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| 1 # Copyright 2014 The Chromium Authors. All rights reserved. | 1 # Copyright 2014 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 import("//build/config/crypto.gni") | 5 import("//build/config/crypto.gni") |
| 6 import("//build/config/features.gni") | 6 import("//build/config/features.gni") |
| 7 | 7 |
| 8 # From third_party/libjingle/libjingle.gyp's target_defaults. | 8 # From third_party/libjingle/libjingle.gyp's target_defaults. |
| 9 config("jingle_unexported_configs") { | 9 config("jingle_unexported_configs") { |
| 10 defines = [ | 10 defines = [ |
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| 509 configs += [ ":jingle_unexported_configs" ] | 509 configs += [ ":jingle_unexported_configs" ] |
| 510 direct_dependent_configs = [ ":jingle_direct_dependent_configs" ] | 510 direct_dependent_configs = [ ":jingle_direct_dependent_configs" ] |
| 511 | 511 |
| 512 deps = [ | 512 deps = [ |
| 513 "//third_party/libsrtp", | 513 "//third_party/libsrtp", |
| 514 "//third_party/webrtc/modules/media_file", | 514 "//third_party/webrtc/modules/media_file", |
| 515 "//third_party/webrtc/modules/video_capture", | 515 "//third_party/webrtc/modules/video_capture", |
| 516 "//third_party/webrtc/modules/video_render", | 516 "//third_party/webrtc/modules/video_render", |
| 517 ] | 517 ] |
| 518 | 518 |
| 519 # TODO(GYP) this should be removed and we should get this config by one of | |
| 520 # the webrtc targets specifying it for the direct_dependent_configs. | |
| 521 #configs += [ "//third_party/webrtc:common_config" ] # TODO(GYP) | |
| 522 | |
| 523 if (!is_ios) { | 519 if (!is_ios) { |
| 524 # TODO(mallinath) - Enable SCTP for iOS. | 520 # TODO(mallinath) - Enable SCTP for iOS. |
| 525 sources += [ | 521 sources += [ |
| 526 "source/talk/media/sctp/sctpdataengine.cc", | 522 "source/talk/media/sctp/sctpdataengine.cc", |
| 527 "source/talk/media/sctp/sctpdataengine.h", | 523 "source/talk/media/sctp/sctpdataengine.h", |
| 528 ] | 524 ] |
| 529 defines = [ "HAVE_SCTP" ] | 525 defines = [ "HAVE_SCTP" ] |
| 530 deps += [ "//third_party/usrsctp" ] | 526 deps += [ "//third_party/usrsctp" ] |
| 531 } | 527 } |
| 532 | 528 |
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| 556 ":libjingle_webrtc_common", | 552 ":libjingle_webrtc_common", |
| 557 "//third_party/webrtc", | 553 "//third_party/webrtc", |
| 558 "//third_party/webrtc/system_wrappers", | 554 "//third_party/webrtc/system_wrappers", |
| 559 "//third_party/webrtc/voice_engine", | 555 "//third_party/webrtc/voice_engine", |
| 560 ] | 556 ] |
| 561 } | 557 } |
| 562 | 558 |
| 563 } # enable_webrtc | 559 } # enable_webrtc |
| 564 | 560 |
| 565 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. | 561 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. |
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