d8c4de7172
VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream. Therefore this cl: - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience. - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing. - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used. Bug: none Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38698}
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