af74dff19e
This makes it consistent with how things are done in webrtc_video_engine.cc This will improve the JS code by not having to initialize an audio track every time frames need to be sent over, especially from another peer connection in case of encoded transforms. Bug: chromium:1477192 Change-Id: I3f938ad812ff377599a3799d4c2d2cd85149189e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322702 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tony Herre <herre@google.com> Commit-Queue: Palak Agarwal <agpalak@google.com> Cr-Commit-Position: refs/heads/main@{#40917}
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