| commit | 914351de5c82bcb11da179bc65673fb28a5fa449 | [log] [tgz] |
|---|---|---|
| author | Per Kjellander <perkj@webrtc.org> | Fri Feb 15 09:54:55 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Feb 15 10:57:38 2019 |
| tree | fde71b948dafe4963b4383cbd20d3eb6e013c87a | |
| parent | 106d92d4c930088d2e7ae6ca74fa863a5edb9db8 [diff] |
Reland "Always offer transport sequence number header extension for audio"" (reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1) Original cl description: 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. Patchset 3 contain the only change: Add the field trial WebRTC-Audio-SendSideBwe to call/rampup_tests.cc TBR: srte@webrtc.org,ossu@webrtc.org Bug: webrtc:10309 webrtc:10286 Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f Reviewed-on: https://webrtc-review.googlesource.com/c/123183 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26706}
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.