Adding stub files for RtpSender/RtpReceiver.

talk/app/webrtc/rtpsender.cc

Index: talk/app/webrtc/rtpsender.cc
diff --git a/talk/session/media/voicechannel.h b/talk/app/webrtc/rtpsender.cc
similarity index 90%
copy from talk/session/media/voicechannel.h
copy to talk/app/webrtc/rtpsender.cc
index 6c1b6afdd55aab2df3325297afe0efde2b76b727..3049222f2a4c9d615c03c132dd34d7a69b398012 100644
--- a/talk/session/media/voicechannel.h
+++ b/talk/app/webrtc/rtpsender.cc
@@ -1,6 +1,6 @@
 /*
  * libjingle
- * Copyright 2004 Google Inc.
+ * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
@@ -25,9 +25,6 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#ifndef _VOICECHANNEL_H_
-#define _VOICECHANNEL_H_
+#include "talk/app/webrtc/rtpsender.h"
 
-#include "talk/session/media/channel.h"
-
-#endif // _VOICECHANNEL_H_
+// This file is currently stubbed so that Chromium's build files can be updated.