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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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289 public_configs = [ ":jingle_public_configs" ] | 289 public_configs = [ ":jingle_public_configs" ] |
290 public_deps = [ | 290 public_deps = [ |
291 ":libjingle_webrtc_common", | 291 ":libjingle_webrtc_common", |
292 ] | 292 ] |
293 } | 293 } |
294 | 294 |
295 # Note: this does not support the shared library build of libpeerconnection | 295 # Note: this does not support the shared library build of libpeerconnection |
296 # as is supported in the GYP build. It's not clear what this is used for. | 296 # as is supported in the GYP build. It's not clear what this is used for. |
297 source_set("libjingle_webrtc_common") { | 297 source_set("libjingle_webrtc_common") { |
298 sources = [ | 298 sources = [ |
| 299 "../webrtc/api/audiotrack.cc", |
| 300 "../webrtc/api/audiotrack.h", |
| 301 "../webrtc/api/datachannel.cc", |
| 302 "../webrtc/api/datachannel.h", |
| 303 "../webrtc/api/dtlsidentitystore.cc", |
| 304 "../webrtc/api/dtlsidentitystore.h", |
| 305 "../webrtc/api/dtmfsender.cc", |
| 306 "../webrtc/api/dtmfsender.h", |
| 307 "../webrtc/api/jsep.h", |
| 308 "../webrtc/api/jsepicecandidate.cc", |
| 309 "../webrtc/api/jsepicecandidate.h", |
| 310 "../webrtc/api/jsepsessiondescription.cc", |
| 311 "../webrtc/api/jsepsessiondescription.h", |
| 312 "../webrtc/api/localaudiosource.cc", |
| 313 "../webrtc/api/localaudiosource.h", |
| 314 "../webrtc/api/mediaconstraintsinterface.cc", |
| 315 "../webrtc/api/mediaconstraintsinterface.h", |
| 316 "../webrtc/api/mediacontroller.cc", |
| 317 "../webrtc/api/mediacontroller.h", |
| 318 "../webrtc/api/mediastream.cc", |
| 319 "../webrtc/api/mediastream.h", |
| 320 "../webrtc/api/mediastreamhandler.cc", |
| 321 "../webrtc/api/mediastreamhandler.h", |
| 322 "../webrtc/api/mediastreaminterface.h", |
| 323 "../webrtc/api/mediastreamobserver.cc", |
| 324 "../webrtc/api/mediastreamobserver.h", |
| 325 "../webrtc/api/mediastreamprovider.h", |
| 326 "../webrtc/api/mediastreamproxy.h", |
| 327 "../webrtc/api/mediastreamtrack.h", |
| 328 "../webrtc/api/mediastreamtrackproxy.h", |
| 329 "../webrtc/api/notifier.h", |
| 330 "../webrtc/api/peerconnection.cc", |
| 331 "../webrtc/api/peerconnection.h", |
| 332 "../webrtc/api/peerconnectionfactory.cc", |
| 333 "../webrtc/api/peerconnectionfactory.h", |
| 334 "../webrtc/api/peerconnectioninterface.h", |
| 335 "../webrtc/api/portallocatorfactory.cc", |
| 336 "../webrtc/api/portallocatorfactory.h", |
| 337 "../webrtc/api/remoteaudiosource.cc", |
| 338 "../webrtc/api/remoteaudiosource.h", |
| 339 "../webrtc/api/remoteaudiotrack.cc", |
| 340 "../webrtc/api/remoteaudiotrack.h", |
| 341 "../webrtc/api/remotevideocapturer.cc", |
| 342 "../webrtc/api/remotevideocapturer.h", |
| 343 "../webrtc/api/rtpreceiver.cc", |
| 344 "../webrtc/api/rtpreceiver.h", |
| 345 "../webrtc/api/rtpreceiverinterface.h", |
| 346 "../webrtc/api/rtpsender.cc", |
| 347 "../webrtc/api/rtpsender.h", |
| 348 "../webrtc/api/rtpsenderinterface.h", |
| 349 "../webrtc/api/sctputils.cc", |
| 350 "../webrtc/api/sctputils.h", |
| 351 "../webrtc/api/statscollector.cc", |
| 352 "../webrtc/api/statscollector.h", |
| 353 "../webrtc/api/statstypes.cc", |
| 354 "../webrtc/api/statstypes.h", |
| 355 "../webrtc/api/streamcollection.h", |
| 356 "../webrtc/api/umametrics.h", |
| 357 "../webrtc/api/videosource.cc", |
| 358 "../webrtc/api/videosource.h", |
| 359 "../webrtc/api/videosourceinterface.h", |
| 360 "../webrtc/api/videosourceproxy.h", |
| 361 "../webrtc/api/videotrack.cc", |
| 362 "../webrtc/api/videotrack.h", |
| 363 "../webrtc/api/videotrackrenderers.cc", |
| 364 "../webrtc/api/videotrackrenderers.h", |
| 365 "../webrtc/api/webrtcsdp.cc", |
| 366 "../webrtc/api/webrtcsdp.h", |
