third_party/libjingle/BUILD.gn

   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
   3 # found in the LICENSE file.
   4
   5 import("//build/config/crypto.gni")
   6 import("//build/config/features.gni")
   7
   8 # From third_party/libjingle/libjingle.gyp's target_defaults.
   9 config("jingle_unexported_configs") {
  10   defines = [
  11     "EXPAT_RELATIVE_PATH",
  12     "FEATURE_ENABLE_SSL",
  13     "GTEST_RELATIVE_PATH",
  14     "HAVE_SRTP",
  15     "HAVE_WEBRTC_VIDEO",
  16     "HAVE_WEBRTC_VOICE",
  17     "LOGGING_INSIDE_WEBRTC",
  18     "NO_MAIN_THREAD_WRAPPING",
  19     "NO_SOUND_SYSTEM",
  20     "SRTP_RELATIVE_PATH",
  21     "USE_WEBRTC_DEV_BRANCH",
  22     "ENABLE_EXTERNAL_AUTH",
  23     "WEBRTC_CHROMIUM_BUILD",
  24   ]
  25
  26   include_dirs = [
  27     "overrides",
  28     "../../third_party/webrtc/overrides",
  29     "source",
  30     "../../testing/gtest/include",
  31     "../../third_party",
  32     "../../third_party/libyuv/include",
  33     "../../third_party/usrsctp",
  34   ]
  35
  36   # Assumes libpeer is linked statically.
  37   defines += [ "LIBPEERCONNECTION_LIB=1" ]
  38
  39   if (is_win && current_cpu == "x86") {
  40     defines += [ "_USE_32BIT_TIME_T" ]
  41   }
  42
  43   if (use_openssl) {
  44     defines += [
  45       "SSL_USE_OPENSSL",
  46       "HAVE_OPENSSL_SSL_H",
  47     ]
  48   } else {
  49     defines += [
  50       "SSL_USE_NSS",
  51       "HAVE_NSS_SSL_H",
  52       "SSL_USE_NSS_RNG",
  53     ]
  54   }
  55 }
  56
  57 # From third_party/libjingle/libjingle.gyp's target_defaults.
  58 config("jingle_public_configs") {
  59   include_dirs = [
  60     "../../third_party/webrtc/overrides",
  61     "overrides",
  62     "source",
  63     "../../testing/gtest/include",
  64     "../../third_party",
  65   ]
  66   defines = [
  67     "FEATURE_ENABLE_SSL",
  68     "FEATURE_ENABLE_VOICEMAIL",
  69     "EXPAT_RELATIVE_PATH",
  70     "GTEST_RELATIVE_PATH",
  71     "NO_MAIN_THREAD_WRAPPING",
  72     "NO_SOUND_SYSTEM",
  73   ]
  74
  75   # TODO(GYP): Port is_win blocks.
  76   if (is_linux) {
  77     defines += [
  78       "LINUX",
  79       "WEBRTC_LINUX",
  80     ]
  81   }
  82   if (is_mac) {
  83     defines += [
  84       "OSX",
  85       "WEBRTC_MAC",
  86     ]
  87   }
  88   if (is_ios) {
  89     defines += [
  90       "IOS",
  91       "WEBRTC_MAC",
  92       "WEBRTC_IOS",
  93     ]
  94   }
  95   if (is_win) {
  96     defines += [ "WEBRTC_WIN" ]
  97   }
  98   if (is_android) {
  99     defines += [ "ANDROID" ]
 100   }
 101   if (is_posix) {
 102     defines += [ "WEBRTC_POSIX" ]
 103   }
 104
 105   # TODO(GYP): Support these in GN.
 106   # if (is_bsd) {
 107   #   defines += [ "BSD" ]
 108   # }
 109   # if (is_openbsd) {
 110   #   defines += [ "OPENBSD" ]
 111   # }
 112   # if (is_freebsd) {
 113   #   defines += [ "FREEBSD" ]
 114   # }
 115   if (is_chromeos) {
 116     defines += [ "CHROMEOS" ]
 117   }
 118 }
 119
 120 # From third_party/libjingle/libjingle.gyp's target_defaults.
