Move webrtc/audio_*.h to webrtc/api/call

webrtc/media/media.gyp

 # Copyright (c) 2016 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 178 matching lines @@
   ],  # targets.
   'conditions': [
     ['include_tests==1', {
       'targets' : [
         {
           'target_name': 'rtc_unittest_main',
           'type': 'static_library',
           'dependencies': [
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/api/api.gyp:call_api',
             '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
           ],
           'direct_dependent_settings': {
             'include_dirs': [
               '<(libyuv_dir)/include',
               '<(DEPTH)/testing/gmock/include',
             ],
           },
           'conditions': [
             ['build_libyuv==1', {
@@ skipping 142 matching lines @@
               'sources': [
                 'rtc_media_unittests.isolate',
               ],
             },
           ],
         }],
       ],  # conditions
     }],  # include_tests==1
   ],  # conditions
 }