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 467cbe35f74daf8e6459575f096488f78c166f62..3762a7c3df0ace065529f8570e666d79f773b2ab 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -664,6 +664,9 @@ 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);
 }