commit | 65d9c4d761b90fd1f7dc9438b6dc1df7ff0899d6 | [log] [tgz] |
---|---|---|
author | Sergey Silkin <ssilkin@webrtc.org> | Wed Jun 12 09:02:30 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jun 12 13:34:24 2019 |
tree | c5855ef17b5fd4a93b878e0c27cc25fad030b33b | |
parent | f53cfa9ebef2dfe6c3d6f17242c90617ec38f703 [diff] |
Create rate allocator after codec bitrates are set. Before this change the max bitrate could be updated after it was passed to rate allocator. Bug: none Change-Id: I742fca0f122bef3e95c1a768d6e844f8c28b6279 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141661 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28253}
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.