| commit | 88532898db9f01bedc3b2caae8b2cf4e7bea8dcb | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Mon Aug 01 21:17:27 2016 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Mon Aug 01 21:17:41 2016 |
| tree | 5040e094507c49e29b3f703caafaec5ef288d133 | |
| parent | 19ee1e6eb102b2282a9993d6c373c7185ad9ba34 [diff] |
Un-flaking TestSrtpError by using a fake clock. This test verifies that SRTP errors are only signaled if a specific amount of time has passed, and was using ProcessMessages(time) to wait for an amount of time. But ProcessMessages may sometimes wait for longer than the requested time (especially on TSAN bot, etc.). BUG=webrtc:4690 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/2184083004 . Cr-Commit-Position: refs/heads/master@{#13597}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.