Add separate send-side UMA stats for screenshare and video.

talk/media/webrtc/webrtcvideoengine2.cc

Index: talk/media/webrtc/webrtcvideoengine2.cc
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 0f1587d54fbe0ec208f88f633a55760b956b5492..6cbb0b83b8d387e9d17e089fb2b3caaab1f24ff8 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -2142,6 +2142,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
     // Configured using the same parameters, do not reconfigure.
     return;
   }
+  bool switched_content_type = last_dimensions_.is_screencast != is_screencast;
   LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
                << (is_screencast ? " (screencast)" : " (not screencast)");
 
@@ -2149,6 +2150,12 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
   last_dimensions_.height = height;
   last_dimensions_.is_screencast = is_screencast;
 
+  if (switched_content_type) {
+    // If we change content type, reset entire stream so that state is reset
+    // (stats flushed etc).
+    RecreateWebRtcStream();

[mflodman, 2015/12/02 08:59:13]

As I read the code, this will generate a new sequence number and timestamp base
for the same ssrc and srtp instance and won't work well.

[mflodman, 2015/12/02 09:12:25]

Now I agree this seems to work, given the magic in VideoSendStream. I'd still
like to discuss this though.

+  }
+
   RTC_DCHECK(!parameters_.encoder_config.streams.empty());
 
   RTC_CHECK(parameters_.codec_settings);