Adding WebRTC.Audio.JitterBufferFullPerMinute to metrics.

Bug: 896193
Change-Id: I046d763c004b2fafa66df76704489a1ab5d5821d
Reviewed-on: https://chromium-review.googlesource.com/c/1286415
Commit-Queue: Minyue Li <minyue@chromium.org>
Reviewed-by: Robert Kaplow (sloooow) <rkaplow@chromium.org>
Cr-Commit-Position: refs/heads/master@{#600745}

tools/metrics/histograms/histograms.xml

@@ -119416,6 +119416,16 @@ uploading your change for review.
   </summary>
 </histogram>
 
+<histogram name="WebRTC.Audio.JitterBufferFullPerMinute" units="events/minute">
+  <owner>minyue@chromium.org</owner>
+  <summary>
+    Frequency that audio packets hits the capacity of WebRTC jitter buffer. A
+    larger value indicates that the capacity is not big enough, and/or audio
+    packets are not processed quickly enough. The metric is recorded every
+    minute.
+  </summary>
+</histogram>
+
 <histogram name="WebRTC.Audio.NumOfAecSystemDelayJumps" units="jumps">
   <owner>hlundin@chromium.org</owner>
   <summary>