commit | a881ac0ec9e79874e6b2e97cac5f66ca24958076 | [log] [tgz] |
---|---|---|
author | Seth Hampson <shampson@webrtc.org> | Mon Feb 12 22:14:39 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Feb 13 19:47:56 2018 |
tree | 3cb9fc057eb0581d07cbe8b15478236eaef1f2b0 | |
parent | 73f29cbcc104063b7297f785bf9bab417e72fbfb [diff] |
Updated comments for RtpEncodingParameters. Currently with the RtpEncodingParameters the active field is supported per simulcast layer, but max_bitrate_bps and bitrate_priority are just supoorted per rtp sender. Updated the comments to make this more clear and added TODOs with bugs. Bug: webrtc:8819 Change-Id: I130f6dda0896079b5082af58a2693b898d6e22f0 Reviewed-on: https://webrtc-review.googlesource.com/52141 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22007}
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.