Move array_view.h to webrtc/api/

webrtc/modules/audio_device/fine_audio_buffer.h

 /*
  *  Copyright (c) 2013 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_FINE_AUDIO_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 
 #include <memory>
 
-#include "webrtc/rtc_base/array_view.h"
+#include "webrtc/api/array_view.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioDeviceBuffer;
 
 // FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data
 // corresponding to 10ms of data. It then allows for this data to be pulled in
 // a finer or coarser granularity. I.e. interacting with this class instead of
@@ skipping 55 matching lines @@
   // in any size using GetPlayoutData().
   rtc::BufferT<int8_t> playout_buffer_;
   // 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::BufferT<int8_t> record_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_