8dda03515b
Thanks to the CL https://webrtc-review.googlesource.com/c/src/+/83822 there is now no need to destroy the stream asynchronously. But the CL above introduced a leak, the streams were stopped but not destroyed. This CL essentially fixes the leak, it is now safe to destroy the stream right after being stopped as everything happen in the same capture thread. Bug: chromium:851883 Change-Id: I4bf7409246f3957d90040d0d8cf09e98f28d6559 Reviewed-on: https://webrtc-review.googlesource.com/96621 Reviewed-by: Brave Yao <braveyao@webrtc.org> Reviewed-by: Zijie He <zijiehe@chromium.org> Commit-Queue: Brave Yao <braveyao@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24519}
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 http://www.webrtc.org/native-code/development 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