Add UMA for send bwe and pacer bitrate.

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 1926 matching lines @@
     bool use_red_;
     Call* sender_call_;
     Call* receiver_call_;
     int64_t start_runtime_ms_;
   } test(use_rtx, use_red);
 
   test::ClearHistograms();
   RunBaseTest(&test, FakeNetworkPipe::Config());
 
   // Delete the call for Call stats to be reported.
+  sender_call_.reset();
   receiver_call_.reset();
 
   // Verify that stats have been updated once.
   EXPECT_EQ(
       1, test::NumHistogramSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
   EXPECT_EQ(1,
             test::NumHistogramSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.BitrateReceivedInKbps"));
+  EXPECT_EQ(
+      1, test::NumHistogramSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.PacerBitrateInKbps"));
 
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.NackPacketsSentPerMinute"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.NackPacketsReceivedPerMinute"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.FirPacketsSentPerMinute"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.FirPacketsReceivedPerMinute"));
   EXPECT_EQ(1, test::NumHistogramSamples(
@@ skipping 1215 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc