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Issue 534813002: Remove some rtc hacks in the GN build. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: merge Created 6 years, 3 months ago
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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 = [
(...skipping 498 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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