| Index: webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
|
| index 7a0bb1a6afce52a56344941634d0496bafe40a93..f5fe16691e326eaf8adf32af3655ce7c1847e99e 100644
|
| --- a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
|
| @@ -10,8 +10,9 @@
|
|
|
| #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
|
|
|
| +#include <memory>
|
| +
|
| #include "webrtc/base/checks.h"
|
| -#include "webrtc/base/scoped_ptr.h"
|
|
|
| namespace webrtc {
|
| namespace test {
|
| @@ -22,7 +23,7 @@ bool ResampleInputAudioFile::Read(size_t samples,
|
| const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
|
| RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
|
| << "Frame size and sample rates don't add up to an integer.";
|
| - rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
|
| + std::unique_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
|
| if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
|
| return false;
|
| resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
|
|
|