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Issue 2675173003: Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. (Closed)
Patch Set: Add memcheck suppression for end-to-end tests. Created 3 years, 9 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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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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