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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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65 const int RtpExtension::kTransportSequenceNumberDefaultId = 5; | 65 const int RtpExtension::kTransportSequenceNumberDefaultId = 5; |
66 | 66 |
67 // This extension allows applications to adaptively limit the playout delay | 67 // This extension allows applications to adaptively limit the playout delay |
68 // 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 |
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 int RtpExtension::kMinId = 1; |
| 76 const int RtpExtension::kMaxId = 14; |
| 77 |
75 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { | 78 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { |
76 return uri == webrtc::RtpExtension::kAudioLevelUri || | 79 return uri == webrtc::RtpExtension::kAudioLevelUri || |
77 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; | 80 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; |
78 } | 81 } |
79 | 82 |
80 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { | 83 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
81 return uri == webrtc::RtpExtension::kTimestampOffsetUri || | 84 return uri == webrtc::RtpExtension::kTimestampOffsetUri || |
82 uri == webrtc::RtpExtension::kAbsSendTimeUri || | 85 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
83 uri == webrtc::RtpExtension::kVideoRotationUri || | 86 uri == webrtc::RtpExtension::kVideoRotationUri || |
84 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || | 87 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
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199 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( | 202 VideoEncoderConfig::Vp9EncoderSpecificSettings::Vp9EncoderSpecificSettings( |
200 const VideoCodecVP9& specifics) | 203 const VideoCodecVP9& specifics) |
201 : specifics_(specifics) {} | 204 : specifics_(specifics) {} |
202 | 205 |
203 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( | 206 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( |
204 VideoCodecVP9* vp9_settings) const { | 207 VideoCodecVP9* vp9_settings) const { |
205 *vp9_settings = specifics_; | 208 *vp9_settings = specifics_; |
206 } | 209 } |
207 | 210 |
208 } // namespace webrtc | 211 } // namespace webrtc |
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