Index: webrtc/modules/audio_device/fine_audio_buffer.h |
diff --git a/webrtc/modules/audio_device/fine_audio_buffer.h b/webrtc/modules/audio_device/fine_audio_buffer.h |
index 4ab5cd268ccb16176ce9a4e6db6932643633933c..478e0c6391f349b16bfffebcbafa6c901b363e23 100644 |
--- a/webrtc/modules/audio_device/fine_audio_buffer.h |
+++ b/webrtc/modules/audio_device/fine_audio_buffer.h |
@@ -11,7 +11,8 @@ |
#ifndef WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
#define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_ |
-#include "webrtc/base/scoped_ptr.h" |
+#include <memory> |
+ |
#include "webrtc/typedefs.h" |
namespace webrtc { |
@@ -86,14 +87,14 @@ class FineAudioBuffer { |
// Number of audio bytes per 10ms. |
const size_t bytes_per_10_ms_; |
// Storage for output samples that are not yet asked for. |
- rtc::scoped_ptr<int8_t[]> playout_cache_buffer_; |
+ std::unique_ptr<int8_t[]> playout_cache_buffer_; |
// Location of first unread output sample. |
size_t playout_cached_buffer_start_; |
// Number of bytes stored in output (contain samples to be played out) cache. |
size_t playout_cached_bytes_; |
// Storage for input samples that are about to be delivered to the WebRTC |
// ADB or remains from the last successful delivery of a 10ms audio buffer. |
- rtc::scoped_ptr<int8_t[]> record_cache_buffer_; |
+ std::unique_ptr<int8_t[]> record_cache_buffer_; |
// Required (max) size in bytes of the |record_cache_buffer_|. |
const size_t required_record_buffer_size_bytes_; |
// Number of bytes in input (contains recorded samples) cache. |