| 367 "../webrtc/api/webrtcsession.cc", |
| 368 "../webrtc/api/webrtcsession.h", |
| 369 "../webrtc/api/webrtcsessiondescriptionfactory.cc", |
| 370 "../webrtc/api/webrtcsessiondescriptionfactory.h", |
299 "../webrtc/media/base/audiorenderer.h", | 371 "../webrtc/media/base/audiorenderer.h", |
300 "../webrtc/media/base/capturemanager.cc", | 372 "../webrtc/media/base/capturemanager.cc", |
301 "../webrtc/media/base/capturemanager.h", | 373 "../webrtc/media/base/capturemanager.h", |
302 "../webrtc/media/base/capturerenderadapter.cc", | 374 "../webrtc/media/base/capturerenderadapter.cc", |
303 "../webrtc/media/base/capturerenderadapter.h", | 375 "../webrtc/media/base/capturerenderadapter.h", |
304 "../webrtc/media/base/codec.cc", | 376 "../webrtc/media/base/codec.cc", |
305 "../webrtc/media/base/codec.h", | 377 "../webrtc/media/base/codec.h", |
306 "../webrtc/media/base/constants.cc", | 378 "../webrtc/media/base/constants.cc", |
307 "../webrtc/media/base/constants.h", | 379 "../webrtc/media/base/constants.h", |
308 "../webrtc/media/base/cryptoparams.h", | 380 "../webrtc/media/base/cryptoparams.h", |
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331 "../webrtc/media/base/videoframefactory.cc", | 403 "../webrtc/media/base/videoframefactory.cc", |
332 "../webrtc/media/base/videoframefactory.h", | 404 "../webrtc/media/base/videoframefactory.h", |
333 "../webrtc/media/devices/dummydevicemanager.cc", | 405 "../webrtc/media/devices/dummydevicemanager.cc", |
334 "../webrtc/media/devices/dummydevicemanager.h", | 406 "../webrtc/media/devices/dummydevicemanager.h", |
335 "../webrtc/media/webrtc/webrtccommon.h", | 407 "../webrtc/media/webrtc/webrtccommon.h", |
336 "../webrtc/media/webrtc/webrtcvideoframe.cc", | 408 "../webrtc/media/webrtc/webrtcvideoframe.cc", |
337 "../webrtc/media/webrtc/webrtcvideoframe.h", | 409 "../webrtc/media/webrtc/webrtcvideoframe.h", |
338 "../webrtc/media/webrtc/webrtcvideoframefactory.cc", | 410 "../webrtc/media/webrtc/webrtcvideoframefactory.cc", |
339 "../webrtc/media/webrtc/webrtcvideoframefactory.h", | 411 "../webrtc/media/webrtc/webrtcvideoframefactory.h", |
340 "../webrtc/media/webrtc/webrtcvoe.h", | 412 "../webrtc/media/webrtc/webrtcvoe.h", |
341 "source/talk/app/webrtc/audiotrack.cc", | |
342 "source/talk/app/webrtc/audiotrack.h", | |
343 "source/talk/app/webrtc/datachannel.cc", | |
344 "source/talk/app/webrtc/datachannel.h", | |
345 "source/talk/app/webrtc/dtlsidentitystore.cc", | |
346 "source/talk/app/webrtc/dtlsidentitystore.h", | |
347 "source/talk/app/webrtc/dtmfsender.cc", | |
348 "source/talk/app/webrtc/dtmfsender.h", | |
349 "source/talk/app/webrtc/jsep.h", | |
350 "source/talk/app/webrtc/jsepicecandidate.cc", | |
351 "source/talk/app/webrtc/jsepicecandidate.h", | |
352 "source/talk/app/webrtc/jsepsessiondescription.cc", | |
353 "source/talk/app/webrtc/jsepsessiondescription.h", | |
354 "source/talk/app/webrtc/localaudiosource.cc", | |
355 "source/talk/app/webrtc/localaudiosource.h", | |
356 "source/talk/app/webrtc/mediaconstraintsinterface.cc", | |
357 "source/talk/app/webrtc/mediaconstraintsinterface.h", | |
358 "source/talk/app/webrtc/mediacontroller.cc", | |
359 "source/talk/app/webrtc/mediacontroller.h", | |
360 "source/talk/app/webrtc/mediastream.cc", | |
361 "source/talk/app/webrtc/mediastream.h", | |
362 "source/talk/app/webrtc/mediastreamhandler.cc", | |
363 "source/talk/app/webrtc/mediastreamhandler.h", | |
364 "source/talk/app/webrtc/mediastreaminterface.h", | |
365 "source/talk/app/webrtc/mediastreamobserver.cc", | |
366 "source/talk/app/webrtc/mediastreamobserver.h", | |
367 "source/talk/app/webrtc/mediastreamprovider.h", | |
368 "source/talk/app/webrtc/mediastreamproxy.h", | |
369 "source/talk/app/webrtc/mediastreamtrack.h", | |
370 "source/talk/app/webrtc/mediastreamtrackproxy.h", | |
