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]; |