 121 config("jingle_all_dependent_configs") {
 122   if (is_debug) {
 123     # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
 124     # _DEBUG and remove this define. See GYP file as well.
 125     defines = [ "_DEBUG" ]
 126   }
 127 }
 128
 129 # From third_party/libjingle/libjingle.gyp's target_defaults.
 130 group("jingle_deps") {
 131   public_deps = [
 132     "//third_party/expat",
 133   ]
 134   deps = [
 135     "//base",
 136     "//net",
 137     "//crypto:platform",
 138   ]
 139 }
 140
 141 # GYP version: third_party/libjingle.gyp:libjingle
 142 static_library("libjingle") {
 143   p2p_dir = "../webrtc/p2p"
 144   xmllite_dir = "../webrtc/libjingle/xmllite"
 145   xmpp_dir = "../webrtc/libjingle/xmpp"
 146   sources = [
 147     # List from third_party/libjingle/libjingle_common.gypi
 148     "$p2p_dir/base/asyncstuntcpsocket.cc",
 149     "$p2p_dir/base/asyncstuntcpsocket.h",
 150     "$p2p_dir/base/basicpacketsocketfactory.cc",
 151     "$p2p_dir/base/basicpacketsocketfactory.h",
 152     "$p2p_dir/base/candidate.h",
 153     "$p2p_dir/base/common.h",
 154     "$p2p_dir/base/constants.cc",
 155     "$p2p_dir/base/constants.h",
 156     "$p2p_dir/base/dtlstransport.h",
 157     "$p2p_dir/base/dtlstransportchannel.cc",
 158     "$p2p_dir/base/dtlstransportchannel.h",
 159     "$p2p_dir/base/p2ptransport.cc",
 160     "$p2p_dir/base/p2ptransport.h",
 161     "$p2p_dir/base/p2ptransportchannel.cc",
 162     "$p2p_dir/base/p2ptransportchannel.h",
 163     "$p2p_dir/base/port.cc",
 164     "$p2p_dir/base/port.h",
 165     "$p2p_dir/base/portallocator.cc",
 166     "$p2p_dir/base/portallocator.h",
 167     "$p2p_dir/base/pseudotcp.cc",
 168     "$p2p_dir/base/pseudotcp.h",
 169     "$p2p_dir/base/rawtransport.cc",
 170     "$p2p_dir/base/rawtransport.h",
 171     "$p2p_dir/base/rawtransportchannel.cc",
 172     "$p2p_dir/base/rawtransportchannel.h",
 173     "$p2p_dir/base/relayport.cc",
 174     "$p2p_dir/base/relayport.h",
 175     "$p2p_dir/base/session.cc",
 176     "$p2p_dir/base/session.h",
 177     "$p2p_dir/base/sessiondescription.cc",
 178     "$p2p_dir/base/sessiondescription.h",
 179     "$p2p_dir/base/sessionid.h",
 180     "$p2p_dir/base/stun.cc",
 181     "$p2p_dir/base/stun.h",
 182     "$p2p_dir/base/stunport.cc",
 183     "$p2p_dir/base/stunport.h",
 184     "$p2p_dir/base/stunrequest.cc",
 185     "$p2p_dir/base/stunrequest.h",
 186     "$p2p_dir/base/tcpport.cc",
 187     "$p2p_dir/base/tcpport.h",
 188     "$p2p_dir/base/transport.cc",
 189     "$p2p_dir/base/transport.h",
 190     "$p2p_dir/base/transportchannel.cc",
 191     "$p2p_dir/base/transportchannel.h",
 192     "$p2p_dir/base/transportchannelimpl.h",
 193     "$p2p_dir/base/transportchannelproxy.cc",
 194     "$p2p_dir/base/transportchannelproxy.h",
 195     "$p2p_dir/base/transportdescription.cc",
 196     "$p2p_dir/base/transportdescription.h",
 197     "$p2p_dir/base/transportdescriptionfactory.cc",
 198     "$p2p_dir/base/transportdescriptionfactory.h",
 199     "$p2p_dir/base/turnport.cc",
 200     "$p2p_dir/base/turnport.h",
