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

content/renderer/media/webrtc_audio_renderer.cc

 // 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.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ skipping 633 matching lines @@
         (audio_fifo_ &&
          audio_fifo_->SizeInFrames() != source_frames_per_buffer)) {
       audio_fifo_.reset(new media::AudioPullFifo(
           kChannels, source_frames_per_buffer,
           base::Bind(&WebRtcAudioRenderer::SourceCallback,
                      base::Unretained(this))));
     }
     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);
 }
 
 }  // namespace content