Index: webrtc/audio_receive_stream.h |
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h |
index d3ca1b72b976dfd6400717ae0b240e22b18e499e..daf45985d33c2523f85ee26891f4134643b66243 100644 |
--- a/webrtc/audio_receive_stream.h |
+++ b/webrtc/audio_receive_stream.h |
@@ -15,7 +15,7 @@ |
#include <string> |
#include <vector> |
-#include "webrtc/base/scoped_ref_ptr.h" |
+#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/config.h" |
#include "webrtc/stream.h" |
#include "webrtc/transport.h" |
@@ -106,12 +106,12 @@ |
// Sets an audio sink that receives unmixed audio from the receive stream. |
// Ownership of the sink is passed to the stream and can be used by the |
// caller to do lifetime management (i.e. when the sink's dtor is called). |
- // Only one sink can be set and passing a null sink clears an existing one. |
+ // Only one sink can be set and passing a null sink, clears an existing one. |
// NOTE: Audio must still somehow be pulled through AudioTransport for audio |
// to stream through this sink. In practice, this happens if mixed audio |
// is being pulled+rendered and/or if audio is being pulled for the purposes |
// of feeding to the AEC. |
- virtual void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) = 0; |
+ virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; |
}; |
} // namespace webrtc |