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Issue 1834163002: Revert of Remove libpeerconnection target from third_party/libjingle (Closed) Base URL: http://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 8 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/features.gni") 5 import("//build/config/features.gni")
6 6
7 # From third_party/libjingle/libjingle.gyp's target_defaults. 7 # From third_party/libjingle/libjingle.gyp's target_defaults.
8 config("jingle_unexported_configs") { 8 config("jingle_unexported_configs") {
9 defines = [ 9 defines = [
10 "EXPAT_RELATIVE_PATH", 10 "EXPAT_RELATIVE_PATH",
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362 "../webrtc/media/base/videocapturer.cc", 362 "../webrtc/media/base/videocapturer.cc",
363 "../webrtc/media/base/videocapturer.h", 363 "../webrtc/media/base/videocapturer.h",
364 "../webrtc/media/base/videocommon.cc", 364 "../webrtc/media/base/videocommon.cc",
365 "../webrtc/media/base/videocommon.h", 365 "../webrtc/media/base/videocommon.h",
366 "../webrtc/media/base/videoframe.cc", 366 "../webrtc/media/base/videoframe.cc",
367 "../webrtc/media/base/videoframe.h", 367 "../webrtc/media/base/videoframe.h",
368 "../webrtc/media/base/videoframefactory.cc", 368 "../webrtc/media/base/videoframefactory.cc",
369 "../webrtc/media/base/videoframefactory.h", 369 "../webrtc/media/base/videoframefactory.h",
370 "../webrtc/media/base/videosourcebase.cc", 370 "../webrtc/media/base/videosourcebase.cc",
371 "../webrtc/media/base/videosourcebase.h", 371 "../webrtc/media/base/videosourcebase.h",
372 "../webrtc/media/engine/simulcast.cc",
373 "../webrtc/media/engine/simulcast.h",
374 "../webrtc/media/engine/webrtccommon.h", 372 "../webrtc/media/engine/webrtccommon.h",
375 "../webrtc/media/engine/webrtcmediaengine.cc",
376 "../webrtc/media/engine/webrtcmediaengine.h",
377 "../webrtc/media/engine/webrtcvideoengine2.cc",
378 "../webrtc/media/engine/webrtcvideoengine2.h",
379 "../webrtc/media/engine/webrtcvideoframe.cc", 373 "../webrtc/media/engine/webrtcvideoframe.cc",
380 "../webrtc/media/engine/webrtcvideoframe.h", 374 "../webrtc/media/engine/webrtcvideoframe.h",
381 "../webrtc/media/engine/webrtcvideoframefactory.cc", 375 "../webrtc/media/engine/webrtcvideoframefactory.cc",
382 "../webrtc/media/engine/webrtcvideoframefactory.h", 376 "../webrtc/media/engine/webrtcvideoframefactory.h",
383 "../webrtc/media/engine/webrtcvoe.h", 377 "../webrtc/media/engine/webrtcvoe.h",
384 "../webrtc/media/engine/webrtcvoiceengine.cc",
385 "../webrtc/media/engine/webrtcvoiceengine.h",
386 "../webrtc/pc/audiomonitor.cc", 378 "../webrtc/pc/audiomonitor.cc",
387 "../webrtc/pc/audiomonitor.h", 379 "../webrtc/pc/audiomonitor.h",
388 "../webrtc/pc/bundlefilter.cc", 380 "../webrtc/pc/bundlefilter.cc",
389 "../webrtc/pc/bundlefilter.h", 381 "../webrtc/pc/bundlefilter.h",
390 "../webrtc/pc/channel.cc", 382 "../webrtc/pc/channel.cc",
391 "../webrtc/pc/channel.h", 383 "../webrtc/pc/channel.h",
392 "../webrtc/pc/channelmanager.cc", 384 "../webrtc/pc/channelmanager.cc",
393 "../webrtc/pc/channelmanager.h", 385 "../webrtc/pc/channelmanager.h",
394 "../webrtc/pc/currentspeakermonitor.cc", 386 "../webrtc/pc/currentspeakermonitor.cc",
