f83a94a41e
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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.