Mark UMA stat WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds obsolete

Client-side cleanup:
https://webrtc-review.googlesource.com/c/src/+/111506

Bug: webrtc:7882
Change-Id: I7985e72c4455c39b54ce87e1ca56bce995d1b494
Reviewed-on: https://chromium-review.googlesource.com/c/1344141Reviewed-by: Alexei Svitkine <asvitkine@chromium.org>
Commit-Queue: Sam Zackrisson <saza@chromium.org>
Cr-Commit-Position: refs/heads/master@{#610839}

tools/metrics/histograms/histograms.xml

@@ -123428,6 +123428,9 @@ uploading your change for review.
 </histogram>
 
 <histogram name="WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds" units="s">
+  <obsolete>
+    Deprecated 11/2018 due to little use and high implementation complexity.
+  </obsolete>
   <owner>saza@chromium.org</owner>
   <summary>
     The amount of time between sending the first and the last audio RTP packets