 201     "$p2p_dir/client/basicportallocator.cc",
 202     "$p2p_dir/client/basicportallocator.h",
 203     "$p2p_dir/client/httpportallocator.cc",
 204     "$p2p_dir/client/httpportallocator.h",
 205     "$p2p_dir/client/socketmonitor.cc",
 206     "$p2p_dir/client/socketmonitor.h",
 207     "$xmllite_dir/qname.cc",
 208     "$xmllite_dir/qname.h",
 209     "$xmllite_dir/xmlbuilder.cc",
 210     "$xmllite_dir/xmlbuilder.h",
 211     "$xmllite_dir/xmlconstants.cc",
 212     "$xmllite_dir/xmlconstants.h",
 213     "$xmllite_dir/xmlelement.cc",
 214     "$xmllite_dir/xmlelement.h",
 215     "$xmllite_dir/xmlnsstack.cc",
 216     "$xmllite_dir/xmlnsstack.h",
 217     "$xmllite_dir/xmlparser.cc",
 218     "$xmllite_dir/xmlparser.h",
 219     "$xmllite_dir/xmlprinter.cc",
 220     "$xmllite_dir/xmlprinter.h",
 221     "$xmpp_dir/asyncsocket.h",
 222     "$xmpp_dir/constants.cc",
 223     "$xmpp_dir/constants.h",
 224     "$xmpp_dir/jid.cc",
 225     "$xmpp_dir/jid.h",
 226     "$xmpp_dir/plainsaslhandler.h",
 227     "$xmpp_dir/prexmppauth.h",
 228     "$xmpp_dir/saslcookiemechanism.h",
 229     "$xmpp_dir/saslhandler.h",
 230     "$xmpp_dir/saslmechanism.cc",
 231     "$xmpp_dir/saslmechanism.h",
 232     "$xmpp_dir/saslplainmechanism.h",
 233     "$xmpp_dir/xmppclient.cc",
 234     "$xmpp_dir/xmppclient.h",
 235     "$xmpp_dir/xmppclientsettings.h",
 236     "$xmpp_dir/xmppengine.h",
 237     "$xmpp_dir/xmppengineimpl.cc",
 238     "$xmpp_dir/xmppengineimpl.h",
 239     "$xmpp_dir/xmppengineimpl_iq.cc",
 240     "$xmpp_dir/xmpplogintask.cc",
 241     "$xmpp_dir/xmpplogintask.h",
 242     "$xmpp_dir/xmppstanzaparser.cc",
 243     "$xmpp_dir/xmppstanzaparser.h",
 244     "$xmpp_dir/xmpptask.cc",
 245     "$xmpp_dir/xmpptask.h",
 246   ]
 247   sources -= [
 248     # Compiled as part of libjingle_p2p_constants.
 249     "$p2p_dir/base/constants.cc",
 250     "$p2p_dir/base/constants.h",
 251   ]
 252
 253   # TODO(jschuh): crbug.com/167187 fix size_t to int truncations.
 254   configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
 255
 256   public_deps = [
 257     ":jingle_deps",
 258   ]
 259   deps = [
 260     "//third_party/webrtc/base:rtc_base",
 261     ":libjingle_p2p_constants",
 262   ]
 263
 264   # From libjingle_common.gypi's conditions list.
 265   if (is_win) {
 266     cflags = [ "/wd4005" ]
 267   }
 268
 269   configs += [ ":jingle_unexported_configs" ]
 270   public_configs = [ ":jingle_public_configs" ]
 271   all_dependent_configs = [ ":jingle_all_dependent_configs" ]
 272 }
 273
 274 # This has to be is a separate project due to a bug in MSVS 2008 and the
 275 # current toolset on android.  The problem is that we have two files named
 276 # "constants.cc" and MSVS/android doesn't handle this properly.
 277 # GYP currently has guards to catch this, so if you want to remove it,
 278 # run GYP and if GYP has removed the validation check, then we can assume
 279 # that the toolchains have been fixed (we currently use VS2010 and later,
 280 # so VS2008 isn't a concern anymore).
