cdecc4e6df
This code was extracted to make the next following CL easier to review. This CL simply exposes the getters, setters and callbacks to set the buffered amount low threshold on a specific SCTP stream. It will be used in a follow-up CL, but is just boilerplate. Bug: chromium:40072842 Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41943}
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