| Index: webrtc/api/test/fakeaudiocapturemodule.h
|
| diff --git a/webrtc/api/test/fakeaudiocapturemodule.h b/webrtc/api/test/fakeaudiocapturemodule.h
|
| index 9200bdf6f318e414a895436274d60dddcae0a014..30ad3f8b71feaca1b42234b11bcdc3bd4c989984 100644
|
| --- a/webrtc/api/test/fakeaudiocapturemodule.h
|
| +++ b/webrtc/api/test/fakeaudiocapturemodule.h
|
| @@ -20,6 +20,8 @@
|
| #ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
|
| #define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
|
|
|
| +#include <memory>
|
| +
|
| #include "webrtc/base/basictypes.h"
|
| #include "webrtc/base/criticalsection.h"
|
| #include "webrtc/base/messagehandler.h"
|
| @@ -247,7 +249,7 @@ class FakeAudioCaptureModule
|
| bool started_;
|
| uint32_t next_frame_time_;
|
|
|
| - rtc::scoped_ptr<rtc::Thread> process_thread_;
|
| + std::unique_ptr<rtc::Thread> process_thread_;
|
|
|
| // Buffer for storing samples received from the webrtc::AudioTransport.
|
| char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
|
|
|