Adds Analog Gain Control (AGC) to the WebRTC client.

media/media.gyp

 # Copyright (c) 2012 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 {
   'variables': {
     'chromium_code': 1,
     # Override to dynamically link the PulseAudio library.
     'use_pulseaudio%': 0,
   },
@@ skipping 15 matching lines @@
       ],
       'include_dirs': [
         '..',
       ],
       'sources': [
         'audio/audio_buffers_state.cc',
         'audio/audio_buffers_state.h',
         'audio/audio_io.h',
         'audio/audio_input_controller.cc',
         'audio/audio_input_controller.h',
+        'audio/audio_input_stream_impl.cc',
+        'audio/audio_input_stream_impl.h',
         'audio/audio_device_name.cc',
         'audio/audio_device_name.h',
         'audio/audio_manager.cc',
         'audio/audio_manager.h',
         'audio/audio_manager_base.cc',
         'audio/audio_manager_base.h',
         'audio/audio_output_controller.cc',
         'audio/audio_output_controller.h',
         'audio/audio_output_dispatcher.cc',
         'audio/audio_output_dispatcher.h',
@@ skipping 902 matching lines @@
             '../third_party/ffmpeg/ffmpeg.gyp:ffmpeg',
           ],
           'sources': [
             'tools/media_bench/media_bench.cc',
           ],
         },
       ],
     }]
   ],
 }