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

After calling Stop(), the timer should just stop.


BUG=244192
TEST= https://webrtc.googlecode.com/svn/trunk/samples/js/demos/html/local-audio-rendering.html
Calling Start/Stop, the timer used to start/stop counting.

Review URL: https://chromiumcodereview.appspot.com/16045004

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@202419 0039d316-1c4b-4281-b951-d872f2087c98

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";
     }