Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo

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 4045 matching lines @@
       // Unwrap packet id and verify uniqueness.
       int64_t packet_id =
           unwrapper_.Unwrap(header.extension.transportSequenceNumber);
       EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
 
       if (header.ssrc == kVideoSendSsrcs[0])
         video_observed_ = true;
       if (header.ssrc == kAudioSendSsrc)
         audio_observed_ = true;
       if (audio_observed_ && video_observed_ &&
-          received_packet_ids_.size() == 50) {
+          received_packet_ids_.size() >= 50) {

[åsapersson, 2017/03/03 11:16:19]

maybe add a constant for 50

[danilchap, 2017/03/03 12:14:11]

Done.

         size_t packet_id_range =
             *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
         EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
         observation_complete_.Set();
       }
       return SEND_PACKET;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
                              "packets with transport sequence number.";
+      // Double check conditions for successfull test to produce better error
+      // message when this test fail.
+      EXPECT_TRUE(video_observed_);
+      EXPECT_TRUE(audio_observed_);
+      EXPECT_GE(received_packet_ids_.size(), 50u);
     }
 
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
@@ skipping 93 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