commit | 1e51a388bcf7b644bf4f1a2fb09363f866c67fcf | [log] [tgz] |
---|---|---|
author | Erik Språng <sprang@webrtc.org> | Wed Dec 11 15:47:09 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Dec 11 16:32:14 2019 |
tree | 40e8a382cc2208933040be1db2c41d8be870f076 | |
parent | 184da528a739e270baf3ca866409f8e0c7d77a6f [diff] |
Makes padding prefer video SSRCs instead of audio. Some clients will not count audio packets into the bandwidth estimate despite negotiating e.g. abs-send-time for that SSRC. If padding is sent on such an RTP module, we might get stuck in a low resolution. This CL works around that by preferring to send padding on video SSRCs. Bug: webrtc:11196 Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30066}
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.