69dd142797
Currently, the RtpRtcp module of AudioSendStream is (de)registered in the packet router on calls to (Register|Reset)SenderCongestionControlObjects. This CL changes that to happen on Start/Stop instead, which allows us to safely call (Get|Set)RtpState on suspend/resume without the need for extra locking in the rtp module. See also https://webrtc-review.googlesource.com/c/src/+/228430 Bug: webrtc:11340 Change-Id: I54243a9ace8a7659924269418468b49b967b9465 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228433 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34738}
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