8b7ca4abb2
Currently we prefer the last added rtp module that supports rtx, and assume this is the HD stream. If we suffer a network degradation and stop sending HD, the current behavior will trigger RTX padding on an inactive stream, which is not very useful. With this change, we will prefer the rtp module that last sent media, which will spread the load a bit across active media streams, but will be biased toward the one with highest packet rate. Bug: webrtc:8975 Change-Id: Id52865ccd5263722c66d327b8c80457f63b90385 Reviewed-on: https://webrtc-review.googlesource.com/77360 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23281}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct