fc83cdc819
This bypasses the proxy for the following properties: * MediaStream::id() * AudioTrack::kind() and AudioTrack::id() * VideoTrack::kind() and VideoTrack::id() * RtpReceiver::media_type() and RtpReceiver::id() * RtpSender::media_type() and RtpSender::id() * VideoTrackSource::remote() and VideoTrackSource::is_screencast() * RtpTransceiver::media_type() Bug: webrtc:11923 Change-Id: If7edea1781f778af3775515fc4af9a9e151c8103 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183767 Reviewed-by: Chen Xing <chxg@google.com> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32071}
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