Add chromium support for MediaStreamAudioDestinationNode - part II

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 7486e083ccda1410968b7a32d34f30b8ded4b2b8..5d64781e61cba2fab8696e1141ce90840ca6cdc3 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -113,9 +113,34 @@ void WebRtcAudioCapturer::SetCapturerSource(
   }
 
   // Detach the old source from normal recording.
-  if (old_source)
+  if (old_source) {
     old_source->Stop();
 
+    // TODO(henrika, crogers): I have hard coded parameters given knowledge
+    // of a file source (44.1kHz, mono) here and use these parameters to reset
+    // our member parameters. I then dispatch the new parameters both to the
+    // sink(s) and to the new source. The idea is to get rid of any dependency
+    // of the microphone parameters which are uses as base otherwise.
+
+    // I guess we could add this info to AudioCapturerSource to enable a
+    // query here. E.g.:
+
+    // source->ModifyAudioParameters(&params_);
+
+    params_.Reset(params_.format(),
+                  media::CHANNEL_LAYOUT_MONO,  // <-- get this from WebKit?
+                  44100,                       // <-- get this from WebKit?
+                  16,                          // <-- get this from WebKit?
+                  440); // requires knowledge about WebRTC @ 10ms
+
+    buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]);
+
+    for (SinkList::const_iterator it = sinks_.begin();
+         it != sinks_.end(); ++it) {
+      (*it)->SetCaptureFormat(params_);
+    }
+  }
+
   if (source)
     source->Initialize(params_, this, this);
 }