421088815f
RtpVideoSender now stores fec type and overhead instead of querying the generator all the time. Setting of protection parameters and asking for current bitrate is also now handled just by the VideoFecGenerator instance, instead of going via RtpVideoSender. Finally, adds method to query for RtpState in VideoFecGenerator interface. This avoids an ugly cast that would have been even more trouble after moving fec generation. For context, see https://webrtc-review.googlesource.com/c/src/+/173708 Bug: webrtc:11340 Change-Id: Ia5e6cd919e71850c9cc5ed5a4f4417338d577162 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174203 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31166}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct