Add UMA histograms for time spent getting WebRTC audio render data.

content/renderer/media/webrtc_audio_renderer.h

Index: content/renderer/media/webrtc_audio_renderer.h
diff --git a/content/renderer/media/webrtc_audio_renderer.h b/content/renderer/media/webrtc_audio_renderer.h
index f4f677ee58f58e34b0918217ceae891ee3b8da92..170cd770d265779cf8f2bd96d179eae8ca4e2f01 100644
--- a/content/renderer/media/webrtc_audio_renderer.h
+++ b/content/renderer/media/webrtc_audio_renderer.h
@@ -250,9 +250,13 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // before being destructed (PlayingState object goes out of scope).
   SourcePlayingStates source_playing_states_;
 
-  // Used for triggering new UMA histogram. Counts number of render
-  // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
-  int render_callback_count_;
+  // Flag which is set when the first render call is made to the source. This is
+  // used to not log max render time if we never measured it.
+  bool source_render_called_;
+
+  // Stores the maximum time spent waiting for render data from the source. Used
+  // for logging UMA data. Logged and reset when Stop() is called.
+  base::TimeDelta max_render_time_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };