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 99eb38499b2118d1b87d29a4bc45beb3070db677..0510d70ee8310923ce0d7cfe0a521d8df2a4d76f 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
@@ -492,7 +492,7 @@ 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"); |