Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

webrtc/modules/audio_device/android/opensles_player.h

Index: webrtc/modules/audio_device/android/opensles_player.h
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h
index fa9e931218cebfd0332404198d4a82189434d0aa..084546dbf7b3ab2d5fc2a32816d5441aa6388d96 100644
--- a/webrtc/modules/audio_device/android/opensles_player.h
+++ b/webrtc/modules/audio_device/android/opensles_player.h
@@ -11,11 +11,12 @@
 #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"
@@ -150,7 +151,7 @@ class OpenSLESPlayer {
   // 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
@@ -162,7 +163,7 @@ class OpenSLESPlayer {
   // 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, ...