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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. |
| (...skipping 10 matching lines...) Expand all Loading... |
| 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', |
| (...skipping 291 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 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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