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 4007 matching lines @@
   FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport);
   EXPECT_EQ(-1, default_receive_config.payload_type)
       << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
   EXPECT_EQ(0U, default_receive_config.remote_ssrc)
       << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
   EXPECT_TRUE(default_receive_config.protected_media_ssrcs.empty())
       << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
 }
 
 TEST_P(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
-  static const int kExtensionId = 8;
+  static constexpr int kExtensionId = 8;
+  static constexpr size_t kMinPacketsToWaitFor = 50;
   class TransportSequenceNumberTest : public test::EndToEndTest {
    public:
     TransportSequenceNumberTest()
         : EndToEndTest(kDefaultTimeoutMs),
           video_observed_(false),
           audio_observed_(false) {
       parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
                                           kExtensionId);
     }
 
@@ skipping 17 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() >= kMinPacketsToWaitFor) {
         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(), kMinPacketsToWaitFor);
     }
 
    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