commit | f83a94a41e82c27b01166d590a422ca63a0a7bca | [log] [tgz] |
---|---|---|
author | deadbeef <deadbeef@webrtc.org> | Fri Jun 03 16:05:26 2016 -0700 |
committer | Commit bot <commit-bot@chromium.org> | Fri Jun 03 23:05:30 2016 +0000 |
tree | 801fd9de1817aa6edcd56622cbbde100e0c5c343 | |
parent | 080be512944615936058a40419fec726cf4bcb03 [diff] |
Revert of Improving the fake clock and using it to fix a flaky STUN timeout test. (patchset #10 id:180001 of https://codereview.webrtc.org/2024813004/ ) Reason for revert: There seems to be a TSan warning that wasn't caught by the trybot: https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/6732/steps/peerconnection_unittests/logs/stdio Apparently a thread is still alive even after destroying WebRTCSession. Need to think of a way to fix this, without adding a critical section around g_clock (since that would hurt performance). Original issue's description: > Improving the fake clock and using it to fix a flaky STUN timeout test. > > When the fake clock's time is advanced, it now ensures all pending > queued messages have been dispatched. This allows us to write a > "SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up > until the target time. > > Useful in this case, where we know the STUN timeout should take a total > of 9500ms, but it would be overly complex to write test code that waits > for each individual timeout, ensures a STUN packet has been > retransmited, etc. > > (The test described above *should* be written, but it belongs in > p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc). > > Committed: https://crrev.com/ffbe0e17e2c9b7fe101023acf40574dc0c95631a > Cr-Commit-Position: refs/heads/master@{#13043} TBR=pthatcher@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2038213002 Cr-Commit-Position: refs/heads/master@{#13045}
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.