 281 #
 282 # GYP version: third_party/libjingle.gyp:libjingle_p2p_constants
 283 static_library("libjingle_p2p_constants") {
 284   p2p_dir = "../webrtc/p2p"
 285   sources = [
 286     "$p2p_dir/base/constants.cc",
 287     "$p2p_dir/base/constants.h",
 288   ]
 289   public_deps = [
 290     ":jingle_deps",
 291   ]
 292   configs += [ ":jingle_unexported_configs" ]
 293   public_configs = [ ":jingle_public_configs" ]
 294   all_dependent_configs = [ ":jingle_all_dependent_configs" ]
 295 }
 296
 297 if (enable_webrtc) {
 298   source_set("libjingle_webrtc") {
 299     sources = [
 300       "overrides/init_webrtc.cc",
 301       "overrides/init_webrtc.h",
 302     ]
 303     configs += [ ":jingle_unexported_configs" ]
 304     public_configs = [ ":jingle_public_configs" ]
 305     deps = [
 306       ":libjingle_webrtc_common",
 307     ]
 308   }
 309
 310   # Note: this does not support the shared library build of libpeerconnection
 311   # as is supported in the GYP build. It's not clear what this is used for.
 312   source_set("libjingle_webrtc_common") {
 313     sources = [
 314       "overrides/talk/media/webrtc/webrtcexport.h",
 315       "source/talk/app/webrtc/audiotrack.cc",
 316       "source/talk/app/webrtc/audiotrack.h",
 317       "source/talk/app/webrtc/audiotrackrenderer.cc",
 318       "source/talk/app/webrtc/audiotrackrenderer.h",
 319       "source/talk/app/webrtc/datachannel.cc",
 320       "source/talk/app/webrtc/datachannel.h",
 321       "source/talk/app/webrtc/dtlsidentityservice.cc",
 322       "source/talk/app/webrtc/dtlsidentityservice.h",
 323       "source/talk/app/webrtc/dtlsidentitystore.cc",
 324       "source/talk/app/webrtc/dtlsidentitystore.h",
 325       "source/talk/app/webrtc/dtmfsender.cc",
 326       "source/talk/app/webrtc/dtmfsender.h",
 327       "source/talk/app/webrtc/jsep.h",
 328       "source/talk/app/webrtc/jsepicecandidate.cc",
 329       "source/talk/app/webrtc/jsepicecandidate.h",
 330       "source/talk/app/webrtc/jsepsessiondescription.cc",
 331       "source/talk/app/webrtc/jsepsessiondescription.h",
 332       "source/talk/app/webrtc/localaudiosource.cc",
 333       "source/talk/app/webrtc/localaudiosource.h",
 334       "source/talk/app/webrtc/mediaconstraintsinterface.cc",
 335       "source/talk/app/webrtc/mediaconstraintsinterface.h",
 336       "source/talk/app/webrtc/mediastream.cc",
 337       "source/talk/app/webrtc/mediastream.h",
 338       "source/talk/app/webrtc/mediastreamhandler.cc",
 339       "source/talk/app/webrtc/mediastreamhandler.h",
 340       "source/talk/app/webrtc/mediastreaminterface.h",
 341       "source/talk/app/webrtc/mediastreamprovider.h",
 342       "source/talk/app/webrtc/mediastreamproxy.h",
 343       "source/talk/app/webrtc/mediastreamsignaling.cc",
 344       "source/talk/app/webrtc/mediastreamsignaling.h",
 345       "source/talk/app/webrtc/mediastreamtrack.h",
 346       "source/talk/app/webrtc/mediastreamtrackproxy.h",
 347       "source/talk/app/webrtc/notifier.h",
 348       "source/talk/app/webrtc/peerconnection.cc",
 349       "source/talk/app/webrtc/peerconnection.h",
 350       "source/talk/app/webrtc/peerconnectionfactory.cc",
 351       "source/talk/app/webrtc/peerconnectionfactory.h",
 352       "source/talk/app/webrtc/peerconnectioninterface.h",
 353       "source/talk/app/webrtc/portallocatorfactory.cc",
 354       "source/talk/app/webrtc/portallocatorfactory.h",
 355       "source/talk/app/webrtc/remoteaudiosource.cc",
 356       "source/talk/app/webrtc/remoteaudiosource.h",
 357       "source/talk/app/webrtc/remotevideocapturer.cc",
 358       "source/talk/app/webrtc/remotevideocapturer.h",
 359       "source/talk/app/webrtc/sctputils.cc",
