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Issue 534713003: Add some more libjingle targets to the GN build. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: git cl try 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 7
7 # From third_party/libjingle/libjingle.gyp's target_defaults. 8 # From third_party/libjingle/libjingle.gyp's target_defaults.
8 config("jingle_unexported_configs") { 9 config("jingle_unexported_configs") {
9 defines = [ 10 defines = [
10 "EXPAT_RELATIVE_PATH", 11 "EXPAT_RELATIVE_PATH",
11 "FEATURE_ENABLE_SSL", 12 "FEATURE_ENABLE_SSL",
12 "GTEST_RELATIVE_PATH", 13 "GTEST_RELATIVE_PATH",
13 "HAVE_SRTP", 14 "HAVE_SRTP",
14 "HAVE_WEBRTC_VIDEO", 15 "HAVE_WEBRTC_VIDEO",
15 "HAVE_WEBRTC_VOICE", 16 "HAVE_WEBRTC_VOICE",
(...skipping 318 matching lines...) Expand 10 before | Expand all | Expand 10 after
334 ] 335 ]
335 all_dependent_configs = [ 336 all_dependent_configs = [
336 ":jingle_all_dependent_configs", 337 ":jingle_all_dependent_configs",
337 ] 338 ]
338 if (is_win) { 339 if (is_win) {
339 # TODO(jschuh): crbug.com/167187 fix size_t to int truncations. 340 # TODO(jschuh): crbug.com/167187 fix size_t to int truncations.
340 cflags = [ "/wd4309" ] 341 cflags = [ "/wd4309" ]
341 } 342 }
342 } 343 }
343 344
345 if (enable_webrtc) {
346
347 source_set("libjingle_webrtc") {
348 sources = [
349 "overrides/init_webrtc.cc",
350 "overrides/init_webrtc.h",
351 ]
352 deps = [ ":libjingle_webrtc_common" ]
353 }
354
355 # Note: this does not support the shared library build of libpeerconnection
356 # as is supported in the GYP build. It's not clear what this is used for.
357 source_set("libjingle_webrtc_common") {
358 sources = [
359 "overrides/talk/media/webrtc/webrtcexport.h",
360
361 "source/talk/app/webrtc/audiotrack.cc",
362 "source/talk/app/webrtc/audiotrack.h",
363 "source/talk/app/webrtc/audiotrackrenderer.cc",
364 "source/talk/app/webrtc/audiotrackrenderer.h",
365 "source/talk/app/webrtc/datachannel.cc",
366 "source/talk/app/webrtc/datachannel.h",
367 "source/talk/app/webrtc/dtmfsender.cc",
368 "source/talk/app/webrtc/dtmfsender.h",
369 "source/talk/app/webrtc/jsep.h",
370 "source/talk/app/webrtc/jsepicecandidate.cc",
371 "source/talk/app/webrtc/jsepicecandidate.h",
372 "source/talk/app/webrtc/jsepsessiondescription.cc",
373 "source/talk/app/webrtc/jsepsessiondescription.h",
374 "source/talk/app/webrtc/localaudiosource.cc",
375 "source/talk/app/webrtc/localaudiosource.h",
376 "source/talk/app/webrtc/mediaconstraintsinterface.cc",
377 "source/talk/app/webrtc/mediaconstraintsinterface.h",
378 "source/talk/app/webrtc/mediastream.cc",
379 "source/talk/app/webrtc/mediastream.h",
380 "source/talk/app/webrtc/mediastreamhandler.cc",
381 "source/talk/app/webrtc/mediastreamhandler.h",
382 "source/talk/app/webrtc/mediastreaminterface.h",
383 "source/talk/app/webrtc/mediastreamprovider.h",
384 "source/talk/app/webrtc/mediastreamproxy.h",
385 "source/talk/app/webrtc/mediastreamsignaling.cc",
386 "source/talk/app/webrtc/mediastreamsignaling.h",
387 "source/talk/app/webrtc/mediastreamtrack.h",
388 "source/talk/app/webrtc/mediastreamtrackproxy.h",
389 "source/talk/app/webrtc/notifier.h",
390 "source/talk/app/webrtc/peerconnection.cc",
391 "source/talk/app/webrtc/peerconnection.h",
392 "source/talk/app/webrtc/peerconnectionfactory.cc",
393 "source/talk/app/webrtc/peerconnectionfactory.h",
394 "source/talk/app/webrtc/peerconnectioninterface.h",
395 "source/talk/app/webrtc/portallocatorfactory.cc",
396 "source/talk/app/webrtc/portallocatorfactory.h",
397 "source/talk/app/webrtc/remoteaudiosource.cc",
