Disable CallReportsRttForSender on Mac.

webrtc/video/end_to_end_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 <algorithm>
@@ skipping 3225 matching lines @@
     NetworkState sender_state_ GUARDED_BY(test_crit_);
     int sender_rtp_ GUARDED_BY(test_crit_);
     int sender_rtcp_ GUARDED_BY(test_crit_);
     int receiver_rtcp_ GUARDED_BY(test_crit_);
     int down_frames_ GUARDED_BY(test_crit_);
   } test;
 
   RunBaseTest(&test);
 }
 
-TEST_F(EndToEndTest, CallReportsRttForSender) {
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5938
+#ifdef WEBRTC_MAC
+#define MAYBE_CallReportsRttForSender DISABLED_CallReportsRttForSender
+#else
+#define MAYBE_CallReportsRttForSender CallReportsRttForSender
+#endif
+TEST_F(EndToEndTest, MAYBE_CallReportsRttForSender) {
   static const int kSendDelayMs = 30;
   static const int kReceiveDelayMs = 70;
 
   CreateCalls(Call::Config(), Call::Config());
 
   FakeNetworkPipe::Config config;
   config.queue_delay_ms = kSendDelayMs;
   test::DirectTransport sender_transport(config, sender_call_.get());
   config.queue_delay_ms = kReceiveDelayMs;
   test::DirectTransport receiver_transport(config, receiver_call_.get());
@@ skipping 241 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc