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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 #include "webrtc/config.h" | 10 #include "webrtc/config.h" |
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80 "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; | 80 "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; |
81 const int RtpExtension::kVideoContentTypeDefaultId = 7; | 81 const int RtpExtension::kVideoContentTypeDefaultId = 7; |
82 | 82 |
83 const char RtpExtension::kVideoTimingUri[] = | 83 const char RtpExtension::kVideoTimingUri[] = |
84 "http://www.webrtc.org/experiments/rtp-hdrext/video-timing"; | 84 "http://www.webrtc.org/experiments/rtp-hdrext/video-timing"; |
85 const int RtpExtension::kVideoTimingDefaultId = 8; | 85 const int RtpExtension::kVideoTimingDefaultId = 8; |
86 | 86 |
87 const char RtpExtension::kEncryptHeaderExtensionsUri[] = | 87 const char RtpExtension::kEncryptHeaderExtensionsUri[] = |
88 "urn:ietf:params:rtp-hdrext:encrypt"; | 88 "urn:ietf:params:rtp-hdrext:encrypt"; |
89 | 89 |
| 90 // This extensions provides meta-information about the RTP streams outside the |
| 91 // encrypted media payload, an RTP switch can do codec-agnostic |
| 92 // selective forwarding without decrypting the payload |
| 93 const char* RtpExtension::kFrameMarkingUri = |
| 94 "urn:ietf:params:rtp-hdrext:framemarking"; |
| 95 const int RtpExtension::kFrameMarkingDefaultId = 9; |
| 96 |
90 const int RtpExtension::kMinId = 1; | 97 const int RtpExtension::kMinId = 1; |
91 const int RtpExtension::kMaxId = 14; | 98 const int RtpExtension::kMaxId = 14; |
92 | 99 |
93 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { | 100 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { |
94 return uri == webrtc::RtpExtension::kAudioLevelUri || | 101 return uri == webrtc::RtpExtension::kAudioLevelUri || |
95 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; | 102 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; |
96 } | 103 } |
97 | 104 |
98 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { | 105 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
99 return uri == webrtc::RtpExtension::kTimestampOffsetUri || | 106 return uri == webrtc::RtpExtension::kTimestampOffsetUri || |
100 uri == webrtc::RtpExtension::kAbsSendTimeUri || | 107 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
101 uri == webrtc::RtpExtension::kVideoRotationUri || | 108 uri == webrtc::RtpExtension::kVideoRotationUri || |
102 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || | 109 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
103 uri == webrtc::RtpExtension::kPlayoutDelayUri || | 110 uri == webrtc::RtpExtension::kPlayoutDelayUri || |
104 uri == webrtc::RtpExtension::kVideoContentTypeUri || | 111 uri == webrtc::RtpExtension::kVideoContentTypeUri || |
105 uri == webrtc::RtpExtension::kVideoTimingUri; | 112 uri == webrtc::RtpExtension::kVideoTimingUri || |
| 113 uri == webrtc::RtpExtension::kFrameMarkingUri; |
106 } | 114 } |
107 | 115 |
108 bool RtpExtension::IsEncryptionSupported(const std::string& uri) { | 116 bool RtpExtension::IsEncryptionSupported(const std::string& uri) { |
109 return uri == webrtc::RtpExtension::kAudioLevelUri || | 117 return uri == webrtc::RtpExtension::kAudioLevelUri || |
110 uri == webrtc::RtpExtension::kTimestampOffsetUri || | 118 uri == webrtc::RtpExtension::kTimestampOffsetUri || |
111 #if !defined(ENABLE_EXTERNAL_AUTH) | 119 #if !defined(ENABLE_EXTERNAL_AUTH) |
112 // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri" | 120 // TODO(jbauch): Figure out a way to always allow "kAbsSendTimeUri" |
113 // here and filter out later if external auth is really used in | 121 // here and filter out later if external auth is really used in |
114 // srtpfilter. External auth is used by Chromium and replaces the | 122 // srtpfilter. External auth is used by Chromium and replaces the |
115 // extension header value of "kAbsSendTimeUri", so it must not be | 123 // extension header value of "kAbsSendTimeUri", so it must not be |
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274 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( | 282 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( |
275 const VideoCodecVP9& specifics) | 283 const VideoCodecVP9& specifics) |
276 : specifics_(specifics) {} | 284 : specifics_(specifics) {} |
277 | 285 |
278 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( | 286 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( |
279 VideoCodecVP9* vp9_settings) const { | 287 VideoCodecVP9* vp9_settings) const { |
280 *vp9_settings = specifics_; | 288 *vp9_settings = specifics_; |
281 } | 289 } |
282 | 290 |
283 } // namespace webrtc | 291 } // namespace webrtc |
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