 360       "source/talk/app/webrtc/sctputils.h",
 361       "source/talk/app/webrtc/statscollector.cc",
 362       "source/talk/app/webrtc/statscollector.h",
 363       "source/talk/app/webrtc/statstypes.cc",
 364       "source/talk/app/webrtc/statstypes.h",
 365       "source/talk/app/webrtc/streamcollection.h",
 366       "source/talk/app/webrtc/umametrics.h",
 367       "source/talk/app/webrtc/videosource.cc",
 368       "source/talk/app/webrtc/videosource.h",
 369       "source/talk/app/webrtc/videosourceinterface.h",
 370       "source/talk/app/webrtc/videosourceproxy.h",
 371       "source/talk/app/webrtc/videotrack.cc",
 372       "source/talk/app/webrtc/videotrack.h",
 373       "source/talk/app/webrtc/videotrackrenderers.cc",
 374       "source/talk/app/webrtc/videotrackrenderers.h",
 375       "source/talk/app/webrtc/webrtcsdp.cc",
 376       "source/talk/app/webrtc/webrtcsdp.h",
 377       "source/talk/app/webrtc/webrtcsession.cc",
 378       "source/talk/app/webrtc/webrtcsession.h",
 379       "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc",
 380       "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h",
 381       "source/talk/media/base/audiorenderer.h",
 382       "source/talk/media/base/capturemanager.cc",
 383       "source/talk/media/base/capturemanager.h",
 384       "source/talk/media/base/capturerenderadapter.cc",
 385       "source/talk/media/base/capturerenderadapter.h",
 386       "source/talk/media/base/codec.cc",
 387       "source/talk/media/base/codec.h",
 388       "source/talk/media/base/constants.cc",
 389       "source/talk/media/base/constants.h",
 390       "source/talk/media/base/cryptoparams.h",
 391       "source/talk/media/base/hybriddataengine.h",
 392       "source/talk/media/base/mediachannel.h",
 393       "source/talk/media/base/mediaengine.cc",
 394       "source/talk/media/base/mediaengine.h",
 395       "source/talk/media/base/rtpdataengine.cc",
 396       "source/talk/media/base/rtpdataengine.h",
 397       "source/talk/media/base/rtpdump.cc",
 398       "source/talk/media/base/rtpdump.h",
 399       "source/talk/media/base/rtputils.cc",
 400       "source/talk/media/base/rtputils.h",
 401       "source/talk/media/base/streamparams.cc",
 402       "source/talk/media/base/streamparams.h",
 403       "source/talk/media/base/videoadapter.cc",
 404       "source/talk/media/base/videoadapter.h",
 405       "source/talk/media/base/videocapturer.cc",
 406       "source/talk/media/base/videocapturer.h",
 407       "source/talk/media/base/videocommon.cc",
 408       "source/talk/media/base/videocommon.h",
 409       "source/talk/media/base/videoframe.cc",
 410       "source/talk/media/base/videoframe.h",
 411       "source/talk/media/base/videoframefactory.cc",
 412       "source/talk/media/base/videoframefactory.h",
 413       "source/talk/media/devices/dummydevicemanager.cc",
 414       "source/talk/media/devices/dummydevicemanager.h",
 415       "source/talk/media/devices/filevideocapturer.cc",
 416       "source/talk/media/devices/filevideocapturer.h",
 417       "source/talk/media/webrtc/webrtccommon.h",
 418       "source/talk/media/webrtc/webrtcpassthroughrender.cc",
 419       "source/talk/media/webrtc/webrtcpassthroughrender.h",
 420       "source/talk/media/webrtc/webrtcvideoframe.cc",
 421       "source/talk/media/webrtc/webrtcvideoframe.h",
 422       "source/talk/media/webrtc/webrtcvideoframefactory.cc",
 423       "source/talk/media/webrtc/webrtcvideoframefactory.h",
 424       "source/talk/media/webrtc/webrtcvoe.h",
 425       "source/talk/session/media/audiomonitor.cc",
 426       "source/talk/session/media/audiomonitor.h",
 427       "source/talk/session/media/bundlefilter.cc",
 428       "source/talk/session/media/bundlefilter.h",
