commit | 135d9a386e46c2961d5dc068b39289f0d79a589c | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Mon Feb 17 12:21:17 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Feb 17 14:50:58 2020 |
tree | d06d3b2dbe09af68557426659f5e5536b1762acf | |
parent | 0bfaa7fb5f3614cc7b75c99833977fbd79c80d5d [diff] |
Update dependency descriptor rtp header extension uri to match one in av1 rtp spec examples: https://aomediacodec.github.io/av1-rtp-spec/#73-example Bug: webrtc:10342 Change-Id: Ib108b90f6103d050d61d40fc36ad1c2a358f3f21 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168641 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30531}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.