Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.

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.
 {
   'variables': {
     'webrtc_all_dependencies': [
@@ skipping 92 matching lines @@
             'test/webrtc_test_common.gyp:*',
             'webrtc_tests',
           ],
         }],
       ],
     },
     {
       'target_name': 'webrtc',
       'type': 'static_library',
       'sources': [
-        'audio_receive_stream.h',
         'audio_send_stream.h',
         'audio_state.h',
         'call.h',
-        'config.h',
-        'frame_callback.h',
-        'stream.h',
-        'transport.h',
-        'video_receive_stream.h',
-        'video_renderer.h',
-        'video_send_stream.h',
-
+        'video_frame.h',
+        'video_decoder.h',
+        'video_encoder.h',
         '<@(webrtc_audio_sources)',
         '<@(webrtc_call_sources)',
         '<@(webrtc_video_sources)',
       ],
       'dependencies': [
         'common.gyp:*',
         '<@(webrtc_audio_dependencies)',
         '<@(webrtc_call_dependencies)',
         '<@(webrtc_video_dependencies)',
         'rtc_event_log',
@@ skipping 25 matching lines @@
           ],
           'defines': [
             'ENABLE_RTC_EVENT_LOG',
           ],
         }],
       ],
     },
 
   ],
 }