Index: webrtc/api/rtpsenderreceiver_unittest.cc |
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc |
index 188264e7b2c4ff2c7b6afb64d26d9a80471fff30..4cd142567ada966f753e33f7e49ff940f8eb4440 100644 |
--- a/webrtc/api/rtpsenderreceiver_unittest.cc |
+++ b/webrtc/api/rtpsenderreceiver_unittest.cc |
@@ -8,6 +8,7 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
+#include <memory> |
#include <string> |
#include <utility> |
@@ -58,12 +59,12 @@ class MockAudioProvider : public AudioProviderInterface { |
bool(uint32_t ssrc, const RtpParameters&)); |
void SetRawAudioSink(uint32_t, |
- rtc::scoped_ptr<AudioSinkInterface> sink) override { |
+ std::unique_ptr<AudioSinkInterface> sink) override { |
sink_ = std::move(sink); |
} |
private: |
- rtc::scoped_ptr<AudioSinkInterface> sink_; |
+ std::unique_ptr<AudioSinkInterface> sink_; |
}; |
// Helper class to test RtpSender/RtpReceiver. |