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()); |