Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

webrtc/modules/audio_device/android/opensles_player.h

 /*
  *  Copyright (c) 2015 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_ANDROID_OPENSLES_PLAYER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
 
+#include <memory>
+
 #include <SLES/OpenSLES.h>
 #include <SLES/OpenSLES_Android.h>
 #include <SLES/OpenSLES_AndroidConfiguration.h>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/opensles_common.h"
 #include "webrtc/modules/audio_device/include/audio_device_defines.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/utility/include/helpers_android.h"
 
 namespace webrtc {
 
@@ skipping 114 matching lines @@
   SLDataFormat_PCM pcm_format_;
 
   // Number of bytes per audio buffer in each |audio_buffers_[i]|.
   // Typical sizes are 480 or 512 bytes corresponding to native output buffer
   // sizes of 240 or 256 audio frames respectively.
   size_t bytes_per_buffer_;
 
   // Queue of audio buffers to be used by the player object for rendering
   // audio. They will be used in a Round-robin way and the size of each buffer
   // is given by FineAudioBuffer::RequiredBufferSizeBytes().
-  rtc::scoped_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
+  std::unique_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
 
   // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
   // in chunks of 10ms. It then allows for this data to be pulled in
   // a finer or coarser granularity. I.e. interacting with this class instead
   // of directly with the AudioDeviceBuffer one can ask for any number of
   // audio data samples.
   // Example: native buffer size is 240 audio frames at 48kHz sample rate.
   // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 240
   // in each callback (one every 5ms). This class can then ask for 240 and the
   // FineAudioBuffer will ask WebRTC for new data only every second callback
   // and also cach non-utilized audio.
-  rtc::scoped_ptr<FineAudioBuffer> fine_buffer_;
+  std::unique_ptr<FineAudioBuffer> fine_buffer_;
 
   // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
   // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
   int buffer_index_;
 
   // The engine object which provides the SLEngineItf interface.
   // Created by the global Open SL ES constructor slCreateEngine().
   webrtc::ScopedSLObjectItf engine_object_;
 
   // This interface exposes creation methods for all the OpenSL ES object types.
@@ skipping 20 matching lines @@
   // properties. This interface is supported on the Audio Player object.
   SLVolumeItf volume_;
 
   // Last time the OpenSL ES layer asked for audio data to play out.
   uint32_t last_play_time_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_