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

content/renderer/media/webrtc_audio_renderer.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
@@ skipping 232 matching lines @@
   std::string output_device_id_;
   url::Origin security_origin_;
 
   // Maps audio sources to a list of active audio renderers.
   // Pointers to PlayingState objects are only kept in this map while the
   // associated renderer is actually playing the stream.  Ownership of the
   // state objects lies with the renderers and they must leave the playing state
   // 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_;
+  // 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);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_