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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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79 | 79 |
80 // This extension allows applications to adaptively limit the playout delay | 80 // This extension allows applications to adaptively limit the playout delay |
81 // on frames as per the current needs. For example, a gaming application | 81 // on frames as per the current needs. For example, a gaming application |
82 // has very different needs on end-to-end delay compared to a video-conference | 82 // has very different needs on end-to-end delay compared to a video-conference |
83 // application. | 83 // application. |
84 const char* RtpExtension::kPlayoutDelayUri = | 84 const char* RtpExtension::kPlayoutDelayUri = |
85 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; | 85 "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"; |
86 const int RtpExtension::kPlayoutDelayDefaultId = 6; | 86 const int RtpExtension::kPlayoutDelayDefaultId = 6; |
87 | 87 |
88 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { | 88 bool RtpExtension::IsSupportedForAudio(const std::string& uri) { |
89 return uri == webrtc::RtpExtension::kAbsSendTimeUri || | 89 return uri == webrtc::RtpExtension::kAudioLevelUri || |
90 uri == webrtc::RtpExtension::kAudioLevelUri || | |
91 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; | 90 uri == webrtc::RtpExtension::kTransportSequenceNumberUri; |
92 } | 91 } |
93 | 92 |
94 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { | 93 bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
95 return uri == webrtc::RtpExtension::kTimestampOffsetUri || | 94 return uri == webrtc::RtpExtension::kTimestampOffsetUri || |
96 uri == webrtc::RtpExtension::kAbsSendTimeUri || | 95 uri == webrtc::RtpExtension::kAbsSendTimeUri || |
97 uri == webrtc::RtpExtension::kVideoRotationUri || | 96 uri == webrtc::RtpExtension::kVideoRotationUri || |
98 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || | 97 uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
99 uri == webrtc::RtpExtension::kPlayoutDelayUri; | 98 uri == webrtc::RtpExtension::kPlayoutDelayUri; |
100 } | 99 } |
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217 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( | 216 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9( |
218 VideoCodecVP9* vp9_settings) const { | 217 VideoCodecVP9* vp9_settings) const { |
219 *vp9_settings = specifics_; | 218 *vp9_settings = specifics_; |
220 } | 219 } |
221 | 220 |
222 DecoderSpecificSettings::DecoderSpecificSettings() = default; | 221 DecoderSpecificSettings::DecoderSpecificSettings() = default; |
223 | 222 |
224 DecoderSpecificSettings::~DecoderSpecificSettings() = default; | 223 DecoderSpecificSettings::~DecoderSpecificSettings() = default; |
225 | 224 |
226 } // namespace webrtc | 225 } // namespace webrtc |
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