commit | fd965c008c7bc395bb276f260262ac11ccd25406 | [log] [tgz] |
---|---|---|
author | Per Kjellander <perkj@webrtc.org> | Thu Feb 14 11:54:55 2019 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 14 15:28:07 2019 +0000 |
tree | 30922bd01ad06e8a37e8faeafca2e6881d493714 | |
parent | 92e7c69c28b4893686fe484e4b1e7c17759b4577 [diff] |
Always offer transport sequence number header extension for audio If the extension is negotiated, it will only be used if the field trial WebRTC-Audio-SendSideBwe is enabled. This allows simpler experimentation if it should be used or not. Bug: webrtc:10309 webrtc:10286 Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b Reviewed-on: https://webrtc-review.googlesource.com/c/122542 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26689}
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.