Implement AudioReceiveStream::GetStats().

webrtc/test/webrtc_test_common.gyp

 # Copyright (c) 2013 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 12 matching lines @@
         'encoder_settings.cc',
         'encoder_settings.h',
         'fake_audio_device.cc',
         'fake_audio_device.h',
         'fake_decoder.cc',
         'fake_decoder.h',
         'fake_encoder.cc',
         'fake_encoder.h',
         'fake_network_pipe.cc',
         'fake_network_pipe.h',
+        'fake_voice_engine.h',
         'frame_generator_capturer.cc',
         'frame_generator_capturer.h',
         'layer_filtering_transport.cc',
         'layer_filtering_transport.h',
         'mock_transport.h',
         'null_transport.cc',
         'null_transport.h',
         'random.cc',
         'random.h',
         'rtp_rtcp_observer.h',
@@ skipping 120 matching lines @@
             'fake_network_pipe_unittest.cc',
             'frame_generator_unittest.cc',
             'rtp_file_reader_unittest.cc',
             'rtp_file_writer_unittest.cc',
           ],
         },
       ],  #targets
     }],  # include_tests
   ],  # conditions
 }