Storing raw audio sink for default audio track.

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 9184b93e099bc3df8338212ec668fae55a8006ab..74414eec6b11f0634e0df8700d754b97cdfab46d 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -14,6 +14,7 @@
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@@ -193,7 +194,7 @@ public:
         CriticalSectionWrapper* callbackCritSect);
     int32_t UpdateLocalTimeStamp();
 
-    void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+    void SetSink(rtc::scoped_refptr<AudioSinkInterface> sink);
 
     // API methods
 
@@ -511,7 +512,7 @@ private:
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
-    rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
+    rtc::scoped_refptr<AudioSinkInterface> audio_sink_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;