60b6c1dfa9
This fixes a crash that occurs with this sequence of events: 1. AddTrack. SetLocalDescription(CreateOffer()) 2. RemoveTrack. SetLocalDescription(CreateOffer()) 3. AddTrack. When AddTrack is called again it re-uses the RtpTransceiver/ RtpSender and try to configure the underlying MediaChannel. But the MediaChannel would DCHECK since the send stream had been destroyed by the SLD in 2. and would not get created until SLD is called again. Bug: webrtc:9401 Change-Id: I4b5572886e17263aaa4ce0408663444d72e09243 Reviewed-on: https://webrtc-review.googlesource.com/83420 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23605}
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