andrew@webrtc.org | a7b57da | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1 | # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 2 | # |
| 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 |
| 5 | # tree. An additional intellectual property rights grant can be found |
| 6 | # in the file PATENTS. All contributing project authors may |
| 7 | # be found in the AUTHORS file in the root of the source tree. |
| 8 | |
| 9 | { |
| 10 | 'targets': [ |
| 11 | { |
| 12 | 'target_name': 'voice_engine_core', |
| 13 | 'type': '<(library)', |
| 14 | 'dependencies': [ |
| 15 | '<(webrtc_root)/common_audio/common_audio.gyp:resampler', |
| 16 | '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing', |
| 17 | '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
| 18 | '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
| 19 | '<(webrtc_root)/modules/modules.gyp:audio_device', |
| 20 | '<(webrtc_root)/modules/modules.gyp:audio_processing', |
| 21 | '<(webrtc_root)/modules/modules.gyp:media_file', |
| 22 | '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
| 23 | '<(webrtc_root)/modules/modules.gyp:udp_transport', |
| 24 | '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
| 25 | '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', |
| 26 | ], |
| 27 | 'include_dirs': [ |
| 28 | 'include', |
| 29 | '<(webrtc_root)/modules/audio_device', |
| 30 | ], |
| 31 | 'direct_dependent_settings': { |
| 32 | 'include_dirs': [ |
| 33 | 'include', |
| 34 | ], |
| 35 | }, |
| 36 | 'sources': [ |
| 37 | '../common_types.h', |
| 38 | '../engine_configurations.h', |
| 39 | '../typedefs.h', |
| 40 | 'include/voe_audio_processing.h', |
| 41 | 'include/voe_base.h', |
| 42 | 'include/voe_call_report.h', |
| 43 | 'include/voe_codec.h', |
| 44 | 'include/voe_dtmf.h', |
| 45 | 'include/voe_encryption.h', |
| 46 | 'include/voe_errors.h', |
| 47 | 'include/voe_external_media.h', |
| 48 | 'include/voe_file.h', |
| 49 | 'include/voe_hardware.h', |
| 50 | 'include/voe_neteq_stats.h', |
| 51 | 'include/voe_network.h', |
| 52 | 'include/voe_rtp_rtcp.h', |
| 53 | 'include/voe_video_sync.h', |
| 54 | 'include/voe_volume_control.h', |
| 55 | 'channel.cc', |
| 56 | 'channel.h', |
| 57 | 'channel_manager.cc', |
| 58 | 'channel_manager.h', |
| 59 | 'channel_manager_base.cc', |
| 60 | 'channel_manager_base.h', |
| 61 | 'dtmf_inband.cc', |
| 62 | 'dtmf_inband.h', |
| 63 | 'dtmf_inband_queue.cc', |
| 64 | 'dtmf_inband_queue.h', |
| 65 | 'level_indicator.cc', |
| 66 | 'level_indicator.h', |
| 67 | 'monitor_module.cc', |
| 68 | 'monitor_module.h', |
| 69 | 'output_mixer.cc', |
| 70 | 'output_mixer.h', |
| 71 | 'output_mixer_internal.cc', |
| 72 | 'output_mixer_internal.h', |
| 73 | 'shared_data.cc', |
| 74 | 'shared_data.h', |
| 75 | 'statistics.cc', |
| 76 | 'statistics.h', |
| 77 | 'transmit_mixer.cc', |
| 78 | 'transmit_mixer.h', |
| 79 | 'utility.cc', |
| 80 | 'utility.h', |
| 81 | 'voe_audio_processing_impl.cc', |
| 82 | 'voe_audio_processing_impl.h', |
| 83 | 'voe_base_impl.cc', |
| 84 | 'voe_base_impl.h', |
| 85 | 'voe_call_report_impl.cc', |
| 86 | 'voe_call_report_impl.h', |
| 87 | 'voe_codec_impl.cc', |
| 88 | 'voe_codec_impl.h', |
| 89 | 'voe_dtmf_impl.cc', |
| 90 | 'voe_dtmf_impl.h', |
| 91 | 'voe_encryption_impl.cc', |
| 92 | 'voe_encryption_impl.h', |
| 93 | 'voe_external_media_impl.cc', |
| 94 | 'voe_external_media_impl.h', |
| 95 | 'voe_file_impl.cc', |
| 96 | 'voe_file_impl.h', |
| 97 | 'voe_hardware_impl.cc', |
| 98 | 'voe_hardware_impl.h', |
| 99 | 'voe_neteq_stats_impl.cc', |
| 100 | 'voe_neteq_stats_impl.h', |
| 101 | 'voe_network_impl.cc', |
| 102 | 'voe_network_impl.h', |
| 103 | 'voe_rtp_rtcp_impl.cc', |
| 104 | 'voe_rtp_rtcp_impl.h', |
| 105 | 'voe_video_sync_impl.cc', |
| 106 | 'voe_video_sync_impl.h', |
| 107 | 'voe_volume_control_impl.cc', |
| 108 | 'voe_volume_control_impl.h', |
| 109 | 'voice_engine_defines.h', |
| 110 | 'voice_engine_impl.cc', |
| 111 | 'voice_engine_impl.h', |
| 112 | ], |
| 113 | }, |
| 114 | ], |
| 115 | 'conditions': [ |
| 116 | ['OS=="win"', { |
| 117 | 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',], |
| 118 | }], |
| 119 | ['include_tests==1', { |
| 120 | 'targets': [ |
| 121 | { |
| 122 | 'target_name': 'voice_engine_unittests', |
| 123 | 'type': 'executable', |
| 124 | 'dependencies': [ |
| 125 | 'voice_engine_core', |
| 126 | '<(DEPTH)/testing/gtest.gyp:gtest', |
| 127 | '<(webrtc_root)/test/test.gyp:test_support_main', |
| 128 | # The rest are to satisfy the unittests' include chain. |
| 129 | # This would be unnecessary if we used qualified includes. |
| 130 | '<(webrtc_root)/common_audio/common_audio.gyp:resampler', |
| 131 | '<(webrtc_root)/modules/modules.gyp:audio_device', |
| 132 | '<(webrtc_root)/modules/modules.gyp:audio_processing', |
| 133 | '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
| 134 | '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
| 135 | '<(webrtc_root)/modules/modules.gyp:media_file', |
| 136 | '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
| 137 | '<(webrtc_root)/modules/modules.gyp:udp_transport', |
| 138 | '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
| 139 | '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers', |
| 140 | ], |
| 141 | 'include_dirs': [ |
| 142 | 'include', |
| 143 | ], |
| 144 | 'sources': [ |
| 145 | 'channel_unittest.cc', |
| 146 | 'output_mixer_unittest.cc', |
| 147 | 'transmit_mixer_unittest.cc', |
| 148 | 'voe_audio_processing_unittest.cc', |
| 149 | ], |
| 150 | }, |
| 151 | ], # targets |
| 152 | }], # include_tests |
| 153 | ], # conditions |
| 154 | } |