Fixing the webrtc local audio renderer timer between Stop() and Start().

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index 450a0a59e9987d350a26a3fb712d5a562624c213..f76ca505b0760283f2a200bee264870143cb8e16 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -28,10 +28,6 @@ int WebRtcLocalAudioRenderer::Render(
 
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
 
-  base::Time now = base::Time::Now();
-  total_render_time_ += now - last_render_time_;
-  last_render_time_ = now;
-
   DCHECK(loopback_fifo_.get() != NULL);
 
   // Provide data by reading from the FIFO if the FIFO contains enough
@@ -74,6 +70,10 @@ void WebRtcLocalAudioRenderer::CaptureData(const int16* audio_data,
                                     audio_source->frames(),
                                     sizeof(audio_data[0]));
       loopback_fifo_->Push(audio_source.get());
+
+      base::Time now = base::Time::Now();
+      total_render_time_ += now - last_render_time_;
+      last_render_time_ = now;
     } else {
       DVLOG(1) << "FIFO is full";
     }