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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': [ |
| 11 '../build/common.gypi', | 11 '../build/common.gypi', |
| 12 ], | 12 ], |
| 13 'targets': [ | 13 'targets': [ |
| 14 { | 14 { |
| 15 'target_name': 'voice_engine', | 15 'target_name': 'voice_engine', |
| 16 'type': 'static_library', | 16 'type': 'static_library', |
| 17 'dependencies': [ | 17 'dependencies': [ |
| 18 '<(webrtc_root)/common.gyp:webrtc_common', | 18 '<(webrtc_root)/common.gyp:webrtc_common', |
| 19 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | 19 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
| 20 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 20 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
| 21 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 21 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
| 22 '<(webrtc_root)/modules/modules.gyp:audio_device', | 22 '<(webrtc_root)/modules/modules.gyp:audio_device', |
| 23 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 23 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
| 24 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', | 24 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', |
| 25 '<(webrtc_root)/modules/modules.gyp:media_file', | 25 '<(webrtc_root)/modules/modules.gyp:media_file', |
| 26 '<(webrtc_root)/modules/modules.gyp:paced_sender', |
| 26 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 27 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
| 27 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 28 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
| 28 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | 29 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', |
| 29 '<(webrtc_root)/webrtc.gyp:rtc_event_log', | 30 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
| 30 ], | 31 ], |
| 31 'export_dependent_settings': [ | 32 'export_dependent_settings': [ |
| 32 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 33 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
| 33 ], | 34 ], |
| 34 'sources': [ | 35 'sources': [ |
| 35 'include/voe_audio_processing.h', | 36 'include/voe_audio_processing.h', |
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| 288 'sources': [ | 289 'sources': [ |
| 289 'voe_auto_test.isolate', | 290 'voe_auto_test.isolate', |
| 290 ], | 291 ], |
| 291 }, | 292 }, |
| 292 ], | 293 ], |
| 293 }], | 294 }], |
| 294 ], # conditions | 295 ], # conditions |
| 295 }], # include_tests==1 | 296 }], # include_tests==1 |
| 296 ], # conditions | 297 ], # conditions |
| 297 } | 298 } |
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