Move FilePlayer and FileRecorder to Voice Engine

webrtc/voice_engine/voice_engine.gyp

 # Copyright (c) 2011 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': [
@@ skipping 34 matching lines @@
         'include/voe_network.h',
         'include/voe_rtp_rtcp.h',
         'include/voe_video_sync.h',
         'include/voe_volume_control.h',
         'channel.cc',
         'channel.h',
         'channel_manager.cc',
         'channel_manager.h',
         'channel_proxy.cc',
         'channel_proxy.h',
+        'coder.cc',
+        'coder.h',
+        'file_player.h',
+        'file_player_impl.cc',
+        'file_player_impl.h',
+        'file_recorder.h',
+        'file_recorder_impl.cc',
+        'file_recorder_impl.h',
         'level_indicator.cc',
         'level_indicator.h',
         'monitor_module.cc',
         'monitor_module.h',
         'network_predictor.cc',
         'network_predictor.h',
         'output_mixer.cc',
         'output_mixer.h',
         'shared_data.cc',
         'shared_data.h',
@@ skipping 37 matching lines @@
     }],
     ['include_tests==1', {
       'targets': [
         {
           'target_name': 'voice_engine_unittests',
           'type': '<(gtest_target_type)',
           'dependencies': [
             'voice_engine',
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             # The rest are to satisfy the unittests' include chain.
             # This would be unnecessary if we used qualified includes.
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(webrtc_root)/modules/modules.gyp:audio_device',
             '<(webrtc_root)/modules/modules.gyp:audio_processing',
             '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
             '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
             '<(webrtc_root)/modules/modules.gyp:media_file',
             '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
             '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'sources': [
             'channel_unittest.cc',
+            'file_player_unittests.cc',
             'network_predictor_unittest.cc',
             'transmit_mixer_unittest.cc',
             'utility_unittest.cc',
             'voe_audio_processing_unittest.cc',
             'voe_base_unittest.cc',
             'voe_codec_unittest.cc',
             'voe_network_unittest.cc',
             'voice_engine_fixture.cc',
             'voice_engine_fixture.h',
           ],
@@ skipping 167 matching lines @@
               'sources': [
                 'voe_auto_test.isolate',
               ],
             },
           ],
         }],
       ],  # conditions
     }], # include_tests==1
   ], # conditions
 }