| Index: webrtc/common_audio/audio_ring_buffer_unittest.cc
|
| diff --git a/webrtc/common_audio/audio_ring_buffer_unittest.cc b/webrtc/common_audio/audio_ring_buffer_unittest.cc
|
| index a7a6a9442bca8d0f83641b7652b71a58a83e6950..c5c38de56db4f82f3868ee2a54b09b57980f94bb 100644
|
| --- a/webrtc/common_audio/audio_ring_buffer_unittest.cc
|
| +++ b/webrtc/common_audio/audio_ring_buffer_unittest.cc
|
| @@ -8,6 +8,8 @@
|
| * be found in the AUTHORS file in the root of the source tree.
|
| */
|
|
|
| +#include <memory>
|
| +
|
| #include "webrtc/common_audio/audio_ring_buffer.h"
|
|
|
| #include "testing/gtest/include/gtest/gtest.h"
|
| @@ -27,7 +29,7 @@ void ReadAndWriteTest(const ChannelBuffer<float>& input,
|
| const size_t num_channels = input.num_channels();
|
| const size_t total_frames = input.num_frames();
|
| AudioRingBuffer buf(num_channels, buffer_frames);
|
| - rtc::scoped_ptr<float* []> slice(new float* [num_channels]);
|
| + std::unique_ptr<float* []> slice(new float*[num_channels]);
|
|
|
| size_t input_pos = 0;
|
| size_t output_pos = 0;
|
|
|