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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': [ |
| (...skipping 11 matching lines...) Expand all Loading... |
| 22 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 22 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
| 23 '<(webrtc_root)/modules/modules.gyp:audio_device', | 23 '<(webrtc_root)/modules/modules.gyp:audio_device', |
| 24 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 24 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
| 25 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', | 25 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', |
| 26 '<(webrtc_root)/modules/modules.gyp:media_file', | 26 '<(webrtc_root)/modules/modules.gyp:media_file', |
| 27 '<(webrtc_root)/modules/modules.gyp:paced_sender', | 27 '<(webrtc_root)/modules/modules.gyp:paced_sender', |
| 28 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 28 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
| 29 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 29 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
| 30 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | 30 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', |
| 31 '<(webrtc_root)/webrtc.gyp:rtc_event_log', | 31 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
| 32 'level_indicator', |
| 32 ], | 33 ], |
| 33 'export_dependent_settings': [ | 34 'export_dependent_settings': [ |
| 34 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 35 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
| 35 ], | 36 ], |
| 36 'sources': [ | 37 'sources': [ |
| 37 'include/voe_audio_processing.h', | 38 'include/voe_audio_processing.h', |
| 38 'include/voe_base.h', | 39 'include/voe_base.h', |
| 39 'include/voe_codec.h', | 40 'include/voe_codec.h', |
| 40 'include/voe_errors.h', | 41 'include/voe_errors.h', |
| 41 'include/voe_external_media.h', | 42 'include/voe_external_media.h', |
| 42 'include/voe_file.h', | 43 'include/voe_file.h', |
| 43 'include/voe_hardware.h', | 44 'include/voe_hardware.h', |
| 44 'include/voe_neteq_stats.h', | 45 'include/voe_neteq_stats.h', |
| 45 'include/voe_network.h', | 46 'include/voe_network.h', |
| 46 'include/voe_rtp_rtcp.h', | 47 'include/voe_rtp_rtcp.h', |
| 47 'include/voe_video_sync.h', | 48 'include/voe_video_sync.h', |
| 48 'include/voe_volume_control.h', | 49 'include/voe_volume_control.h', |
| 49 'channel.cc', | 50 'channel.cc', |
| 50 'channel.h', | 51 'channel.h', |
| 51 'channel_manager.cc', | 52 'channel_manager.cc', |
| 52 'channel_manager.h', | 53 'channel_manager.h', |
| 53 'channel_proxy.cc', | 54 'channel_proxy.cc', |
| 54 'channel_proxy.h', | 55 'channel_proxy.h', |
| 55 'level_indicator.cc', | |
| 56 'level_indicator.h', | |
| 57 'monitor_module.cc', | 56 'monitor_module.cc', |
| 58 'monitor_module.h', | 57 'monitor_module.h', |
| 59 'network_predictor.cc', | 58 'network_predictor.cc', |
| 60 'network_predictor.h', | 59 'network_predictor.h', |
| 61 'output_mixer.cc', | 60 'output_mixer.cc', |
| 62 'output_mixer.h', | 61 'output_mixer.h', |
| 63 'shared_data.cc', | 62 'shared_data.cc', |
| 64 'shared_data.h', | 63 'shared_data.h', |
| 65 'statistics.cc', | 64 'statistics.cc', |
| 66 'statistics.h', | 65 'statistics.h', |
| (...skipping 21 matching lines...) Expand all Loading... |
| 88 'voe_rtp_rtcp_impl.h', | 87 'voe_rtp_rtcp_impl.h', |
| 89 'voe_video_sync_impl.cc', | 88 'voe_video_sync_impl.cc', |
| 90 'voe_video_sync_impl.h', | 89 'voe_video_sync_impl.h', |
| 91 'voe_volume_control_impl.cc', | 90 'voe_volume_control_impl.cc', |
| 92 'voe_volume_control_impl.h', | 91 'voe_volume_control_impl.h', |
| 93 'voice_engine_defines.h', | 92 'voice_engine_defines.h', |
| 94 'voice_engine_impl.cc', | 93 'voice_engine_impl.cc', |
| 95 'voice_engine_impl.h', | 94 'voice_engine_impl.h', |
| 96 ], | 95 ], |
| 97 }, | 96 }, |
| 97 { |
| 98 'target_name': 'level_indicator', |
| 99 'type': 'static_library', |
| 100 'dependencies': [ |
| 101 '<(webrtc_root)/base/base.gyp:rtc_base_approved', |
| 102 '<(webrtc_root)/common.gyp:webrtc_common', |
| 103 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
| 104 ], |
| 105 'sources': [ |
| 106 'level_indicator.cc', |
| 107 'level_indicator.h', |
| 108 ] |
| 109 } |
| 98 ], | 110 ], |
| 99 'conditions': [ | 111 'conditions': [ |
| 100 ['OS=="win"', { | 112 ['OS=="win"', { |
| 101 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',], | 113 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',], |
| 102 }], | 114 }], |
| 103 ['include_tests==1', { | 115 ['include_tests==1', { |
| 104 'targets': [ | 116 'targets': [ |
| 105 { | 117 { |
| 106 'target_name': 'voice_engine_unittests', | 118 'target_name': 'voice_engine_unittests', |
| 107 'type': '<(gtest_target_type)', | 119 'type': '<(gtest_target_type)', |
| (...skipping 196 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 304 'sources': [ | 316 'sources': [ |
| 305 'voe_auto_test.isolate', | 317 'voe_auto_test.isolate', |
| 306 ], | 318 ], |
| 307 }, | 319 }, |
| 308 ], | 320 ], |
| 309 }], | 321 }], |
| 310 ], # conditions | 322 ], # conditions |
| 311 }], # include_tests==1 | 323 }], # include_tests==1 |
| 312 ], # conditions | 324 ], # conditions |
| 313 } | 325 } |
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