived::~RtpPacketReceived() {} |
| |
| void RtpPacketReceived::GetHeader(RTPHeader* header) const { |
| header->markerBit = Marker(); |
| header->payloadType = PayloadType(); |
| header->sequenceNumber = SequenceNumber(); |
| header->timestamp = Timestamp(); |
| header->ssrc = Ssrc(); |
| std::vector<uint32_t> csrcs = Csrcs(); |
| header->numCSRCs = rtc::dchecked_cast<uint8_t>(csrcs.size()); |
| for (size_t i = 0; i < csrcs.size(); ++i) { |
| header->arrOfCSRCs[i] = csrcs[i]; |
| } |
| header->paddingLength = padding_size(); |
| header->headerLength = headers_size(); |
| header->payload_type_frequency = payload_type_frequency(); |
| header->extension.hasTransmissionTimeOffset = |
| GetExtension<TransmissionOffset>( |
| &header->extension.transmissionTimeOffset); |
| header->extension.hasAbsoluteSendTime = |
| GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime); |
| header->extension.absolute_capture_time = |
| GetExtension<AbsoluteCaptureTimeExtension>(); |
| header->extension.hasTransportSequenceNumber = |
| GetExtension<TransportSequenceNumberV2>( |
| &header->extension.transportSequenceNumber, |
| &header->extension.feedback_request) || |
| GetExtension<TransportSequenceNumber>( |
| &header->extension.transportSequenceNumber); |
| header->extension.hasAudioLevel = GetExtension<AudioLevel>( |
| &header->extension.voiceActivity, &header->extension.audioLevel); |
| header->extension.hasVideoRotation = |
| GetExtension<VideoOrientation>(&header->extension.videoRotation); |
| header->extension.hasVideoContentType = |
| GetExtension<VideoContentTypeExtension>( |
| &header->extension.videoContentType); |
| header->extension.has_video_timing = |
| GetExtension<VideoTimingExtension>(&header->extension.video_timing); |
| header->extension.has_frame_marking = |
| GetExtension<FrameMarkingExtension>(&header->extension.frame_marking); |
| GetExtension<RtpStreamId>(&header->extension.stream_id); |
| GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id); |
| GetExtension<RtpMid>(&header->extension.mid); |
| GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay); |
| header->extension.color_space = GetExtension<ColorSpaceExtension>(); |
| } |
| |
| } // namespace webrtc |