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/talk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/app/webrtc/peerconnection_unittest.cc b/app/webrtc/peerconnection_unittest.cc
index b395a31..8a098b4 100644
--- a/app/webrtc/peerconnection_unittest.cc
+++ b/app/webrtc/peerconnection_unittest.cc
@@ -90,7 +90,6 @@
// 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 @@
// 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);
