Index: talk/app/webrtc/rtpsenderreceiver_unittest.cc |
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc |
index eceed19b89ed0601df4e2379d6274bba9ca8fd77..a590e1d01f8b07df94c5b9304b04fa4a85e1b158 100644 |
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc |
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc |
@@ -70,14 +70,13 @@ |
cricket::AudioRenderer* renderer)); |
MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume)); |
- void SetRawAudioSink( |
- uint32_t, |
- const rtc::scoped_refptr<AudioSinkInterface>& sink) override { |
- sink_ = sink; |
+ void SetRawAudioSink(uint32_t, |
+ rtc::scoped_ptr<AudioSinkInterface> sink) override { |
+ sink_ = std::move(sink); |
} |
private: |
- rtc::scoped_refptr<AudioSinkInterface> sink_; |
+ rtc::scoped_ptr<AudioSinkInterface> sink_; |
}; |
// Helper class to test RtpSender/RtpReceiver. |