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| 1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| 2 # | 2 # |
| 3 # Use of this source code is governed by a BSD-style license | 3 # Use of this source code is governed by a BSD-style license |
| 4 # that can be found in the LICENSE file in the root of the source | 4 # that can be found in the LICENSE file in the root of the source |
| 5 # tree. An additional intellectual property rights grant can be found | 5 # tree. An additional intellectual property rights grant can be found |
| 6 # in the file PATENTS. All contributing project authors may | 6 # in the file PATENTS. All contributing project authors may |
| 7 # be found in the AUTHORS file in the root of the source tree. | 7 # be found in the AUTHORS file in the root of the source tree. |
| 8 | 8 |
| 9 { | 9 { |
| 10 'includes': [ | 10 'includes': [ |
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| 45 'include/voe_network.h', | 45 'include/voe_network.h', |
| 46 'include/voe_rtp_rtcp.h', | 46 'include/voe_rtp_rtcp.h', |
| 47 'include/voe_video_sync.h', | 47 'include/voe_video_sync.h', |
| 48 'include/voe_volume_control.h', | 48 'include/voe_volume_control.h', |
| 49 'channel.cc', | 49 'channel.cc', |
| 50 'channel.h', | 50 'channel.h', |
| 51 'channel_manager.cc', | 51 'channel_manager.cc', |
| 52 'channel_manager.h', | 52 'channel_manager.h', |
| 53 'channel_proxy.cc', | 53 'channel_proxy.cc', |
| 54 'channel_proxy.h', | 54 'channel_proxy.h', |
| 55 'coder.cc', | |
| 56 'coder.h', | |
| 57 'file_player.h', | |
| 58 'file_player_impl.cc', | |
| 59 'file_player_impl.h', | |
| 60 'file_recorder.h', | |
| 61 'file_recorder_impl.cc', | |
| 62 'file_recorder_impl.h', | |
| 63 'level_indicator.cc', | 55 'level_indicator.cc', |
| 64 'level_indicator.h', | 56 'level_indicator.h', |
| 65 'monitor_module.cc', | 57 'monitor_module.cc', |
| 66 'monitor_module.h', | 58 'monitor_module.h', |
| 67 'network_predictor.cc', | 59 'network_predictor.cc', |
| 68 'network_predictor.h', | 60 'network_predictor.h', |
| 69 'output_mixer.cc', | 61 'output_mixer.cc', |
| 70 'output_mixer.h', | 62 'output_mixer.h', |
| 71 'shared_data.cc', | 63 'shared_data.cc', |
| 72 'shared_data.h', | 64 'shared_data.h', |
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| 110 }], | 102 }], |
| 111 ['include_tests==1', { | 103 ['include_tests==1', { |
| 112 'targets': [ | 104 'targets': [ |
| 113 { | 105 { |
| 114 'target_name': 'voice_engine_unittests', | 106 'target_name': 'voice_engine_unittests', |
| 115 'type': '<(gtest_target_type)', | 107 'type': '<(gtest_target_type)', |
| 116 'dependencies': [ | 108 'dependencies': [ |
| 117 'voice_engine', | 109 'voice_engine', |
| 118 '<(DEPTH)/testing/gmock.gyp:gmock', | 110 '<(DEPTH)/testing/gmock.gyp:gmock', |
| 119 '<(DEPTH)/testing/gtest.gyp:gtest', | 111 '<(DEPTH)/testing/gtest.gyp:gtest', |
| 120 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', | |
| 121 # The rest are to satisfy the unittests' include chain. | 112 # The rest are to satisfy the unittests' include chain. |
| 122 # This would be unnecessary if we used qualified includes. | 113 # This would be unnecessary if we used qualified includes. |
| 123 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | 114 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
| 124 '<(webrtc_root)/modules/modules.gyp:audio_device', | 115 '<(webrtc_root)/modules/modules.gyp:audio_device', |
| 125 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 116 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
| 126 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 117 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
| 127 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 118 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
| 128 '<(webrtc_root)/modules/modules.gyp:media_file', | 119 '<(webrtc_root)/modules/modules.gyp:media_file', |
| 129 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 120 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
| 130 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 121 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
| 131 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, | 122 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, |
| 132 '<(webrtc_root)/test/test.gyp:test_support_main', | 123 '<(webrtc_root)/test/test.gyp:test_support_main', |
| 133 ], | 124 ], |
| 134 'sources': [ | 125 'sources': [ |
| 135 'channel_unittest.cc', | 126 'channel_unittest.cc', |
| 136 'file_player_unittests.cc', | |
| 137 'network_predictor_unittest.cc', | 127 'network_predictor_unittest.cc', |
| 138 'transmit_mixer_unittest.cc', | 128 'transmit_mixer_unittest.cc', |
| 139 'utility_unittest.cc', | 129 'utility_unittest.cc', |
| 140 'voe_audio_processing_unittest.cc', | 130 'voe_audio_processing_unittest.cc', |
| 141 'voe_base_unittest.cc', | 131 'voe_base_unittest.cc', |
| 142 'voe_codec_unittest.cc', | 132 'voe_codec_unittest.cc', |
| 143 'voe_network_unittest.cc', | 133 'voe_network_unittest.cc', |
| 144 'voice_engine_fixture.cc', | 134 'voice_engine_fixture.cc', |
| 145 'voice_engine_fixture.h', | 135 'voice_engine_fixture.h', |
| 146 ], | 136 ], |
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| 314 'sources': [ | 304 'sources': [ |
| 315 'voe_auto_test.isolate', | 305 'voe_auto_test.isolate', |
| 316 ], | 306 ], |
| 317 }, | 307 }, |
| 318 ], | 308 ], |
| 319 }], | 309 }], |
| 320 ], # conditions | 310 ], # conditions |
| 321 }], # include_tests==1 | 311 }], # include_tests==1 |
| 322 ], # conditions | 312 ], # conditions |
| 323 } | 313 } |
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