| commit | 4182b499b39418376fff7c9a68e3aa457ea68c37 | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Wed Oct 21 00:33:03 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Oct 21 22:18:20 2020 |
| tree | 38cf546449005cda594426cf81ac4ab38dad97ef | |
| parent | 236a89b6591f45439b6c36e908dada6622cd2fb4 [diff] |
Avoid duplicate usrsctp_init if the last usrsctp_finish failed. When deinitializing usrsctp, we attempt to call usrsctp_finish in a loop for three seconds (it may fail because another sctp thread is holding a reference to something). If the three seconds run out, usrsctp is left in a still initialized state, and bad things happen down the road if usrsctp_init is called in the state. Bug: chromium:1138878 Change-Id: I9c24d51d5a274b06bdf4183261694fc2989136c5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32467}
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.
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.