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Side by Side Diff: webrtc/voice_engine/voice_engine.gyp

Issue 2221103004: Broke out 'level_indicator' of the voice_engine GN target. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added GN check, Gyp target, correct dependencies. Created 4 years, 4 months ago
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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
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
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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