| 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");
|
|
|
|
|