Move array_view.h to webrtc/api/

webrtc/modules/audio_processing/test/conversational_speech/timing.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_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
 
 #include <string>
 #include <vector>
 
-#include "webrtc/rtc_base/array_view.h"
+#include "webrtc/api/array_view.h"
 
 namespace webrtc {
 namespace test {
 namespace conversational_speech {
 
 struct Turn{
   Turn(std::string new_speaker_name, std::string new_audiotrack_file_name,
        int new_offset)
       : speaker_name(new_speaker_name),
         audiotrack_file_name(new_audiotrack_file_name),
         offset(new_offset) {}
   bool operator==(const Turn &b) const;
   std::string speaker_name;
   std::string audiotrack_file_name;
   int offset;
 };
 
 // Loads a list of turns from a file.
 std::vector<Turn> LoadTiming(const std::string& timing_filepath);
 
 // Writes a list of turns into a file.
 void SaveTiming(const std::string& timing_filepath,
                 rtc::ArrayView<const Turn> timing);
 
 }  // namespace conversational_speech
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_