| commit | 254d869c00479d07d073fdba1779c2deff1513dd | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Wed Nov 21 18:19:00 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Nov 22 17:06:52 2018 |
| tree | 81ced262af39fd9d0572858965057bd7583e9dda | |
| parent | 3890c1ae6da24e15b252b7a36f8632d8e7e40e0c [diff] |
Routing BitrateAllocationUpdate to audio codec. This will be used in a later CL to use the link capacity field in the update to control the Opus encoder. Bug: webrtc:9718 Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba Reviewed-on: https://webrtc-review.googlesource.com/c/111510 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25761}
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.