Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx

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 2107 matching lines @@
         : EndToEndTest(kLongTimeoutMs),
           use_rtx_(use_rtx),
           use_red_(use_red),
           screenshare_(screenshare),
           // This test uses NACK, so to send FEC we can't use a fake encoder.
           vp8_encoder_(
               use_red ? VideoEncoder::Create(VideoEncoder::EncoderType::kVp8)
                       : nullptr),
           sender_call_(nullptr),
           receiver_call_(nullptr),
-          start_runtime_ms_(-1) {}
+          start_runtime_ms_(-1),
+          num_frames_received_(0) {}
 
    private:
-    void OnFrame(const VideoFrame& video_frame) override {}
+    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
+      // |ntp_time_ms| is valid.
+      if (video_frame.ntp_time_ms() > 0 &&
+          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
+              video_frame.ntp_time_ms()) {
+        rtc::CritScope lock(&crit_);
+        ++num_frames_received_;
+      }
+    }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      if (MinMetricRunTimePassed())
+      if (MinMetricRunTimePassed() && MinNumberOfFramesReceived())
         observation_complete_.Set();
 
       return SEND_PACKET;
     }
 
     bool MinMetricRunTimePassed() {
       int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
       if (start_runtime_ms_ == -1) {
         start_runtime_ms_ = now;
         return false;
       }
       int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
       return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
     }
 
+    bool MinNumberOfFramesReceived() const {
+      const int kMinRequiredHistogramSamples = 200;
+      rtc::CritScope lock(&crit_);
+      return num_frames_received_ > kMinRequiredHistogramSamples;
+    }
+
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       static const int kExtensionId = 8;
       send_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
       (*receive_configs)[0].rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
       // NACK
@@ skipping 29 matching lines @@
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
     }
 
+    rtc::CriticalSection crit_;
     const bool use_rtx_;
     const bool use_red_;
     const bool screenshare_;
     const std::unique_ptr<VideoEncoder> vp8_encoder_;
     Call* sender_call_;
     Call* receiver_call_;
     int64_t start_runtime_ms_;
+    int num_frames_received_ GUARDED_BY(&crit_);
   } test(use_rtx, use_red, screenshare);
 
   metrics::Reset();
   RunBaseTest(&test);
 
   // Delete the call for Call stats to be reported.
   sender_call_.reset();
   receiver_call_.reset();
 
   std::string video_prefix =
@@ skipping 1707 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