Changes synchronization offset perfomance tracking

webrtc/call/call_perf_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.
  */
 
@@ skipping 80 matching lines @@
     VideoReceiveStream::Stats stats;
     {
       rtc::CritScope lock(&crit_);
       if (receive_stream_)
         stats = receive_stream_->GetStats();
     }
     if (stats.sync_offset_ms == std::numeric_limits<int>::max())
       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.
     if (time_since_creation < kStartupTimeMs)
       return;
     if (std::abs(stats.sync_offset_ms) < kInSyncThresholdMs) {
       if (first_time_in_sync_ == -1) {
         first_time_in_sync_ = now_ms;
         webrtc::test::PrintResult("sync_convergence_time",
                                   "",
                                   "synchronization",
                                   time_since_creation,
                                   "ms",
                                   false);
       }
       if (time_since_creation > kMinRunTimeMs)
         observation_complete_.Set();
     }
   }
 
   void set_receive_stream(VideoReceiveStream* receive_stream) {
     rtc::CritScope lock(&crit_);
     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,
                                       CreateOrder create_first,
                                       float video_ntp_speed,
                                       float video_rtp_speed,
                                       float audio_rtp_speed) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
@@ skipping 141 matching lines @@
 
   voe_base->DeleteChannel(send_channel_id);
   voe_base->DeleteChannel(recv_channel_id);
   voe_base->Release();
   voe_codec->Release();
 
   DestroyCalls();
 
   VoiceEngine::Delete(voice_engine);
 
-  observer.PrintResults();
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
   TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
                      DriftingClock::PercentsFaster(10.0f),
                      DriftingClock::kNoDrift, DriftingClock::kNoDrift);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
@@ skipping 381 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc