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authorkjellander@webrtc.org <kjellander@webrtc.org>2014-08-25 19:46:26 +0000
committerkjellander@webrtc.org <kjellander@webrtc.org>2014-08-25 19:46:26 +0000
commit146a2419c7f0488d77c1a0b40793d9a578469d4b (patch)
tree6d7d05cacb1408c94a99bd631f83ec70f8b8fa59 /app
parent367913cd6c063a5e918198e7fe9b2869e80a449c (diff)
downloadtalk-146a2419c7f0488d77c1a0b40793d9a578469d4b.tar.gz
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
Diffstat (limited to 'app')
-rw-r--r--app/webrtc/peerconnection_unittest.cc3
1 files changed, 2 insertions, 1 deletions
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 @@ 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);