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 0f2f59e4928f0ea619adb5e7088fc3b524388bf5..96d8946c1b5d87fd915bb6a46da80022b535db81 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -198,7 +198,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
 
   void SetRawAudioSink(
       uint32_t ssrc,
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+      rtc::scoped_refptr<webrtc::AudioSinkInterface> sink) override;
 
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
@@ -269,6 +269,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   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 default stream, which may be set before the stream exists.

[the sun, 2016/01/08 13:14:26]

nit: default->unsignalled since that is the terminology used above, for volume
etc.

[Taylor Brandstetter, 2016/01/08 19:46:58]

Done.

+  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