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

webrtc/webrtc.gyp

 # 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.
 {
   'includes': [
     'build/common.gypi',
     'audio/webrtc_audio.gypi',
     'call/webrtc_call.gypi',
     'video/webrtc_video.gypi',
   ],
   'targets': [
     {
       'target_name': 'webrtc',
       'type': 'static_library',
       'sources': [
-        'audio_receive_stream.h',
-        'audio_send_stream.h',
-        'audio_state.h',
         'call.h',
         'config.h',
         'transport.h',
         'video_receive_stream.h',
         'video_send_stream.h',
 
         '<@(webrtc_audio_sources)',
         '<@(webrtc_call_sources)',
         '<@(webrtc_video_sources)',
       ],
@@ skipping 82 matching lines @@
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             'rtc_event_log_parser',
             'rtc_event_log_proto',
             'test/test.gyp:rtp_test_utils'
           ],
         },
       ],
     }],
   ],  # conditions
 }