cf4ed1516e
GetRtpExtensions() is still used in one corner case for audio receive streams, so GetRtpExtensions has migrated to AudioReceiveStream. Updated FlexfecReceiveStream config management (incl. pass by value) and now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl. Call GetRtpExtensionMap() from call.cc instead of constructing one on the fly for each rtp packet (for video packets at least). Bug: webrtc:11993 Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36839}
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