Changes synchronization offset perfomance tracking

webrtc/call/call_perf_tests.cc

Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 50e1a62cfb512f10a0b47f93c25631b3906c42af..aa2264f1550c0f758344da099a1eea77f0dc4855 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -98,8 +98,6 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
       return;
 
     int64_t now_ms = clock_->TimeInMilliseconds();
-
-    sync_offset_ms_list_.push_back(stats.sync_offset_ms);

[stefan-webrtc, 2016/06/08 11:02:25]

Maybe we should still track this, but track it only after it has been in sync
once? That is, when first_time_in_sync_ != -1.

That way we can track the accuracy and variance of it.

[danilchap, 2016/06/08 11:25:16]

yes, that make sense (I missed the fact that test still run for few seconds
after avsync converges, so there will be more than value)

     int64_t time_since_creation = now_ms - creation_time_ms_;
     // During the first couple of seconds audio and video can falsely be
     // estimated as being synchronized. We don't want to trigger on those.
@@ -125,19 +123,12 @@ class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
     receive_stream_ = receive_stream;
   }
 
-  void PrintResults() {
-    test::PrintResultList("stream_offset", "", "synchronization",
-                          test::ValuesToString(sync_offset_ms_list_), "ms",
-                          false);
-  }
-
  private:
   Clock* const clock_;
   const int64_t creation_time_ms_;
   int64_t first_time_in_sync_;
   rtc::CriticalSection crit_;
   VideoReceiveStream* receive_stream_ GUARDED_BY(crit_);
-  std::vector<int> sync_offset_ms_list_;
 };
 
 void CallPerfTest::TestAudioVideoSync(FecMode fec,
@@ -299,7 +290,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
 
   VoiceEngine::Delete(voice_engine);
 
-  observer.PrintResults();
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
 }