Sebastian Jansson c235a8d2bb Adds trial to always send padding packets when not sending video.
This can be used to avoid getting stuck in a state where the encoder
is paused due to low bandwidth estimate which means no additional
feedback is received to update the bandwidth estimate. This could
happen if feedback packets are lost.

Bug: webrtc:8415
Change-Id: I59cd60c0277e8b31a6b911b25e8e488af9008fc2
Reviewed-on: https://webrtc-review.googlesource.com/80880
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23632}
2018-06-15 13:29:21 +00:00
2018-05-30 08:30:00 +00:00
2018-05-15 16:41:02 +00:00
2018-06-13 14:43:23 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2018-05-18 15:57:56 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-06-08 10:31:38 +00:00
2018-04-19 06:52:18 +00:00
2018-02-23 10:34:16 +00:00

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.

Development

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.

More info

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