Index: webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
index eed77405f8261fb44c58858906fba2fe84a3f6fb..667e1fd6613b9a1bdd9c7b1144c9e08cffc1f716 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
@@ -471,7 +471,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { |
#else |
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness |
#endif |
-TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { |
+TEST_F(NetEqDecodingTest, DISABLED_TestOpusBitExactness) { |
const std::string input_rtp_file = |
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); |