commit 3533bfcb944d520b8491ff46e025c54334f15d66
author buildbot@webrtc.org <buildbot@webrtc.org> Tue Aug 26 15:50:23 2014 +0000
committer buildbot@webrtc.org <buildbot@webrtc.org> Tue Aug 26 15:50:23 2014 +0000
tree 00e789f222e2d9f2e5bf7825ae253203cb5c9ad6
parent 4470d78c9ba6a12ef25c6d66dcf4d3eab4e0e57c

(Auto)update libjingle 74132319-> 74133664

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6983 4adac7df-926f-26a2-2b94-8c16560cd09d

talk/media/webrtc/webrtcvideoengine_unittest.cc

diff --git a/talk/media/webrtc/webrtcvideoengine_unittest.cc b/talk/media/webrtc/webrtcvideoengine_unittest.cc
index c621d6d..35f05e1 100644
--- a/talk/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine_unittest.cc
@@ -2380,9 +2380,6 @@
   EXPECT_FALSE(channel_->SetRecvCodecs(codecs));
 }
 
-// Disable for TSan v2, see
-// https://code.google.com/p/webrtc/issues/detail?id=3671 for details.
-#if !defined(THREAD_SANITIZER)
 TEST_F(WebRtcVideoMediaChannelTest, GetRtpSendTimeExtension) {
   // Enable RTP timestamp extension.
   const int id = 12;
@@ -2394,7 +2391,6 @@
   EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
   EXPECT_EQ(id, channel_->GetRtpSendTimeExtnId());
 }
-#endif  // if !defined(THREAD_SANITIZER)
 
 TEST_F(WebRtcVideoMediaChannelTest, SetSend) {
   Base::SetSend();