Index: webrtc/common_audio/resampler/include/push_resampler.h |
diff --git a/webrtc/common_audio/resampler/include/push_resampler.h b/webrtc/common_audio/resampler/include/push_resampler.h |
index 31e08e3640fefbc18118587b7d07dcc4143473f8..eeda79049707e00600a0ad1d9361834e2f6f738d 100644 |
--- a/webrtc/common_audio/resampler/include/push_resampler.h |
+++ b/webrtc/common_audio/resampler/include/push_resampler.h |
@@ -11,8 +11,7 @@ |
#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
-#include <memory> |
- |
+#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/typedefs.h" |
namespace webrtc { |
@@ -37,15 +36,15 @@ |
int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity); |
private: |
- std::unique_ptr<PushSincResampler> sinc_resampler_; |
- std::unique_ptr<PushSincResampler> sinc_resampler_right_; |
+ rtc::scoped_ptr<PushSincResampler> sinc_resampler_; |
+ rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_; |
int src_sample_rate_hz_; |
int dst_sample_rate_hz_; |
size_t num_channels_; |
- std::unique_ptr<T[]> src_left_; |
- std::unique_ptr<T[]> src_right_; |
- std::unique_ptr<T[]> dst_left_; |
- std::unique_ptr<T[]> dst_right_; |
+ rtc::scoped_ptr<T[]> src_left_; |
+ rtc::scoped_ptr<T[]> src_right_; |
+ rtc::scoped_ptr<T[]> dst_left_; |
+ rtc::scoped_ptr<T[]> dst_right_; |
}; |
} // namespace webrtc |