blob: 6b2cc76981a0f76136daf39a90234d1cb9e942bb (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
|
/*
* 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,
webrtc::Timestamp arrival_time /*= webrtc::Timestamp::MinusInfinity()*/)
: RtpPacket(extensions), arrival_time_(arrival_time) {}
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);
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
|
