commit | a5d9c1a45c2bd748b7ec7b6456b7ddddfc46d2d6 | [log] [tgz] |
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author | Danil Chapovalov <danilchap@webrtc.org> | Tue Jul 14 16:00:17 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Jul 21 14:01:27 2020 |
tree | d741d054e39b63e0f2aac38f851c8cb822239157 | |
parent | 0bc68bd1648454f5ac2987cdf2014b71d1a80e07 [diff] |
In DependencyDescriptor rtp header extension drop partial chain support i.e. when chain are used, require each decode target to be protected by some chain. where previously it was allowed to mark decode target as unprotected. See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125 Bug: webrtc:10342 Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31772}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.