Index: webrtc/modules/audio_device/ios/audio_device_ios.h |
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.h b/webrtc/modules/audio_device/ios/audio_device_ios.h |
index c4eb0d6f6451a5e8b8efa8da3924eb0236cc26bd..73208864d245a71d157138179004ed3dc6d275a3 100644 |
--- a/webrtc/modules/audio_device/ios/audio_device_ios.h |
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.h |
@@ -11,9 +11,10 @@ |
#ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ |
#define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ |
+#include <memory> |
+ |
#include <AudioUnit/AudioUnit.h> |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/base/thread_checker.h" |
#include "webrtc/modules/audio_device/audio_device_generic.h" |
@@ -256,11 +257,11 @@ class AudioDeviceIOS : public AudioDeviceGeneric { |
// can provide audio data frames of size 128 and these are accumulated until |
// enough data to supply one 10ms call exists. This 10ms chunk is then sent |
// to WebRTC and the remaining part is stored. |
- rtc::scoped_ptr<FineAudioBuffer> fine_audio_buffer_; |
+ std::unique_ptr<FineAudioBuffer> fine_audio_buffer_; |
// Extra audio buffer to be used by the playout side for rendering audio. |
// The buffer size is given by FineAudioBuffer::RequiredBufferSizeBytes(). |
- rtc::scoped_ptr<SInt8[]> playout_audio_buffer_; |
+ std::unique_ptr<SInt8[]> playout_audio_buffer_; |
// Provides a mechanism for encapsulating one or more buffers of audio data. |
// Only used on the recording side. |
@@ -268,7 +269,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric { |
// Temporary storage for recorded data. AudioUnitRender() renders into this |
// array as soon as a frame of the desired buffer size has been recorded. |
- rtc::scoped_ptr<SInt8[]> record_audio_buffer_; |
+ std::unique_ptr<SInt8[]> record_audio_buffer_; |
// Set to 1 when recording is active and 0 otherwise. |
volatile int recording_; |