Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

Index: webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
diff --git a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
index 6666364c9e38180d9ce95e1d9972a4e3e4738cc6..ef189d1fef72a79f3cd8e16b1d493bac934c4ba5 100644
--- a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
+++ b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -15,7 +15,6 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
 
 using ::testing::_;
@@ -118,9 +117,9 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
   FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
                               sample_rate);
 
-  rtc::scoped_ptr<int8_t[]> out_buffer;
+  std::unique_ptr<int8_t[]> out_buffer;
   out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
-  rtc::scoped_ptr<int8_t[]> in_buffer;
+  std::unique_ptr<int8_t[]> in_buffer;
   in_buffer.reset(new int8_t[kFrameSizeBytes]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
     fine_buffer.GetPlayoutData(out_buffer.get());