398 "source/talk/app/webrtc/remoteaudiosource.h",
399 "source/talk/app/webrtc/remotevideocapturer.cc",
400 "source/talk/app/webrtc/remotevideocapturer.h",
401 "source/talk/app/webrtc/sctputils.cc",
402 "source/talk/app/webrtc/sctputils.h",
403 "source/talk/app/webrtc/statscollector.cc",
404 "source/talk/app/webrtc/statscollector.h",
405 "source/talk/app/webrtc/statstypes.h",
406 "source/talk/app/webrtc/streamcollection.h",
407 "source/talk/app/webrtc/umametrics.h",
408 "source/talk/app/webrtc/videosource.cc",
409 "source/talk/app/webrtc/videosource.h",
410 "source/talk/app/webrtc/videosourceinterface.h",
411 "source/talk/app/webrtc/videosourceproxy.h",
412 "source/talk/app/webrtc/videotrack.cc",
413 "source/talk/app/webrtc/videotrack.h",
414 "source/talk/app/webrtc/videotrackrenderers.cc",
415 "source/talk/app/webrtc/videotrackrenderers.h",
416 "source/talk/app/webrtc/webrtcsdp.cc",
417 "source/talk/app/webrtc/webrtcsdp.h",
418 "source/talk/app/webrtc/webrtcsession.cc",
419 "source/talk/app/webrtc/webrtcsession.h",
420 "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc",
421 "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h",
422 "source/talk/media/base/audiorenderer.h",
423 "source/talk/media/base/capturemanager.cc",
424 "source/talk/media/base/capturemanager.h",
425 "source/talk/media/base/capturerenderadapter.cc",
426 "source/talk/media/base/capturerenderadapter.h",
427 "source/talk/media/base/codec.cc",
428 "source/talk/media/base/codec.h",
429 "source/talk/media/base/constants.cc",
430 "source/talk/media/base/constants.h",
431 "source/talk/media/base/cryptoparams.h",
432 "source/talk/media/base/filemediaengine.cc",
433 "source/talk/media/base/filemediaengine.h",
434 "source/talk/media/base/hybriddataengine.h",
435 "source/talk/media/base/mediachannel.h",
436 "source/talk/media/base/mediaengine.cc",
437 "source/talk/media/base/mediaengine.h",
438 "source/talk/media/base/rtpdataengine.cc",
439 "source/talk/media/base/rtpdataengine.h",
440 "source/talk/media/base/rtpdump.cc",
441 "source/talk/media/base/rtpdump.h",
442 "source/talk/media/base/rtputils.cc",
443 "source/talk/media/base/rtputils.h",
444 "source/talk/media/base/streamparams.cc",
445 "source/talk/media/base/streamparams.h",
446 "source/talk/media/base/videoadapter.cc",
447 "source/talk/media/base/videoadapter.h",
448 "source/talk/media/base/videocapturer.cc",
449 "source/talk/media/base/videocapturer.h",
450 "source/talk/media/base/videocommon.cc",
451 "source/talk/media/base/videocommon.h",
452 "source/talk/media/base/videoframe.cc",
453 "source/talk/media/base/videoframe.h",
454 "source/talk/media/devices/dummydevicemanager.cc",
455 "source/talk/media/devices/dummydevicemanager.h",
456 "source/talk/media/devices/filevideocapturer.cc",
457 "source/talk/media/devices/filevideocapturer.h",
458 "source/talk/media/webrtc/webrtccommon.h",
459 "source/talk/media/webrtc/webrtcpassthroughrender.cc",
460 "source/talk/media/webrtc/webrtcpassthroughrender.h",
461 "source/talk/media/webrtc/webrtctexturevideoframe.cc",
462 "source/talk/media/webrtc/webrtctexturevideoframe.h",
463 "source/talk/media/webrtc/webrtcvideocapturer.cc",
464 "source/talk/media/webrtc/webrtcvideocapturer.h",
465 "source/talk/media/webrtc/webrtcvideoframe.cc",
466 "source/talk/media/webrtc/webrtcvideoframe.h",
467 "source/talk/media/webrtc/webrtcvideoframefactory.cc",
468 "source/talk/media/webrtc/webrtcvideoframefactory.h",
469 "source/talk/media/webrtc/webrtcvie.h",
470 "source/talk/media/webrtc/webrtcvoe.h",
471 "source/talk/session/media/audiomonitor.cc",
