42bf2c670c
WebRTC can switch encoder on-fly when encoder fails or by request from encoder selector. Putting the current send codec to the front of the codecs list provides a simple way for apps to know what is actually used without retrieving stats. Bug: webrtc:13572 Change-Id: Iaaa5f7ad8667f59016dc92bff9e9a57a7425ef44 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246500 Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35723}
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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation