# Copyright 2014 The Chromium Authors. All rights reserved. # Use of this source code is governed by a BSD-style license that can be # found in the LICENSE file. import("//build/config/features.gni") # From third_party/libjingle/libjingle.gyp's target_defaults. config("jingle_unexported_configs") { defines = [ "EXPAT_RELATIVE_PATH", "FEATURE_ENABLE_SSL", "GTEST_RELATIVE_PATH", "HAVE_OPENSSL_SSL_H", "HAVE_SRTP", "HAVE_WEBRTC_VIDEO", "HAVE_WEBRTC_VOICE", "LOGGING_INSIDE_WEBRTC", "NO_MAIN_THREAD_WRAPPING", "NO_SOUND_SYSTEM", "SRTP_RELATIVE_PATH", "SSL_USE_OPENSSL", "USE_WEBRTC_DEV_BRANCH", "ENABLE_EXTERNAL_AUTH", "WEBRTC_CHROMIUM_BUILD", ] include_dirs = [ "overrides", "../../third_party/webrtc_overrides", "source", "../../testing/gtest/include", "../../third_party", "../../third_party/libyuv/include", "../../third_party/usrsctp/usrsctplib", ] if (is_win && current_cpu == "x86") { defines += [ "_USE_32BIT_TIME_T" ] } } # From third_party/libjingle/libjingle.gyp's target_defaults. config("jingle_public_configs") { include_dirs = [ "../../third_party/webrtc_overrides", "overrides", "source", "../../testing/gtest/include", "../../third_party", ] defines = [ "FEATURE_ENABLE_SSL", "FEATURE_ENABLE_VOICEMAIL", "EXPAT_RELATIVE_PATH", "GTEST_RELATIVE_PATH", "NO_MAIN_THREAD_WRAPPING", "NO_SOUND_SYSTEM", ] # TODO(GYP): Port is_win blocks. if (is_linux) { defines += [ "LINUX", "WEBRTC_LINUX", ] } if (is_mac) { defines += [ "OSX", "WEBRTC_MAC", ] } if (is_ios) { defines += [ "IOS", "WEBRTC_MAC", "WEBRTC_IOS", ] } if (is_win) { defines += [ "WEBRTC_WIN" ] } if (is_android) { defines += [ "ANDROID" ] } if (is_posix) { defines += [ "WEBRTC_POSIX" ] } # TODO(GYP): Support these in GN. # if (is_bsd) { # defines += [ "BSD" ] # } # if (is_openbsd) { # defines += [ "OPENBSD" ] # } # if (is_freebsd) { # defines += [ "FREEBSD" ] # } if (is_chromeos) { defines += [ "CHROMEOS" ] } } # From third_party/libjingle/libjingle.gyp's target_defaults. group("jingle_deps") { public_deps = [ "//third_party/expat", ] deps = [ "//base", "//crypto:platform", "//net", ] } # GYP version: third_party/libjingle.gyp:libjingle static_library("libjingle") { p2p_dir = "../webrtc/p2p" xmllite_dir = "../webrtc/libjingle/xmllite" xmpp_dir = "../webrtc/libjingle/xmpp" sources = [ # List from third_party/libjingle/libjingle_common.gypi "$p2p_dir/base/asyncstuntcpsocket.cc", "$p2p_dir/base/asyncstuntcpsocket.h", "$p2p_dir/base/basicpacketsocketfactory.cc", "$p2p_dir/base/basicpacketsocketfactory.h", "$p2p_dir/base/candidate.h", "$p2p_dir/base/common.h", "$p2p_dir/base/dtlstransport.h", "$p2p_dir/base/dtlstransportchannel.cc", "$p2p_dir/base/dtlstransportchannel.h", "$p2p_dir/base/p2pconstants.cc", "$p2p_dir/base/p2pconstants.h", "$p2p_dir/base/p2ptransport.cc", "$p2p_dir/base/p2ptransport.h", "$p2p_dir/base/p2ptransportchannel.cc", "$p2p_dir/base/p2ptransportchannel.h", "$p2p_dir/base/port.cc", "$p2p_dir/base/port.h", "$p2p_dir/base/portallocator.cc", "$p2p_dir/base/portallocator.h", "$p2p_dir/base/pseudotcp.cc", "$p2p_dir/base/pseudotcp.h", "$p2p_dir/base/rawtransport.cc", "$p2p_dir/base/rawtransport.h", "$p2p_dir/base/rawtransportchannel.cc", "$p2p_dir/base/rawtransportchannel.h", "$p2p_dir/base/relayport.cc", "$p2p_dir/base/relayport.h", "$p2p_dir/base/session.cc", "$p2p_dir/base/session.h", "$p2p_dir/base/sessiondescription.cc", "$p2p_dir/base/sessiondescription.h", "$p2p_dir/base/sessionid.h", "$p2p_dir/base/stun.cc", "$p2p_dir/base/stun.h", "$p2p_dir/base/stunport.cc", "$p2p_dir/base/stunport.h", "$p2p_dir/base/stunrequest.cc", "$p2p_dir/base/stunrequest.h", "$p2p_dir/base/tcpport.cc", "$p2p_dir/base/tcpport.h", "$p2p_dir/base/transport.cc", "$p2p_dir/base/transport.h", "$p2p_dir/base/transportchannel.cc", "$p2p_dir/base/transportchannel.h", "$p2p_dir/base/transportchannelimpl.h", "$p2p_dir/base/transportcontroller.cc", "$p2p_dir/base/transportcontroller.h", "$p2p_dir/base/transportdescription.cc", "$p2p_dir/base/transportdescription.h", "$p2p_dir/base/transportdescriptionfactory.cc", "$p2p_dir/base/transportdescriptionfactory.h", "$p2p_dir/base/turnport.cc", "$p2p_dir/base/turnport.h", "$p2p_dir/client/basicportallocator.cc", "$p2p_dir/client/basicportallocator.h", "$p2p_dir/client/httpportallocator.cc", "$p2p_dir/client/httpportallocator.h", "$p2p_dir/client/socketmonitor.cc", "$p2p_dir/client/socketmonitor.h", "$xmllite_dir/qname.cc", "$xmllite_dir/qname.h", "$xmllite_dir/xmlbuilder.cc", "$xmllite_dir/xmlbuilder.h", "$xmllite_dir/xmlconstants.cc", "$xmllite_dir/xmlconstants.h", "$xmllite_dir/xmlelement.cc", "$xmllite_dir/xmlelement.h", "$xmllite_dir/xmlnsstack.cc", "$xmllite_dir/xmlnsstack.h", "$xmllite_dir/xmlparser.cc", "$xmllite_dir/xmlparser.h", "$xmllite_dir/xmlprinter.cc", "$xmllite_dir/xmlprinter.h", "$xmpp_dir/asyncsocket.h", "$xmpp_dir/constants.cc", "$xmpp_dir/constants.h", "$xmpp_dir/jid.cc", "$xmpp_dir/jid.h", "$xmpp_dir/plainsaslhandler.h", "$xmpp_dir/prexmppauth.h", "$xmpp_dir/saslcookiemechanism.h", "$xmpp_dir/saslhandler.h", "$xmpp_dir/saslmechanism.cc", "$xmpp_dir/saslmechanism.h", "$xmpp_dir/saslplainmechanism.h", "$xmpp_dir/xmppclient.cc", "$xmpp_dir/xmppclient.h", "$xmpp_dir/xmppclientsettings.h", "$xmpp_dir/xmppengine.h", "$xmpp_dir/xmppengineimpl.cc", "$xmpp_dir/xmppengineimpl.h", "$xmpp_dir/xmppengineimpl_iq.cc", "$xmpp_dir/xmpplogintask.cc", "$xmpp_dir/xmpplogintask.h", "$xmpp_dir/xmppstanzaparser.cc", "$xmpp_dir/xmppstanzaparser.h", "$xmpp_dir/xmpptask.cc", "$xmpp_dir/xmpptask.h", ] # TODO(jschuh): crbug.com/167187 fix size_t to int truncations. configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] public_deps = [ ":jingle_deps", ] deps = [ "//third_party/webrtc/base:rtc_base", ] # From libjingle_common.gypi's conditions list. if (is_win) { cflags = [ "/wd4005" ] } if (is_nacl) { # For NACL, we have to add a default implementation for field_trail. deps += [ "//native_client_sdk/src/libraries/nacl_io", "//third_party/webrtc/system_wrappers:field_trial_default", ] } else { # Otherwise, we just add the field_trial which redirects to base. sources += [ "overrides/field_trial.cc" ] } configs += [ ":jingle_unexported_configs" ] public_configs = [ ":jingle_public_configs" ] } if (enable_webrtc) { source_set("libjingle_webrtc") { sources = [ "overrides/init_webrtc.cc", "overrides/init_webrtc.h", ] configs += [ ":jingle_unexported_configs" ] public_configs = [ ":jingle_public_configs" ] public_deps = [ ":libjingle_webrtc_common", ] } source_set("libjingle_webrtc_common") { sources = [ "../webrtc/api/audiotrack.cc", "../webrtc/api/audiotrack.h", "../webrtc/api/datachannel.cc", "../webrtc/api/datachannel.h", "../webrtc/api/dtlsidentitystore.cc", "../webrtc/api/dtlsidentitystore.h", "../webrtc/api/dtmfsender.cc", "../webrtc/api/dtmfsender.h", "../webrtc/api/jsep.h", "../webrtc/api/jsepicecandidate.cc", "../webrtc/api/jsepicecandidate.h", "../webrtc/api/jsepsessiondescription.cc", "../webrtc/api/jsepsessiondescription.h", "../webrtc/api/localaudiosource.cc", "../webrtc/api/localaudiosource.h", "../webrtc/api/mediaconstraintsinterface.cc", "../webrtc/api/mediaconstraintsinterface.h", "../webrtc/api/mediacontroller.cc", "../webrtc/api/mediacontroller.h", "../webrtc/api/mediastream.cc", "../webrtc/api/mediastream.h", "../webrtc/api/mediastreamhandler.cc", "../webrtc/api/mediastreamhandler.h", "../webrtc/api/mediastreaminterface.h", "../webrtc/api/mediastreamobserver.cc", "../webrtc/api/mediastreamobserver.h", "../webrtc/api/mediastreamprovider.h", "../webrtc/api/mediastreamproxy.h", "../webrtc/api/mediastreamtrack.h", "../webrtc/api/mediastreamtrackproxy.h", "../webrtc/api/notifier.h", "../webrtc/api/peerconnection.cc", "../webrtc/api/peerconnection.h", "../webrtc/api/peerconnectionfactory.cc", "../webrtc/api/peerconnectionfactory.h", "../webrtc/api/peerconnectioninterface.h", "../webrtc/api/portallocatorfactory.cc", "../webrtc/api/portallocatorfactory.h", "../webrtc/api/remoteaudiosource.cc", "../webrtc/api/remoteaudiosource.h", "../webrtc/api/remoteaudiotrack.cc", "../webrtc/api/remoteaudiotrack.h", "../webrtc/api/rtpreceiver.cc", "../webrtc/api/rtpreceiver.h", "../webrtc/api/rtpreceiverinterface.h", "../webrtc/api/rtpsender.cc", "../webrtc/api/rtpsender.h", "../webrtc/api/rtpsenderinterface.h", "../webrtc/api/sctputils.cc", "../webrtc/api/sctputils.h", "../webrtc/api/statscollector.cc", "../webrtc/api/statscollector.h", "../webrtc/api/statstypes.cc", "../webrtc/api/statstypes.h", "../webrtc/api/streamcollection.h", "../webrtc/api/umametrics.h", "../webrtc/api/videocapturertracksource.cc", "../webrtc/api/videocapturertracksource.h", "../webrtc/api/videosourceproxy.h", "../webrtc/api/videotrack.cc", "../webrtc/api/videotrack.h", "../webrtc/api/videotracksource.cc", "../webrtc/api/videotracksource.h", "../webrtc/api/webrtcsdp.cc", "../webrtc/api/webrtcsdp.h", "../webrtc/api/webrtcsession.cc", "../webrtc/api/webrtcsession.h", "../webrtc/api/webrtcsessiondescriptionfactory.cc", "../webrtc/api/webrtcsessiondescriptionfactory.h", "../webrtc/media/base/audiorenderer.h", "../webrtc/media/base/codec.cc", "../webrtc/media/base/codec.h", "../webrtc/media/base/cryptoparams.h", "../webrtc/media/base/hybriddataengine.h", "../webrtc/media/base/mediachannel.h", "../webrtc/media/base/mediaconstants.cc", "../webrtc/media/base/mediaconstants.h", "../webrtc/media/base/mediaengine.cc", "../webrtc/media/base/mediaengine.h", "../webrtc/media/base/rtpdataengine.cc", "../webrtc/media/base/rtpdataengine.h", "../webrtc/media/base/rtpdump.cc", "../webrtc/media/base/rtpdump.h", "../webrtc/media/base/rtputils.cc", "../webrtc/media/base/rtputils.h", "../webrtc/media/base/streamparams.cc", "../webrtc/media/base/streamparams.h", "../webrtc/media/base/turnutils.cc", "../webrtc/media/base/turnutils.h", "../webrtc/media/base/videoadapter.cc", "../webrtc/media/base/videoadapter.h", "../webrtc/media/base/videobroadcaster.cc", "../webrtc/media/base/videobroadcaster.h", "../webrtc/media/base/videocapturer.cc", "../webrtc/media/base/videocapturer.h", "../webrtc/media/base/videocommon.cc", "../webrtc/media/base/videocommon.h", "../webrtc/media/base/videoframe.cc", "../webrtc/media/base/videoframe.h", "../webrtc/media/base/videoframefactory.cc", "../webrtc/media/base/videoframefactory.h", "../webrtc/media/base/videosourcebase.cc", "../webrtc/media/base/videosourcebase.h", "../webrtc/media/engine/webrtccommon.h", "../webrtc/media/engine/webrtcvideoframe.cc", "../webrtc/media/engine/webrtcvideoframe.h", "../webrtc/media/engine/webrtcvideoframefactory.cc", "../webrtc/media/engine/webrtcvideoframefactory.h", "../webrtc/media/engine/webrtcvoe.h", "../webrtc/pc/audiomonitor.cc", "../webrtc/pc/audiomonitor.h", "../webrtc/pc/bundlefilter.cc", "../webrtc/pc/bundlefilter.h", "../webrtc/pc/channel.cc", "../webrtc/pc/channel.h", "../webrtc/pc/channelmanager.cc", "../webrtc/pc/channelmanager.h", "../webrtc/pc/currentspeakermonitor.cc", "../webrtc/pc/currentspeakermonitor.h", "../webrtc/pc/externalhmac.cc", "../webrtc/pc/externalhmac.h", "../webrtc/pc/mediamonitor.cc", "../webrtc/pc/mediamonitor.h", "../webrtc/pc/mediasession.cc", "../webrtc/pc/mediasession.h", "../webrtc/pc/mediasink.h", "../webrtc/pc/rtcpmuxfilter.cc", "../webrtc/pc/rtcpmuxfilter.h", "../webrtc/pc/srtpfilter.cc", "../webrtc/pc/srtpfilter.h", "../webrtc/pc/voicechannel.h", ] configs -= [ "//build/config/compiler:chromium_code" ] configs += [ "//build/config/compiler:no_chromium_code" ] configs += [ ":jingle_unexported_configs" ] public_configs = [ ":jingle_public_configs" ] deps = [ ":libjingle", "//third_party/libsrtp", "//third_party/webrtc/modules/media_file", "//third_party/webrtc/modules/video_capture", "//third_party/webrtc/modules/video_render", ] if (!is_ios) { # TODO(mallinath) - Enable SCTP for iOS. sources += [ "../webrtc/media/sctp/sctpdataengine.cc", "../webrtc/media/sctp/sctpdataengine.h", ] defines = [ "HAVE_SCTP" ] deps += [ "//third_party/usrsctp" ] } } source_set("libpeerconnection") { sources = [ "../webrtc/media/engine/simulcast.cc", "../webrtc/media/engine/simulcast.h", "../webrtc/media/engine/webrtcmediaengine.cc", "../webrtc/media/engine/webrtcmediaengine.h", "../webrtc/media/engine/webrtcvideoengine2.cc", "../webrtc/media/engine/webrtcvideoengine2.h", "../webrtc/media/engine/webrtcvoiceengine.cc", "../webrtc/media/engine/webrtcvoiceengine.h", ] configs += [ ":jingle_unexported_configs" ] public_configs = [ ":jingle_public_configs" ] configs -= [ "//build/config/compiler:chromium_code" ] configs += [ "//build/config/compiler:no_chromium_code" ] deps = [ # TODO(GYP): crbug.com/481633. Consider depending on :libjingle_webrtc # instead. ":libjingle_webrtc_common", "//third_party/webrtc", "//third_party/webrtc/system_wrappers", "//third_party/webrtc/voice_engine", ] } source_set("libstunprober") { p2p_dir = "../webrtc/p2p" sources = [ "$p2p_dir/stunprober/stunprober.cc", ] deps = [ ":libjingle_webrtc_common", "//third_party/webrtc/base:rtc_base", ] } } # enable_webrtc # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block.