Disable failing modules_unittests for UBSan.

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 496 matching lines @@
                                     size_t* payload_len) {
   rtp_info->header.sequenceNumber = frame_index;
   rtp_info->header.timestamp = timestamp;
   rtp_info->header.ssrc = 0x1234;  // Just an arbitrary SSRC.
   rtp_info->header.payloadType = 98;  // WB CNG.
   rtp_info->header.markerBit = 0;
   payload[0] = 64;  // Noise level -64 dBov, quite arbitrarily chosen.
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
+// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820
 #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
     (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) &&    \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) &&        \
-    !defined(WEBRTC_ARCH_ARM64)
+    !defined(WEBRTC_ARCH_ARM64) && !defined(UNDEFINED_SANITIZER)
 #define MAYBE_TestBitExactness TestBitExactness
 #else
 #define MAYBE_TestBitExactness DISABLED_TestBitExactness
 #endif
 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
@@ skipping 14 matching lines @@
   if (FLAGS_gen_ref) {
     DecodeAndCompare(input_rtp_file, "", "", "");
   } else {
     DecodeAndCompare(input_rtp_file,
                      input_ref_file,
                      network_stat_ref_file,
                      rtcp_stat_ref_file);
   }
 }
 
+// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
-    defined(WEBRTC_CODEC_OPUS)
+    defined(WEBRTC_CODEC_OPUS) && !defined(UNDEFINED_SANITIZER)
 #define MAYBE_TestOpusBitExactness TestOpusBitExactness
 #else
 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
 #endif
 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
   const std::string input_ref_file =
       // The pcm files were generated by using Opus v1.1.2 to decode the RTC
       // file generated by Opus v1.1
@@ skipping 1005 matching lines @@
     timestamp += kSamples;
 
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
 }  // namespace webrtc