Make EndToEndTest.ContentTypeSwitches more robust by sending more frames before test ends

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 2702 matching lines @@
       }
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (MinNumberOfFramesReceived())
         observation_complete_.Set();
       return SEND_PACKET;
     }
 
     bool MinNumberOfFramesReceived() const {
-      const int kMinRequiredHistogramSamples = 200;
+      // Have some room for frames with wrong content type during switch.
+      const int kMinRequiredHistogramSamples = 200+50;
       rtc::CritScope lock(&crit_);
       return num_frames_received_ > kMinRequiredHistogramSamples;
     }
 
     // May be called several times.
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets.";
       // Reset frame counter so next PerformTest() call will do something.
       {
         rtc::CritScope lock(&crit_);
@@ skipping 1729 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