Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

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 2083 matching lines @@
 
    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (MinMetricRunTimePassed())
         observation_complete_.Set();
 
       // GetStats calls GetSendChannelRtcpStatistics
       // (via VideoSendStream::GetRtt) which updates ReportBlockStats used by
       // WebRTC.Video.SentPacketsLostInPercent.
       // TODO(asapersson): Remove dependency on calling GetStats.
       sender_call_->GetStats();

[åsapersson, 2016/02/08 09:26:14]

can this be removed?

[sprang, 2016/02/08 10:34:24]

Done.

 
       return SEND_PACKET;
     }
 
     bool MinMetricRunTimePassed() {
       int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
       if (start_runtime_ms_ == -1) {
         start_runtime_ms_ = now;
         return false;
       }
@@ skipping 77 matching lines @@
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.PliPacketsSentPerMinute"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.PliPacketsReceivedPerMinute"));
 
   EXPECT_EQ(
       1, test::NumHistogramSamples(video_prefix + "KeyFramesSentInPermille"));
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.KeyFramesReceivedInPermille"));
 
-  EXPECT_EQ(1, test::NumHistogramSamples(
-      "WebRTC.Video.SentPacketsLostInPercent"));
+  if (screenshare) {

[åsapersson, 2016/02/08 09:26:14]

use video_prefix?

[sprang, 2016/02/08 10:34:24]

Done.

+    EXPECT_EQ(1, test::NumHistogramSamples(
+                     "WebRTC.Video.Screenshare.SentPacketsLostInPercent"));
+  } else {
+    EXPECT_EQ(
+        1, test::NumHistogramSamples("WebRTC.Video.SentPacketsLostInPercent"));
+  }
   EXPECT_EQ(1, test::NumHistogramSamples(
       "WebRTC.Video.ReceivedPacketsLostInPercent"));
 
   EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "InputWidthInPixels"));
   EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "InputHeightInPixels"));
   EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentWidthInPixels"));
   EXPECT_EQ(1, test::NumHistogramSamples(video_prefix + "SentHeightInPixels"));
   EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.ReceivedWidthInPixels"));
   EXPECT_EQ(1,
             test::NumHistogramSamples("WebRTC.Video.ReceivedHeightInPixels"));
@@ skipping 1300 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc