Relax expectation in EndToEndTest.CallReportsRttForSender test

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 3489 matching lines @@
   CreateFrameGeneratorCapturer();
   Start();
 
   int64_t start_time_ms = clock_->TimeInMilliseconds();
   while (true) {
     Call::Stats stats = sender_call_->GetStats();
     ASSERT_GE(start_time_ms + kDefaultTimeoutMs,
               clock_->TimeInMilliseconds())
         << "No RTT stats before timeout!";
     if (stats.rtt_ms != -1) {
-      EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs);
+      // To avoid failures caused by rounding or minor ntp clock adjustments,
+      // relax expectation by 1ms.
+      constexpr int kAllowedErrorMs = 1;
+      EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs - kAllowedErrorMs);
       break;
     }
     SleepMs(10);
   }
 
   Stop();
   DestroyStreams();
 }
 
 void EndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
@@ skipping 213 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc