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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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57 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; | 57 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; |
58 const int RtpExtension::kAbsSendTimeDefaultId = 3; | 58 const int RtpExtension::kAbsSendTimeDefaultId = 3; |
59 | 59 |
60 const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation"; | 60 const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation"; |
61 const int RtpExtension::kVideoRotationDefaultId = 4; | 61 const int RtpExtension::kVideoRotationDefaultId = 4; |
62 | 62 |
63 const char* RtpExtension::kTransportSequenceNumberUri = | 63 const char* RtpExtension::kTransportSequenceNumberUri = |
64 "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; | 64 "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; |
65 const int RtpExtension::kTransportSequenceNumberDefaultId = 5; | 65 const int RtpExtension::kTransportSequenceNumberDefaultId = 5; |
66 | 66 |
67 const char* RtpExtension::kVideoContentTypeUri = | |
68 "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; | |
69 const int RtpExtension::kVideoContentTypeDefaultId = 6; | |
70 | |
71 // This extension allows applications to adaptively limit the playout delay | 67 // This extension allows applications to adaptively limit the playout delay |
72 // on frames as per the current needs. For example, a gaming application | 68 // on frames as per the current needs. For example, a gaming application |
73 // 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 |
74 // application. | 70 // application. |
75 const char* RtpExtension::kPlayoutDelayUri = | 71 const char* RtpExtension::kPlayoutDelayUri = |
76 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; | 72 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; |
77 const int RtpExtension::kPlayoutDelayDefaultId = 6; | 73 const int RtpExtension::kPlayoutDelayDefaultId = 6; |
78 | 74 |
79 const int RtpExtension::kMinId = 1; | 75 const int RtpExtension::kMinId = 1; |
80 const int RtpExtension::kMaxId = 14; | 76 const int RtpExtension::kMaxId = 14; |
81 | 77 |
82 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { | 78 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { |
83 return uri == webrtc::RtpExtension::kAudioLevelUri || | 79 return uri == webrtc::RtpExtension::kAudioLevelUri || |
84 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; | 80 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; |
85 } | 81 } |
86 | 82 |
87 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { | 83 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
88 return uri == webrtc::RtpExtension::kTimestampOffsetUri || | 84 return uri == webrtc::RtpExtension::kTimestampOffsetUri || |
89 uri == webrtc::RtpExtension::kAbsSendTimeUri || | 85 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
90 uri == webrtc::RtpExtension::kVideoRotationUri || | 86 uri == webrtc::RtpExtension::kVideoRotationUri || |
91 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || | 87 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
92 uri == webrtc::RtpExtension::kPlayoutDelayUri || | 88 uri == webrtc::RtpExtension::kPlayoutDelayUri; |
93 uri == webrtc::RtpExtension::kVideoContentTypeUri; | |
94 } | 89 } |
95 | 90 |
96 VideoStream::VideoStream() | 91 VideoStream::VideoStream() |
97 : width(0), | 92 : width(0), |
98 height(0), | 93 height(0), |
99 max_framerate(-1), | 94 max_framerate(-1), |
100 min_bitrate_bps(-1), | 95 min_bitrate_bps(-1), |
101 target_bitrate_bps(-1), | 96 target_bitrate_bps(-1), |
102 max_bitrate_bps(-1), | 97 max_bitrate_bps(-1), |
103 max_qp(-1) {} | 98 max_qp(-1) {} |
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207 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( | 202 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( |
208 const VideoCodecVP9& specifics) | 203 const VideoCodecVP9& specifics) |
209 : specifics_(specifics) {} | 204 : specifics_(specifics) {} |
210 | 205 |
211 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( | 206 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( |
212 VideoCodecVP9* vp9_settings) const { | 207 VideoCodecVP9* vp9_settings) const { |
213 *vp9_settings = specifics_; | 208 *vp9_settings = specifics_; |
214 } | 209 } |
215 | 210 |
216 } // namespace webrtc | 211 } // namespace webrtc |
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