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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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| 69 // has very different needs on end-to-end delay compared to a video-conference | 69 // has very different needs on end-to-end delay compared to a video-conference |
| 70 // application. | 70 // application. |
| 71 const char* RtpExtension::kPlayoutDelayUri = | 71 const char* RtpExtension::kPlayoutDelayUri = |
| 72 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; | 72 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; |
| 73 const int RtpExtension::kPlayoutDelayDefaultId = 6; | 73 const int RtpExtension::kPlayoutDelayDefaultId = 6; |
| 74 | 74 |
| 75 const char* RtpExtension::kVideoContentTypeUri = | 75 const char* RtpExtension::kVideoContentTypeUri = |
| 76 "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; | 76 "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; |
| 77 const int RtpExtension::kVideoContentTypeDefaultId = 7; | 77 const int RtpExtension::kVideoContentTypeDefaultId = 7; |
| 78 | 78 |
| 79 // This extensions provides meta-information about the RTP streams outside the |
| 80 // encrypted media payload, an RTP switch can do codec-agnostic |
| 81 // selective forwarding without decrypting the payload |
| 82 const char* RtpExtension::kFrameMarkingUri = |
| 83 "urn:ietf:params:rtp-hdrext:framemarking"; |
| 84 const int RtpExtension::kFrameMarkingDefaultId = 8; |
| 85 |
| 79 const int RtpExtension::kMinId = 1; | 86 const int RtpExtension::kMinId = 1; |
| 80 const int RtpExtension::kMaxId = 14; | 87 const int RtpExtension::kMaxId = 14; |
| 81 | 88 |
| 82 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { | 89 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { |
| 83 return uri == webrtc::RtpExtension::kAudioLevelUri || | 90 return uri == webrtc::RtpExtension::kAudioLevelUri || |
| 84 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; | 91 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; |
| 85 } | 92 } |
| 86 | 93 |
| 87 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { | 94 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
| 88 return uri == webrtc::RtpExtension::kTimestampOffsetUri || | 95 return uri == webrtc::RtpExtension::kTimestampOffsetUri || |
| 89 uri == webrtc::RtpExtension::kAbsSendTimeUri || | 96 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
| 90 uri == webrtc::RtpExtension::kVideoRotationUri || | 97 uri == webrtc::RtpExtension::kVideoRotationUri || |
| 91 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || | 98 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
| 92 uri == webrtc::RtpExtension::kPlayoutDelayUri || | 99 uri == webrtc::RtpExtension::kPlayoutDelayUri || |
| 93 uri == webrtc::RtpExtension::kVideoContentTypeUri; | 100 uri == webrtc::RtpExtension::kVideoContentTypeUri || |
| 101 uri == webrtc::RtpExtension::kFrameMarkingUri; |
| 94 } | 102 } |
| 95 | 103 |
| 96 VideoStream::VideoStream() | 104 VideoStream::VideoStream() |
| 97 : width(0), | 105 : width(0), |
| 98 height(0), | 106 height(0), |
| 99 max_framerate(-1), | 107 max_framerate(-1), |
| 100 min_bitrate_bps(-1), | 108 min_bitrate_bps(-1), |
| 101 target_bitrate_bps(-1), | 109 target_bitrate_bps(-1), |
| 102 max_bitrate_bps(-1), | 110 max_bitrate_bps(-1), |
| 103 max_qp(-1) {} | 111 max_qp(-1) {} |
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| 207 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( | 215 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( |
| 208 const VideoCodecVP9& specifics) | 216 const VideoCodecVP9& specifics) |
| 209 : specifics_(specifics) {} | 217 : specifics_(specifics) {} |
| 210 | 218 |
| 211 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( | 219 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( |
| 212 VideoCodecVP9* vp9_settings) const { | 220 VideoCodecVP9* vp9_settings) const { |
| 213 *vp9_settings = specifics_; | 221 *vp9_settings = specifics_; |
| 214 } | 222 } |
| 215 | 223 |
| 216 } // namespace webrtc | 224 } // namespace webrtc |
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