| Index: webrtc/test/fake_audio_device.h
|
| diff --git a/webrtc/test/fake_audio_device.h b/webrtc/test/fake_audio_device.h
|
| index 180abf6c924c7a2e57e5be17a9cd9dc138218049..39f9310cd2d994112e51c8cf6f12972a55056cb7 100644
|
| --- a/webrtc/test/fake_audio_device.h
|
| +++ b/webrtc/test/fake_audio_device.h
|
| @@ -10,6 +10,7 @@
|
| #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
|
| #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
|
|
|
| +#include <memory>
|
| #include <string>
|
|
|
| #include "webrtc/base/criticalsection.h"
|
| @@ -59,11 +60,11 @@ class FakeAudioDevice : public FakeAudioDeviceModule {
|
| int64_t last_playout_ms_;
|
|
|
| DriftingClock clock_;
|
| - rtc::scoped_ptr<EventTimerWrapper> tick_;
|
| + std::unique_ptr<EventTimerWrapper> tick_;
|
| rtc::CriticalSection lock_;
|
| rtc::PlatformThread thread_;
|
| - rtc::scoped_ptr<ModuleFileUtility> file_utility_;
|
| - rtc::scoped_ptr<FileWrapper> input_stream_;
|
| + std::unique_ptr<ModuleFileUtility> file_utility_;
|
| + std::unique_ptr<FileWrapper> input_stream_;
|
| };
|
| } // namespace test
|
| } // namespace webrtc
|
|
|