Chromium Code Reviews| Index: webrtc/modules/audio_processing/audio_processing_performance_unittest.cc |
| diff --git a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc |
| index bb88c8be4239fd3ebe8c86191ce1023ed8cc3367..9ef353ed9ff3105edffcc8bb604e2277ca44cf31 100644 |
| --- a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc |
| +++ b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc |
| @@ -714,7 +714,7 @@ const float CallSimulator::kRenderInputFloatLevel = 0.5f; |
| const float CallSimulator::kCaptureInputFloatLevel = 0.03125f; |
| } // anonymous namespace |
| -TEST_P(CallSimulator, ApiCallDurationTest) { |
| +TEST_P(CallSimulator, DISABLED_ApiCallDurationTest) { |
|
hlundin-webrtc
2017/05/29 12:57:25
Please, add a comment with reference to issue 7712
peah-webrtc
2017/07/03 15:48:34
Done.
|
| // Run test and verify that it did not time out. |
| EXPECT_EQ(kEventSignaled, Run()); |
| } |