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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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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', |
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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)', |
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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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