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); |