472 "source/talk/session/media/audiomonitor.h",
473 "source/talk/session/media/bundlefilter.cc",
474 "source/talk/session/media/bundlefilter.h",
475 "source/talk/session/media/call.cc",
476 "source/talk/session/media/call.h",
477 "source/talk/session/media/channel.cc",
478 "source/talk/session/media/channel.h",
479 "source/talk/session/media/channelmanager.cc",
480 "source/talk/session/media/channelmanager.h",
481 "source/talk/session/media/currentspeakermonitor.cc",
482 "source/talk/session/media/currentspeakermonitor.h",
483 "source/talk/session/media/externalhmac.cc",
484 "source/talk/session/media/externalhmac.h",
485 "source/talk/session/media/mediamessages.cc",
486 "source/talk/session/media/mediamessages.h",
487 "source/talk/session/media/mediamonitor.cc",
488 "source/talk/session/media/mediamonitor.h",
489 "source/talk/session/media/mediasession.cc",
490 "source/talk/session/media/mediasession.h",
491 "source/talk/session/media/mediasessionclient.cc",
492 "source/talk/session/media/mediasessionclient.h",
493 "source/talk/session/media/mediasink.h",
494 "source/talk/session/media/rtcpmuxfilter.cc",
495 "source/talk/session/media/rtcpmuxfilter.h",
496 "source/talk/session/media/soundclip.cc",
497 "source/talk/session/media/soundclip.h",
498 "source/talk/session/media/srtpfilter.cc",
499 "source/talk/session/media/srtpfilter.h",
500 "source/talk/session/media/typingmonitor.cc",
501 "source/talk/session/media/typingmonitor.h",
502 "source/talk/session/media/voicechannel.h",
503 "source/talk/session/tunnel/pseudotcpchannel.cc",
504 "source/talk/session/tunnel/pseudotcpchannel.h",
505 "source/talk/session/tunnel/tunnelsessionclient.cc",
506 "source/talk/session/tunnel/tunnelsessionclient.h",
507 ]
508
509 configs += [ ":jingle_unexported_configs" ]
510 direct_dependent_configs = [ ":jingle_direct_dependent_configs" ]
511
512 deps = [
513 "//third_party/libsrtp",
514 "//third_party/webrtc/modules/media_file",
515 "//third_party/webrtc/modules/video_capture",
516 "//third_party/webrtc/modules/video_render",
517 ]
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) {
524 # TODO(mallinath) - Enable SCTP for iOS.
525 sources += [
526 "source/talk/media/sctp/sctpdataengine.cc",
527 "source/talk/media/sctp/sctpdataengine.h",
528 ]
529 defines = [ "HAVE_SCTP" ]
530 deps += [ "//third_party/usrsctp" ]
531 }
532
533 if (is_clang) {
534 cflags = [ "-Wno-unused-private-field" ]
535 }
536 }
537
538 # Note: this does not support the shared library build of libpeerconnection
539 # as is supported in the GYP build. It's not clear what this is used for.
540 source_set("libpeerconnection") {
541 sources = [
542 "source/talk/media/webrtc/webrtcmediaengine.cc",
543 "source/talk/media/webrtc/webrtcmediaengine.h",
544 "source/talk/media/webrtc/webrtcvideoengine.cc",
545 "source/talk/media/webrtc/webrtcvideoengine.h",
546 "source/talk/media/webrtc/webrtcvideoengine2.cc",
547 "source/talk/media/webrtc/webrtcvideoengine2.h",
548 "source/talk/media/webrtc/webrtcvoiceengine.cc",
549 "source/talk/media/webrtc/webrtcvoiceengine.h",
550 ]
551
552 configs += [ ":jingle_unexported_configs" ]
553 direct_dependent_configs = [ ":jingle_direct_dependent_configs" ]
554
555 deps = [
556 ":libjingle_webrtc_common",
557 "//third_party/webrtc",
558 "//third_party/webrtc/system_wrappers",
559 "//third_party/webrtc/voice_engine",
560 ]
561 }
562
563 } # enable_webrtc
564
344 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. 565 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.
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