Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo.

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <functional>
@@ skipping 284 matching lines @@
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 }
 
-TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
+// Crashes when run with ASan. See webrtc:5790.

[pbos-webrtc, 2016/04/15 08:19:58]

Not true, it fails, not crashes

+#if defined(ADDRESS_SANITIZER) && defined(WEBRTC_LINUX)

[pbos-webrtc, 2016/04/15 08:19:59]

I don't think you need WEBRTC_LINUX here, we can disable it for asan everywhere

+#define MAYBE_SwitchesToASTThenBackToTOFForVideo \
+    DISABLED_SwitchesToASTThenBackToTOFForVideo
+#else
+#define MAYBE_SwitchesToASTThenBackToTOFForVideo \
+    SwitchesToASTThenBackToTOFForVideo
+#endif
+TEST_F(BitrateEstimatorTest, MAYBE_SwitchesToASTThenBackToTOFForVideo) {
   video_send_config_.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
   streams_.push_back(new Stream(this, false));
   EXPECT_TRUE(receiver_log_.Wait());
 
   video_send_config_.rtp.extensions[0] =
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc