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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc

Issue 1841453004: RtpPacket class introduced. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: added empty lines Created 4 years, 8 months ago
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1 /*
2 * Copyright (c) 2016 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
11 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
12
13 #include "webrtc/base/checks.h"
14 #include "webrtc/base/logging.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
16 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
17
18 namespace webrtc {
19 // Absolute send time in RTP streams.
20 //
21 // The absolute send time is signaled to the receiver in-band using the
22 // general mechanism for RTP header extensions [RFC5285]. The payload
23 // of this extension (the transmitted value) is a 24-bit unsigned integer
24 // containing the sender's current time in seconds as a fixed point number
25 // with 18 bits fractional part.
26 //
27 // The form of the absolute send time extension block:
28 //
29 // 0 1 2 3
30 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
31 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
32 // | ID | len=2 | absolute send time |
33 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
34 const char* AbsoluteSendTime::kName =
35 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
36 bool AbsoluteSendTime::IsSupportedFor(MediaType type) {
37 return true;
38 }
39
40 bool AbsoluteSendTime::Parse(const uint8_t* data, uint32_t* value) {
41 *value = ByteReader<uint32_t, 3>::ReadBigEndian(data);
42 return true;
43 }
44
45 bool AbsoluteSendTime::Write(uint8_t* data, int64_t time_ms) {
46 const uint32_t kAbsSendTimeFraction = 18;
47 uint32_t time_24_bits =
48 static_cast<uint32_t>(((time_ms << kAbsSendTimeFraction) + 500) / 1000) &
49 0x00FFFFFF;
50
51 ByteWriter<uint32_t, 3>::WriteBigEndian(data, time_24_bits);
52 return true;
53 }
54
55 // An RTP Header Extension for Client-to-Mixer Audio Level Indication
56 //
57 // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
58 //
59 // The form of the audio level extension block:
60 //
61 // 0 1
62 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
63 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
64 // | ID | len=0 |V| level |
65 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
66 //
67 const char* AudioLevel::kName = "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
68 bool AudioLevel::IsSupportedFor(MediaType type) {
69 switch (type) {
70 case MediaType::ANY:
71 case MediaType::AUDIO:
72 return true;
73 case MediaType::VIDEO:
74 case MediaType::DATA:
75 return false;
76 }
77 RTC_NOTREACHED();
78 return false;
79 }
80
81 bool AudioLevel::Parse(const uint8_t* data,
82 bool* voice_activity,
83 uint8_t* audio_level) {
84 *voice_activity = (data[0] & 0x80) != 0;
85 *audio_level = data[0] & 0x7F;
86 return true;
87 }
88
89 bool AudioLevel::Write(uint8_t* data,
90 bool voice_activity,
91 uint8_t audio_level) {
92 RTC_CHECK_LE(audio_level, 0x7f);
93 data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
94 return true;
95 }
96
97 // From RFC 5450: Transmission Time Offsets in RTP Streams.
98 //
99 // The transmission time is signaled to the receiver in-band using the
100 // general mechanism for RTP header extensions [RFC5285]. The payload
101 // of this extension (the transmitted value) is a 24-bit signed integer.
102 // When added to the RTP timestamp of the packet, it represents the
103 // "effective" RTP transmission time of the packet, on the RTP
104 // timescale.
105 //
106 // The form of the transmission offset extension block:
107 //
108 // 0 1 2 3
109 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
110 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
111 // | ID | len=2 | transmission offset |
112 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
113 const char* TransmissionOffset::kName = "urn:ietf:params:rtp-hdrext:toffset";
114 bool TransmissionOffset::IsSupportedFor(MediaType type) {
115 switch (type) {
116 case MediaType::ANY:
117 case MediaType::VIDEO:
118 return true;
119 case MediaType::AUDIO:
120 case MediaType::DATA:
121 return false;
122 }
123 RTC_NOTREACHED();
124 return false;
125 }
126
127 bool TransmissionOffset::Parse(const uint8_t* data, int32_t* value) {
128 *value = ByteReader<int32_t, 3>::ReadBigEndian(data);
129 return true;
130 }
131
132 bool TransmissionOffset::Write(uint8_t* data, int64_t value) {
133 RTC_CHECK_LE(value, 0x00ffffff);
134 ByteWriter<int32_t, 3>::WriteBigEndian(data, value);
135 return true;
136 }
137
138 // 0 1 2
139 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
140 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
141 // | ID | L=1 |transport wide sequence number |
142 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
143 const char* TransportSequenceNumber::kName =
144 "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
145 bool TransportSequenceNumber::IsSupportedFor(MediaType type) {
146 return true;
147 }
148
149 bool TransportSequenceNumber::Parse(const uint8_t* data, uint16_t* value) {
150 *value = ByteReader<uint16_t>::ReadBigEndian(data);
151 return true;
152 }
153
154 bool TransportSequenceNumber::Write(uint8_t* data, uint16_t value) {
155 ByteWriter<uint16_t>::WriteBigEndian(data, value);
156 return true;
157 }
158
159 // Coordination of Video Orientation in RTP streams.
160 //
161 // Coordination of Video Orientation consists in signaling of the current
162 // orientation of the image captured on the sender side to the receiver for
163 // appropriate rendering and displaying.
164 //
165 // 0 1
166 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
167 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
168 // | ID | len=0 |0 0 0 0 C F R R|
169 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
170 const char* VideoOrientation::kName = "urn:3gpp:video-orientation";
171 bool VideoOrientation::IsSupportedFor(MediaType type) {
172 switch (type) {
173 case MediaType::ANY:
174 case MediaType::VIDEO:
175 return true;
176 case MediaType::AUDIO:
177 case MediaType::DATA:
178 return false;
179 }
180 RTC_NOTREACHED();
181 return false;
182 }
183
184 bool VideoOrientation::Parse(const uint8_t* data, VideoRotation* rotation) {
185 *rotation = ConvertCVOByteToVideoRotation(data[0] & 0x03);
186 return true;
187 }
188
189 bool VideoOrientation::Write(uint8_t* data, VideoRotation rotation) {
190 data[0] = ConvertVideoRotationToCVOByte(rotation);
191 return true;
192 }
193
194 bool VideoOrientation::Parse(const uint8_t* data, uint8_t* value) {
195 *value = data[0];
196 return true;
197 }
198
199 bool VideoOrientation::Write(uint8_t* data, uint8_t value) {
200 data[0] = value;
201 return true;
202 }
203 } // namespace webrtc
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