Revert of Storing raw audio sink for default audio track.

talk/media/webrtc/webrtcvoiceengine.h

Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index cd010f1a468d49a52b239cf50ebc914b496c2f7d..0f2f59e4928f0ea619adb5e7088fc3b524388bf5 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -198,7 +198,7 @@
 
   void SetRawAudioSink(
       uint32_t ssrc,
-      const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) override;
+      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
 
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
@@ -269,8 +269,6 @@
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
-  // Sink for unsignalled stream, which may be set before the stream exists.
-  rtc::scoped_refptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   // and https://code.google.com/p/chromium/issues/detail?id=547661