Move array_view.h to webrtc/api/

webrtc/modules/audio_processing/aec3/render_delay_controller.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_CONTROLLER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
 
+#include "webrtc/api/array_view.h"
 #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
 #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
-#include "webrtc/rtc_base/array_view.h"
 #include "webrtc/rtc_base/optional.h"
 
 namespace webrtc {
 
 // Class for aligning the render and capture signal using a RenderDelayBuffer.
 class RenderDelayController {
  public:
   static RenderDelayController* Create(
       const AudioProcessing::Config::EchoCanceller3& config,
       int sample_rate_hz);
   virtual ~RenderDelayController() = default;
 
   // Resets the delay controller.
   virtual void Reset() = 0;
 
   // Receives the externally used delay.
   virtual void SetDelay(size_t render_delay) = 0;
 
   // Aligns the render buffer content with the capture signal.
   virtual size_t GetDelay(const DownsampledRenderBuffer& render_buffer,
                           rtc::ArrayView<const float> capture) = 0;
 
   // Returns an approximate value for the headroom in the buffer alignment.
   virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_