| commit | d3e765e4eb4af8eba0c63b6bfe2e17a12dc670dc | [log] [tgz] |
|---|---|---|
| author | Henrik Boström <hbos@webrtc.org> | Thu Mar 23 17:24:55 2023 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Mar 23 18:21:42 2023 |
| tree | e2253a56d32d36ba130970afd7a3c0f4c5d32d88 | |
| parent | adb946054c7daba852625f3af90698ec0525b1bb [diff] |
Disable flaky PeerConnectionSimulcastWithMediaFlowTests on ASAN. The test is flaking during shutdown in general, but this is especially apparent on ASAN bots that don't retry. It is related to heap-use-after-free and is believef to be a test-only problem. Bug: webrtc:15018 Change-Id: I593c18dbbe45605404b17af74fd7101f118db7bb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299003 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#39661}
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.