82f359e45d
From when callTest's send_transport_ is deleted and until the test is completely ended, there is a possibility that the background task webrtc::ModuleRtpRtcpImpl2::MaybeSendRtcpAtOrAfterTimestamp will call send_transport_ which has already been deleted. Fix this by deleting rtp_rtcp_ before send_transport_ is deleted. Bug: webrtc:14202 Change-Id: Ief96c4712875beb55ef232a8bce990d1e9e9efe1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266300 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#37633}
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