Move array_view.h to webrtc/api/

webrtc/modules/audio_processing/aec3/render_delay_buffer.h

 /*
  *  Copyright (c) 2017 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_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
 
 #include <stddef.h>
 #include <array>
 #include <vector>
 
+#include "webrtc/api/array_view.h"
 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
 #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/fft_data.h"
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
-#include "webrtc/rtc_base/array_view.h"
 
 namespace webrtc {
 
 // Class for buffering the incoming render blocks such that these may be
 // extracted with a specified delay.
 class RenderDelayBuffer {
  public:
   static RenderDelayBuffer* Create(size_t num_bands);
   virtual ~RenderDelayBuffer() = default;
 
@@ skipping 17 matching lines @@
   // Returns the render buffer for the echo remover.
   virtual const RenderBuffer& GetRenderBuffer() const = 0;
 
   // Returns the downsampled render buffer.
   virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_