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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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| 358 '<(DEPTH)/third_party/webrtc/pc/rtcpmuxfilter.h', | 358 '<(DEPTH)/third_party/webrtc/pc/rtcpmuxfilter.h', |
| 359 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.cc', | 359 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.cc', |
| 360 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.h', | 360 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.h', |
| 361 '<(DEPTH)/third_party/webrtc/pc/voicechannel.h', | 361 '<(DEPTH)/third_party/webrtc/pc/voicechannel.h', |
| 362 ], | 362 ], |
| 363 'dependencies': [ | 363 'dependencies': [ |
| 364 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', | 364 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', |
| 365 '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib', | 365 '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib', |
| 366 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file', | 366 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file', |
| 367 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture', | 367 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture', |
| 368 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render', | |
| 369 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng
ine', | 368 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng
ine', |
| 370 '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc', | 369 '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc', |
| 371 'libjingle', | 370 'libjingle', |
| 372 ], | 371 ], |
| 373 }, # target libjingle_webrtc_common | 372 }, # target libjingle_webrtc_common |
| 374 { | 373 { |
| 375 # GN version: //third_party/libjingle:libjingle_webrtc | 374 # GN version: //third_party/libjingle:libjingle_webrtc |
| 376 'target_name': 'libjingle_webrtc', | 375 'target_name': 'libjingle_webrtc', |
| 377 'type': 'static_library', | 376 'type': 'static_library', |
| 378 'sources': [ | 377 'sources': [ |
| 379 'overrides/init_webrtc.cc', | 378 'overrides/init_webrtc.cc', |
| 380 'overrides/init_webrtc.h', | 379 'overrides/init_webrtc.h', |
| 381 ], | 380 ], |
| 382 'dependencies': [ | 381 'dependencies': [ |
| 383 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing', | 382 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing', |
| 384 'libjingle_webrtc_common', | 383 'libjingle_webrtc_common', |
| 385 ], | 384 ], |
| 386 }, | 385 }, |
| 387 ], | 386 ], |
| 388 }], | 387 }], |
| 389 ], | 388 ], |
| 390 } | 389 } |
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