Disabling EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx.

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 3079 matching lines @@
   receive_transport.StopSending();
 
   Stop();
   DestroyStreams();
 }
 
 TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpState) {
   TestRtpStatePreservation(false);
 }
 
-#if defined(WEBRTC_MAC)
-#define MAYBE_RestartingSendStreamPreservesRtpStatesWithRtx \
-  DISABLED_RestartingSendStreamPreservesRtpStatesWithRtx
-#else
-#define MAYBE_RestartingSendStreamPreservesRtpStatesWithRtx \
-  RestartingSendStreamPreservesRtpStatesWithRtx
-#endif
-TEST_F(EndToEndTest, MAYBE_RestartingSendStreamPreservesRtpStatesWithRtx) {
+// This test is flaky. See:
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=4332
+TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpStatesWithRtx) {
   TestRtpStatePreservation(true);
 }
 
 TEST_F(EndToEndTest, RespectsNetworkState) {
   // TODO(pbos): Remove accepted downtime packets etc. when signaling network
   // down blocks until no more packets will be sent.
 
   // Pacer will send from its packet list and then send required padding before
   // checking paused_ again. This should be enough for one round of pacing,
   // otherwise increase.
@@ skipping 439 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc