| commit | 987ef482582660003d21ecfb7096943cba623373 | [log] [tgz] |
|---|---|---|
| author | Björn Terelius <terelius@webrtc.org> | Thu Mar 05 15:52:10 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 05 16:29:55 2020 |
| tree | a1f073d70e9cca84c3a2c3fca1dd6cff6439e3bb | |
| parent | 822c3739863d964a832b665fe2e0cdc0ebc08525 [diff] |
Adds field trial to separate audio and video packets for delay-based overuse detection. The decision to route audio packets to a separate overuse detector is off by default and requires the field trial WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/ The parameters control the threshold for switching over to the audio overuse detector if we stop receiving feedback for video. Bug: webrtc:10932 Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30694}
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.