| 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.
|
|
|