| commit | bcf91808a2205e72e9a669ff80f6fe9c15f6d349 | [log] [tgz] |
|---|---|---|
| author | Alex Narest <alexnarest@webrtc.org> | Mon Jun 25 16:08:36 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Jun 27 10:33:40 2018 +0000 |
| tree | 7a697368a51e46c5ef09b860eb3da906507c33d0 | |
| parent | 81f51975125c6c12b6dc9fd470132c78ef4812c9 [diff] |
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead. Bug: webrtc:8243 Change-Id: If961780921d53bdce95b68c26641df6875509c1f Reviewed-on: https://webrtc-review.googlesource.com/84501 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23755}
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.