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1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2016 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': [ '../build/common.gypi', ], | 10 'includes': [ '../build/common.gypi', ], |
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127 }], | 127 }], |
128 ['build_with_chromium==1', { | 128 ['build_with_chromium==1', { |
129 'dependencies': [ | 129 'dependencies': [ |
130 '<(webrtc_root)/modules/modules.gyp:video_capture', | 130 '<(webrtc_root)/modules/modules.gyp:video_capture', |
131 ], | 131 ], |
132 }, { | 132 }, { |
133 'defines': [ | 133 'defines': [ |
134 'HAVE_WEBRTC_VIDEO', | 134 'HAVE_WEBRTC_VIDEO', |
135 'HAVE_WEBRTC_VOICE', | 135 'HAVE_WEBRTC_VOICE', |
136 ], | 136 ], |
137 'direct_dependent_settings': { | 137 'all_dependent_settings': { |
138 'defines': [ | 138 'defines': [ |
139 'HAVE_WEBRTC_VIDEO', | 139 'HAVE_WEBRTC_VIDEO', |
140 'HAVE_WEBRTC_VOICE', | 140 'HAVE_WEBRTC_VOICE', |
141 ], | 141 ], |
142 }, | 142 }, |
143 'dependencies': [ | 143 'dependencies': [ |
144 '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_im pl', | 144 '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_im pl', |
145 ], | 145 ], |
146 }], | 146 }], |
147 ['OS=="linux" and use_gtk==1', { | 147 ['OS=="linux" and use_gtk==1', { |
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247 'engine/fakewebrtcvoiceengine.h', | 247 'engine/fakewebrtcvoiceengine.h', |
248 ], | 248 ], |
249 }, # target rtc_unittest_main | 249 }, # target rtc_unittest_main |
250 { | 250 { |
251 'target_name': 'rtc_media_unittests', | 251 'target_name': 'rtc_media_unittests', |
252 'type': 'executable', | 252 'type': 'executable', |
253 'dependencies': [ | 253 'dependencies': [ |
254 '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', | 254 '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', |
255 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default' , | 255 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default' , |
256 '<(webrtc_root)/test/test.gyp:test_support', | 256 '<(webrtc_root)/test/test.gyp:test_support', |
257 'rtc_unittest_main', | 257 'rtc_unittest_main', |
kjellander_webrtc
2016/08/22 09:37:46
I'd prefer if you add a dependency on 'rtc_media'
nisse-webrtc
2016/08/22 09:45:45
Done.
| |
258 ], | 258 ], |
259 'sources': [ | 259 'sources': [ |
260 'base/codec_unittest.cc', | 260 'base/codec_unittest.cc', |
261 'base/rtpdataengine_unittest.cc', | 261 'base/rtpdataengine_unittest.cc', |
262 'base/rtpdump_unittest.cc', | 262 'base/rtpdump_unittest.cc', |
263 'base/rtputils_unittest.cc', | 263 'base/rtputils_unittest.cc', |
264 'base/streamparams_unittest.cc', | 264 'base/streamparams_unittest.cc', |
265 'base/turnutils_unittest.cc', | 265 'base/turnutils_unittest.cc', |
266 'base/videoadapter_unittest.cc', | 266 'base/videoadapter_unittest.cc', |
267 'base/videobroadcaster_unittest.cc', | 267 'base/videobroadcaster_unittest.cc', |
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356 'sources': [ | 356 'sources': [ |
357 'rtc_media_unittests.isolate', | 357 'rtc_media_unittests.isolate', |
358 ], | 358 ], |
359 }, | 359 }, |
360 ], | 360 ], |
361 }], | 361 }], |
362 ], # conditions | 362 ], # conditions |
363 }], # include_tests==1 | 363 }], # include_tests==1 |
364 ], # conditions | 364 ], # conditions |
365 } | 365 } |
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