 429       "source/talk/session/media/channel.cc",
 430       "source/talk/session/media/channel.h",
 431       "source/talk/session/media/channelmanager.cc",
 432       "source/talk/session/media/channelmanager.h",
 433       "source/talk/session/media/currentspeakermonitor.cc",
 434       "source/talk/session/media/currentspeakermonitor.h",
 435       "source/talk/session/media/externalhmac.cc",
 436       "source/talk/session/media/externalhmac.h",
 437       "source/talk/session/media/mediamonitor.cc",
 438       "source/talk/session/media/mediamonitor.h",
 439       "source/talk/session/media/mediasession.cc",
 440       "source/talk/session/media/mediasession.h",
 441       "source/talk/session/media/mediasink.h",
 442       "source/talk/session/media/rtcpmuxfilter.cc",
 443       "source/talk/session/media/rtcpmuxfilter.h",
 444       "source/talk/session/media/soundclip.cc",
 445       "source/talk/session/media/soundclip.h",
 446       "source/talk/session/media/srtpfilter.cc",
 447       "source/talk/session/media/srtpfilter.h",
 448       "source/talk/session/media/typingmonitor.cc",
 449       "source/talk/session/media/typingmonitor.h",
 450       "source/talk/session/media/voicechannel.h",
 451     ]
 452
 453     configs -= [ "//build/config/compiler:chromium_code" ]
 454     configs += [ "//build/config/compiler:no_chromium_code" ]
 455
 456     configs += [ ":jingle_unexported_configs" ]
 457     public_configs = [ ":jingle_public_configs" ]
 458
 459     deps = [
 460       ":libjingle",
 461       "//third_party/libsrtp",
 462       "//third_party/webrtc/modules/media_file",
 463       "//third_party/webrtc/modules/video_capture",
 464       "//third_party/webrtc/modules/video_render",
 465     ]
 466
 467     if (!is_ios) {
 468       # TODO(mallinath) - Enable SCTP for iOS.
 469       sources += [
 470         "source/talk/media/sctp/sctpdataengine.cc",
 471         "source/talk/media/sctp/sctpdataengine.h",
 472       ]
 473       defines = [ "HAVE_SCTP" ]
 474       deps += [ "//third_party/usrsctp" ]
 475     }
 476   }
 477
 478   # Note: this does not support the shared library build of libpeerconnection
 479   # as is supported in the GYP build. It's not clear what this is used for.
 480   source_set("libpeerconnection") {
 481     sources = [
 482       "source/talk/media/webrtc/simulcast.cc",
 483       "source/talk/media/webrtc/simulcast.h",
 484       "source/talk/media/webrtc/webrtcmediaengine.cc",
 485       "source/talk/media/webrtc/webrtcmediaengine.h",
 486       "source/talk/media/webrtc/webrtcvideoengine2.cc",
 487       "source/talk/media/webrtc/webrtcvideoengine2.h",
 488       "source/talk/media/webrtc/webrtcvoiceengine.cc",
 489       "source/talk/media/webrtc/webrtcvoiceengine.h",
 490     ]
 491
 492     configs += [ ":jingle_unexported_configs" ]
 493     public_configs = [ ":jingle_public_configs" ]
 494     configs -= [ "//build/config/compiler:chromium_code" ]
 495     configs += [ "//build/config/compiler:no_chromium_code" ]
 496
 497     deps = [
 498       # TODO(GYP): crbug.com/481633. Consider depending on :libjingle_webrtc
 499       # instead.
 500       ":libjingle_webrtc_common",
 501       "//third_party/webrtc",
 502       "//third_party/webrtc/system_wrappers",
 503       "//third_party/webrtc/voice_engine",
 504     ]
 505   }
 506
 507   source_set("libstunprober") {
 508     p2p_dir = "../webrtc/p2p"
 509     sources = [
 510       "$p2p_dir/stunprober/stunprober.cc",
 511     ]
 512
 513     deps = [
 514       "//third_party/webrtc/base:rtc_base",
 515       ":libjingle_webrtc_common",
 516     ]
 517   }
 518 }  # enable_webrtc
 519 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.