modules: more interface -> include renames

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  Copyright (c) 2012 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.
  */
 
 #if !defined(__has_feature) || !__has_feature(objc_arc)
 #error "This file requires ARC support."
 #endif
 
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
 
 #include "webrtc/base/atomicops.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
 
 namespace webrtc {
 
 #define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
 
 #define LOG_AND_RETURN_IF_ERROR(error, message) \
   do {                                          \
     OSStatus err = error;                       \
     if (err) {                                  \
       LOG(LS_ERROR) << message << ": " << err;  \
@@ skipping 882 matching lines @@
   // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
   // the native I/O audio unit) to a preallocated intermediate buffer and
   // copy the result to the audio buffer in the |io_data| destination.
   SInt8* source = playout_audio_buffer_.get();
   fine_audio_buffer_->GetPlayoutData(source);
   memcpy(destination, source, dataSizeInBytes);
   return noErr;
 }
 
 }  // namespace webrtc