UMA stats for browser/renderer audio rendering buffer size mismatch.

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index e3ce2b0f6775112015b1586bed60f1a90d0f97d2..2f8f83ef83f0db7a262390cfd6c49d05e71ba56e 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -651,6 +651,9 @@ void WebRtcAudioRenderer::PrepareSink() {
     sink_params_ = new_sink_params;
   }
 
+  // Specify the latency info to be passed to the browser side.
+  new_sink_params.set_latency_tag(AudioDeviceFactory::GetSourceLatencyType(
+      AudioDeviceFactory::AudioDeviceFactory::kSourceWebRtc));
   sink_->Initialize(new_sink_params, this);
 }