Disable EndToEndTest.ContentTypeSwitches test on Windows due to flakiness

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 2675 matching lines @@
 
   int num_red_samples = use_red ? 1 : 0;
   EXPECT_EQ(num_red_samples,
             metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
   EXPECT_EQ(num_red_samples,
             metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
   EXPECT_EQ(num_red_samples,
             metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
 }
 
-TEST_F(EndToEndTest, ContentTypeSwitches) {
+#if defined(WEBRTC_WIN)
+// Disabled due to flakiness on Windows (bugs.webrtc.org/7483).
+#define MAYBE_ContentTypeSwitches DISABLED_ContentTypeSwitches
+#else
+#define MAYBE_ContentTypeSwitches ContentTypeSwitches
+#endif
+TEST_F(EndToEndTest, MAYBE_ContentTypeSwitches) {
   class StatsObserver : public test::BaseTest,
                         public rtc::VideoSinkInterface<VideoFrame> {
    public:
     StatsObserver() : BaseTest(kLongTimeoutMs), num_frames_received_(0) {}
 
     bool ShouldCreateReceivers() const override { return true; }
 
     void OnFrame(const VideoFrame& video_frame) override {
       // The RTT is needed to estimate |ntp_time_ms| which is used by
       // end-to-end delay stats. Therefore, start counting received frames once
@@ skipping 1756 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc