commit | bbbe4e1a15edbbfe99bb13f69593414865f720f2 | [log] [tgz] |
---|---|---|
author | Alex Narest <alexnarest@webrtc.org> | Fri Jul 13 08:32:58 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Jul 13 09:06:06 2018 |
tree | 00296e593d15fd96e90054631ca0f05fd521e259 | |
parent | e9a18b263570b8478207e2006a448c61b3f6655b [diff] |
Better handle target audio bitrate allocation. Do not update target audio bitrate if WebRTC-Audio-SendSideBwe-For-Video is enabled but other side does not support TWCC Bug: webrtc:8243 Change-Id: I6c3c4f223dc5168d726996324717d7ba9ec96e6c Reviewed-on: https://webrtc-review.googlesource.com/88440 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23963}
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.