| 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.
|
|
|