a136ed4085
Setting the transport cc flag was only possible post creation for audio receive streams, while video receive streams need to be recreated. This CL moves the setter for transport_cc() to where the getter is and adds boiler plate implementations for the video streams. For audio streams this splits "SetUseTransportCcAndNackHistory" into two methods, SetTransportCc and SetNackHistory. Bug: none Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37038}
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