395 "../webrtc/pc/currentspeakermonitor.h", 387 "../webrtc/pc/currentspeakermonitor.h",
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409 401
410 configs -= [ "//build/config/compiler:chromium_code" ] 402 configs -= [ "//build/config/compiler:chromium_code" ]
411 configs += [ "//build/config/compiler:no_chromium_code" ] 403 configs += [ "//build/config/compiler:no_chromium_code" ]
412 404
413 configs += [ ":jingle_unexported_configs" ] 405 configs += [ ":jingle_unexported_configs" ]
414 public_configs = [ ":jingle_public_configs" ] 406 public_configs = [ ":jingle_public_configs" ]
415 407
416 deps = [ 408 deps = [
417 ":libjingle", 409 ":libjingle",
418 "//third_party/libsrtp", 410 "//third_party/libsrtp",
419 "//third_party/webrtc",
420 "//third_party/webrtc/modules/media_file", 411 "//third_party/webrtc/modules/media_file",
421 "//third_party/webrtc/modules/video_capture", 412 "//third_party/webrtc/modules/video_capture",
422 "//third_party/webrtc/modules/video_render", 413 "//third_party/webrtc/modules/video_render",
423 "//third_party/webrtc/system_wrappers",
424 "//third_party/webrtc/voice_engine",
425 ] 414 ]
426 415
427 if (!is_ios) { 416 if (!is_ios) {
428 # TODO(mallinath) - Enable SCTP for iOS. 417 # TODO(mallinath) - Enable SCTP for iOS.
429 sources += [ 418 sources += [
430 "../webrtc/media/sctp/sctpdataengine.cc", 419 "../webrtc/media/sctp/sctpdataengine.cc",
431 "../webrtc/media/sctp/sctpdataengine.h", 420 "../webrtc/media/sctp/sctpdataengine.h",
432 ] 421 ]
433 defines = [ "HAVE_SCTP" ] 422 defines = [ "HAVE_SCTP" ]
434 deps += [ "//third_party/usrsctp" ] 423 deps += [ "//third_party/usrsctp" ]
435 } 424 }
436 } 425 }
437 426
427 source_set("libpeerconnection") {
428 sources = [
429 "../webrtc/media/engine/simulcast.cc",
430 "../webrtc/media/engine/simulcast.h",
431 "../webrtc/media/engine/webrtcmediaengine.cc",
432 "../webrtc/media/engine/webrtcmediaengine.h",
433 "../webrtc/media/engine/webrtcvideoengine2.cc",
434 "../webrtc/media/engine/webrtcvideoengine2.h",
435 "../webrtc/media/engine/webrtcvoiceengine.cc",
436 "../webrtc/media/engine/webrtcvoiceengine.h",
437 ]
438
439 configs += [ ":jingle_unexported_configs" ]
440 public_configs = [ ":jingle_public_configs" ]
441 configs -= [ "//build/config/compiler:chromium_code" ]
442 configs += [ "//build/config/compiler:no_chromium_code" ]
443
444 deps = [
445 # TODO(GYP): crbug.com/481633. Consider depending on :libjingle_webrtc
446 # instead.
447 ":libjingle_webrtc_common",
448 "//third_party/webrtc",
449 "//third_party/webrtc/system_wrappers",
450 "//third_party/webrtc/voice_engine",
451 ]
452 }
453
438 source_set("libstunprober") { 454 source_set("libstunprober") {
439 p2p_dir = "../webrtc/p2p" 455 p2p_dir = "../webrtc/p2p"
440 sources = [ 456 sources = [
441 "$p2p_dir/stunprober/stunprober.cc", 457 "$p2p_dir/stunprober/stunprober.cc",
442 ] 458 ]
443 459
444 deps = [ 460 deps = [
445 ":libjingle_webrtc_common", 461 ":libjingle_webrtc_common",
446 "//third_party/webrtc/base:rtc_base", 462 "//third_party/webrtc/base:rtc_base",
447 ] 463 ]
448 } 464 }
449 } # enable_webrtc 465 } # enable_webrtc
450 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. 466 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.
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