Add received audio and video call duration metrics based on packets.

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 2626 matching lines @@
 
   // Delete the call for Call stats to be reported.
   sender_call_.reset();
   receiver_call_.reset();
 
   std::string video_prefix =
       screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
 
   // Verify that stats have been updated once.
   EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
+  EXPECT_EQ(1, metrics::NumSamples(
+                   "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
 
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
   EXPECT_EQ(1,
             metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
 
@@ skipping 2034 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