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Issue 2809653004: Revert of Add content type information to encoded images and corresponding rtp extension header (Closed)
Patch Set: Created 3 years, 8 months ago
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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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