From e9bfed0648656a22b41a9357e50a57d3c2d17e14 Mon Sep 17 00:00:00 2001 From: "kjellander@webrtc.org" Date: Mon, 25 Aug 2014 19:46:26 +0000 Subject: [PATCH] Move constant so it is not stripped out for TSAN bots. BUG= R=henrike@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d --- talk/app/webrtc/peerconnection_unittest.cc | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc index b395a315a0..8a098b423c 100644 --- a/talk/app/webrtc/peerconnection_unittest.cc +++ b/talk/app/webrtc/peerconnection_unittest.cc @@ -90,7 +90,6 @@ static const int kMaxWaitMs = 2000; // warnings. #if !defined(THREAD_SANITIZER) static const int kMaxWaitForStatsMs = 3000; -static const int kMaxWaitForAudioDataMs = 10000; static const int kMaxWaitForRembMs = 5000; #endif static const int kMaxWaitForFramesMs = 10000; @@ -1047,6 +1046,8 @@ class P2PTestConductor : public testing::Test { // Wait until 'size' bytes of audio has been seen by the receiver, on the // first audio stream. void WaitForAudioData(int size) { + static const int kMaxWaitForAudioDataMs = 10000; + StreamCollectionInterface* local_streams = initializing_client()->local_streams(); ASSERT_GT(local_streams->count(), 0u);