| commit | b77c716d8afc3333c8d65b2738cb645ad4ed00f7 | [log] [tgz] |
|---|---|---|
| author | stefan <stefan@webrtc.org> | Mon Feb 06 06:29:38 2017 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Feb 06 14:29:38 2017 +0000 |
| tree | 2aaadd01170c64080042c67f8a25dbd102b1ca5b | |
| parent | fd8d2654d704afa80c2fcecff5f3092c381ed249 [diff] |
Enable send-side BWE by default for video in call tests. Also fixes a bug where RTCP transport feedback was sent even though RTCP was disabled. May affect perf numbers since the behavior of the send-side BWE differs a lot from the recv-side BWE. BUG=webrtc:7111 Review-Url: https://codereview.webrtc.org/2669413003 Cr-Commit-Position: refs/heads/master@{#16451}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.