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

webrtc/video/end_to_end_tests.cc

Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 5385450298716337561a3cfb0676cdec1c7b2d2e..5ba7c1ab77f44b37d3d237f5005203b7badcff69 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2199,8 +2199,13 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx,
   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"));