371 "source/talk/app/webrtc/notifier.h", | |
372 "source/talk/app/webrtc/peerconnection.cc", | |
373 "source/talk/app/webrtc/peerconnection.h", | |
374 "source/talk/app/webrtc/peerconnectionfactory.cc", | |
375 "source/talk/app/webrtc/peerconnectionfactory.h", | |
376 "source/talk/app/webrtc/peerconnectioninterface.h", | |
377 "source/talk/app/webrtc/portallocatorfactory.cc", | |
378 "source/talk/app/webrtc/portallocatorfactory.h", | |
379 "source/talk/app/webrtc/remoteaudiosource.cc", | |
380 "source/talk/app/webrtc/remoteaudiosource.h", | |
381 "source/talk/app/webrtc/remoteaudiotrack.cc", | |
382 "source/talk/app/webrtc/remoteaudiotrack.h", | |
383 "source/talk/app/webrtc/remotevideocapturer.cc", | |
384 "source/talk/app/webrtc/remotevideocapturer.h", | |
385 "source/talk/app/webrtc/rtpreceiver.cc", | |
386 "source/talk/app/webrtc/rtpreceiver.h", | |
387 "source/talk/app/webrtc/rtpreceiverinterface.h", | |
388 "source/talk/app/webrtc/rtpsender.cc", | |
389 "source/talk/app/webrtc/rtpsender.h", | |
390 "source/talk/app/webrtc/rtpsenderinterface.h", | |
391 "source/talk/app/webrtc/sctputils.cc", | |
392 "source/talk/app/webrtc/sctputils.h", | |
393 "source/talk/app/webrtc/statscollector.cc", | |
394 "source/talk/app/webrtc/statscollector.h", | |
395 "source/talk/app/webrtc/statstypes.cc", | |
396 "source/talk/app/webrtc/statstypes.h", | |
397 "source/talk/app/webrtc/streamcollection.h", | |
398 "source/talk/app/webrtc/umametrics.h", | |
399 "source/talk/app/webrtc/videosource.cc", | |
400 "source/talk/app/webrtc/videosource.h", | |
401 "source/talk/app/webrtc/videosourceinterface.h", | |
402 "source/talk/app/webrtc/videosourceproxy.h", | |
403 "source/talk/app/webrtc/videotrack.cc", | |
404 "source/talk/app/webrtc/videotrack.h", | |
405 "source/talk/app/webrtc/videotrackrenderers.cc", | |
406 "source/talk/app/webrtc/videotrackrenderers.h", | |
407 "source/talk/app/webrtc/webrtcsdp.cc", | |
408 "source/talk/app/webrtc/webrtcsdp.h", | |
409 "source/talk/app/webrtc/webrtcsession.cc", | |
410 "source/talk/app/webrtc/webrtcsession.h", | |
411 "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc", | |
412 "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h", | |
413 "source/talk/session/media/audiomonitor.cc", | 413 "source/talk/session/media/audiomonitor.cc", |
414 "source/talk/session/media/audiomonitor.h", | 414 "source/talk/session/media/audiomonitor.h", |
415 "source/talk/session/media/bundlefilter.cc", | 415 "source/talk/session/media/bundlefilter.cc", |
416 "source/talk/session/media/bundlefilter.h", | 416 "source/talk/session/media/bundlefilter.h", |
417 "source/talk/session/media/channel.cc", | 417 "source/talk/session/media/channel.cc", |
418 "source/talk/session/media/channel.h", | 418 "source/talk/session/media/channel.h", |
419 "source/talk/session/media/channelmanager.cc", | 419 "source/talk/session/media/channelmanager.cc", |
420 "source/talk/session/media/channelmanager.h", | 420 "source/talk/session/media/channelmanager.h", |
421 "source/talk/session/media/currentspeakermonitor.cc", | 421 "source/talk/session/media/currentspeakermonitor.cc", |
422 "source/talk/session/media/currentspeakermonitor.h", | 422 "source/talk/session/media/currentspeakermonitor.h", |
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494 "$p2p_dir/stunprober/stunprober.cc", | 494 "$p2p_dir/stunprober/stunprober.cc", |
495 ] | 495 ] |
496 | 496 |
497 deps = [ | 497 deps = [ |
498 ":libjingle_webrtc_common", | 498 ":libjingle_webrtc_common", |
499 "//third_party/webrtc/base:rtc_base", | 499 "//third_party/webrtc/base:rtc_base", |
500 ] | 500 ] |
501 } | 501 } |
502 } # enable_webrtc | 502 } # enable_webrtc |
503 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. | 503 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. |
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