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1 # Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 # Copyright (c) 2012 The Chromium Authors. All rights reserved. |
2 # Use of this source code is governed by a BSD-style license that can be | 2 # Use of this source code is governed by a BSD-style license that can be |
3 # found in the LICENSE file. | 3 # found in the LICENSE file. |
4 | 4 |
5 { | 5 { |
6 'includes': [ | 6 'includes': [ |
7 '../../build/win_precompile.gypi', | 7 '../../build/win_precompile.gypi', |
8 ], | 8 ], |
9 # Most of these settings have been split according to their scope into | 9 # Most of these settings have been split according to their scope into |
10 # :jingle_unexported_configs and :jingle_public_config in the GN build. | 10 # :jingle_unexported_configs and :jingle_public_config in the GN build. |
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21 'HAVE_WEBRTC_VOICE', | 21 'HAVE_WEBRTC_VOICE', |
22 'LOGGING_INSIDE_WEBRTC', | 22 'LOGGING_INSIDE_WEBRTC', |
23 'NO_MAIN_THREAD_WRAPPING', | 23 'NO_MAIN_THREAD_WRAPPING', |
24 'NO_SOUND_SYSTEM', | 24 'NO_SOUND_SYSTEM', |
25 'SRTP_RELATIVE_PATH', | 25 'SRTP_RELATIVE_PATH', |
26 'SSL_USE_OPENSSL', | 26 'SSL_USE_OPENSSL', |
27 'USE_WEBRTC_DEV_BRANCH', | 27 'USE_WEBRTC_DEV_BRANCH', |
28 'WEBRTC_CHROMIUM_BUILD', | 28 'WEBRTC_CHROMIUM_BUILD', |
29 ], | 29 ], |
30 'include_dirs': [ | 30 'include_dirs': [ |
31 './overrides', | |
32 '../../third_party/webrtc_overrides', | 31 '../../third_party/webrtc_overrides', |
33 './source', | |
34 '../..', | 32 '../..', |
35 '../../testing/gtest/include', | 33 '../../testing/gtest/include', |
36 '../../third_party', | 34 '../../third_party', |
37 '../../third_party/libyuv/include', | 35 '../../third_party/libyuv/include', |
38 '../../third_party/usrsctp/usrsctplib', | 36 '../../third_party/usrsctp/usrsctplib', |
39 ], | 37 ], |
40 # These dependencies have been translated into :jingle_deps in the GN build. | 38 # These dependencies have been translated into :jingle_deps in the GN build. |
41 'dependencies': [ | 39 'dependencies': [ |
42 '<(DEPTH)/base/base.gyp:base', | 40 '<(DEPTH)/base/base.gyp:base', |
43 '<(DEPTH)/net/net.gyp:net', | 41 '<(DEPTH)/net/net.gyp:net', |
44 '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl', | 42 '<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl', |
45 '<(DEPTH)/third_party/expat/expat.gyp:expat', | 43 '<(DEPTH)/third_party/expat/expat.gyp:expat', |
46 ], | 44 ], |
47 'export_dependent_settings': [ | 45 'export_dependent_settings': [ |
48 '<(DEPTH)/third_party/expat/expat.gyp:expat', | 46 '<(DEPTH)/third_party/expat/expat.gyp:expat', |
49 ], | 47 ], |
50 'direct_dependent_settings': { | 48 'direct_dependent_settings': { |
51 'include_dirs': [ | 49 'include_dirs': [ |
52 '../../third_party/webrtc_overrides', | 50 '../../third_party/webrtc_overrides', |
53 './overrides', | |
54 './source', | |
55 '../..', | 51 '../..', |
56 '../../testing/gtest/include', | 52 '../../testing/gtest/include', |
57 '../../third_party', | 53 '../../third_party', |
58 ], | 54 ], |
59 'defines': [ | 55 'defines': [ |
60 'FEATURE_ENABLE_SSL', | 56 'FEATURE_ENABLE_SSL', |
61 'FEATURE_ENABLE_VOICEMAIL', | 57 'FEATURE_ENABLE_VOICEMAIL', |
62 'EXPAT_RELATIVE_PATH', | 58 'EXPAT_RELATIVE_PATH', |
63 'GTEST_RELATIVE_PATH', | 59 'GTEST_RELATIVE_PATH', |
64 'NO_MAIN_THREAD_WRAPPING', | 60 'NO_MAIN_THREAD_WRAPPING', |
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356 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', | 352 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', |
357 '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib', | 353 '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib', |
358 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file', | 354 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file', |
359 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture', | 355 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture', |
360 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng
ine', | 356 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng
ine', |
361 '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc', | 357 '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc', |
362 'libjingle', | 358 'libjingle', |
363 ], | 359 ], |
364 }, # target libjingle_webrtc_common | 360 }, # target libjingle_webrtc_common |
365 { | 361 { |
| 362 # TODO(kjellander): Move this target into |
| 363 # //third_party/webrtc_overrides as soon as the work in |
| 364 # bugs.webrtc.org/4256 has gotten rid of the duplicated source |
| 365 # listings above. |
366 # GN version: //third_party/libjingle:libjingle_webrtc | 366 # GN version: //third_party/libjingle:libjingle_webrtc |
367 'target_name': 'libjingle_webrtc', | 367 'target_name': 'libjingle_webrtc', |
368 'type': 'static_library', | 368 'type': 'static_library', |
369 'sources': [ | 369 'sources': [ |
370 'overrides/init_webrtc.cc', | 370 '../webrtc_overrides/init_webrtc.cc', |
371 'overrides/init_webrtc.h', | 371 '../webrtc_overrides/init_webrtc.h', |
372 ], | 372 ], |
373 'dependencies': [ | 373 'dependencies': [ |
374 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing', | 374 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing', |
375 'libjingle_webrtc_common', | 375 'libjingle_webrtc_common', |
376 ], | 376 ], |
377 }, | 377 }, |
378 ], | 378 ], |
379 }], | 379 }], |
380 ], | 380 ], |
381 } | 381 } |
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