Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

webrtc/modules/audio_device/ios/audio_device_ios.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #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"
 
 namespace webrtc {
 
 class FineAudioBuffer;
 
 // Implements full duplex 16-bit mono PCM audio support for iOS using a
 // Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
 // supports audio echo cancellation. It also adds automatic gain control,
@@ skipping 222 matching lines @@
   // audio data samples. Is also supports a similar scheme for the recording
   // side.
   // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
   // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
   // in each callback (one every 8ms). This class can then ask for 128 and the
   // FineAudioBuffer will ask WebRTC for new data only when needed and also
   // cache non-utilized audio between callbacks. On the recording side, iOS
   // 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.
   AudioBufferList audio_record_buffer_list_;
 
   // 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_;
 
   // Set to 1 when playout is active and 0 otherwise.
   volatile int playing_;
 
   // Set to true after successful call to Init(), false otherwise.
   bool initialized_;
 
   // Set to true after successful call to InitRecording(), false otherwise.
   bool rec_is_initialized_;
 
   // Set to true after successful call to InitPlayout(), false otherwise.
   bool play_is_initialized_;
 
   // Audio interruption observer instance.
   void* audio_interruption_observer_;
   void* route_change_observer_;
 
   // Contains the audio data format specification for a stream of audio.
   AudioStreamBasicDescription application_format_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_