Disable some Opus tests pending an update

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 453 matching lines @@
 }
 
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     defined(WEBRTC_CODEC_OPUS) &&                                   \
     !WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #define MAYBE_TestOpusBitExactness TestOpusBitExactness
 #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");
 
   const std::string output_checksum = PlatformChecksum(
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e");
 
   const std::string network_stats_checksum =
@@ skipping 1141 matching lines @@
   // Pull out data once.
   AudioFrame output;
   bool muted;
   ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
 
   EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
             neteq_->LastDecodedTimestamps());
 }
 
 }  // namespace webrtc