Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base.

webrtc/base/base.gyp

 # Copyright (c) 2014 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.
 
 {
   'includes': [ '../build/common.gypi', ],
@@ skipping 11 matching lines @@
         }],
       ],
     }],
   ],
   'targets': [
     {
       # The subset of rtc_base approved for use outside of libjingle.
       'target_name': 'rtc_base_approved',
       'type': 'static_library',
       'sources': [
+        'arraysize.h',
         'array_view.h',
         'atomicops.h',
         'bind.h',
         'bitbuffer.cc',
         'bitbuffer.h',
         'buffer.h',
         'bufferqueue.cc',
         'bufferqueue.h',
         'bytebuffer.cc',
         'bytebuffer.h',
@@ skipping 182 matching lines @@
         'rtc_base_approved',
       ],
       'defines': [
         'FEATURE_ENABLE_SSL',
         'SSL_USE_OPENSSL',
         'HAVE_OPENSSL_SSL_H',
         'LOGGING=1',
       ],
       'sources': [
         'applefilesystem.mm',
-        'arraysize.h',
         'asyncfile.cc',
         'asyncfile.h',
         'asyncinvoker.cc',
         'asyncinvoker.h',
         'asyncinvoker-inl.h',
         'asyncpacketsocket.cc',
         'asyncpacketsocket.h',
         'asyncresolverinterface.cc',
         'asyncresolverinterface.h',
         'asyncsocket.cc',
@@ skipping 424 matching lines @@
     },
     {
      'target_name': 'gtest_prod',
       'type': 'static_library',
       'sources': [
         'gtest_prod_util.h',
       ],
     },
   ],
 }