# 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", "$p2p_dir/base/constants.cc", "$p2p_dir/base/constants.h", "$p2p_dir/base/dtlstransport.h", "$p2p_dir/base/dtlstransportchannel.cc", "$p2p_dir/base/dtlstransportchannel.h", "$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", ] sources -= [ # Compiled as part of libjingle_p2p_constants. "$p2p_dir/base/constants.cc", "$p2p_dir/base/constants.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 = [ ":libjingle_p2p_constants", "//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" ] } # This has to be is a separate project due to a bug in MSVS 2008 and the # current toolset on android. The problem is that we have two files named # "constants.cc" and MSVS/android doesn't handle this properly. # GYP currently has guards to catch this, so if you want to remove it, # run GYP and if GYP has removed the validation check, then we can assume # that the toolchains have been fixed (we currently use VS2010 and later, # so VS2008 isn't a concern anymore). # # GYP version: third_party/libjingle.gyp:libjingle_p2p_constants static_library("libjingle_p2p_constants") { p2p_dir = "../webrtc/p2p" sources = [ "$p2p_dir/base/constants.cc", "$p2p_dir/base/constants.h", ] public_deps = [ ":jingle_deps", ] 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", ] } # Note: this does not support the shared library build of libpeerconnection # as is supported in the GYP build. It's not clear what this is used for. source_set("libjingle_webrtc_common") { sources = [ "../webrtc/media/base/audiorenderer.h", "../webrtc/media/base/capturemanager.cc", "../webrtc/media/base/capturemanager.h", "../webrtc/media/base/capturerenderadapter.cc", "../webrtc/media/base/capturerenderadapter.h", "../webrtc/media/base/codec.cc", "../webrtc/media/base/codec.h", "../webrtc/media/base/constants.cc", "../webrtc/media/base/constants.h", "../webrtc/media/base/cryptoparams.h", "../webrtc/media/base/hybriddataengine.h", "../webrtc/media/base/mediachannel.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/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/devices/dummydevicemanager.cc", "../webrtc/media/devices/dummydevicemanager.h", "../webrtc/media/devices/filevideocapturer.cc", "../webrtc/media/devices/filevideocapturer.h", "../webrtc/media/webrtc/webrtccommon.h", "../webrtc/media/webrtc/webrtcvideoframe.cc", "../webrtc/media/webrtc/webrtcvideoframe.h", "../webrtc/media/webrtc/webrtcvideoframefactory.cc", "../webrtc/media/webrtc/webrtcvideoframefactory.h", "../webrtc/media/webrtc/webrtcvoe.h", "source/talk/app/webrtc/audiotrack.cc", "source/talk/app/webrtc/audiotrack.h", "source/talk/app/webrtc/datachannel.cc", "source/talk/app/webrtc/datachannel.h", "source/talk/app/webrtc/dtlsidentitystore.cc", "source/talk/app/webrtc/dtlsidentitystore.h", "source/talk/app/webrtc/dtmfsender.cc", "source/talk/app/webrtc/dtmfsender.h", "source/talk/app/webrtc/jsep.h", "source/talk/app/webrtc/jsepicecandidate.cc", "source/talk/app/webrtc/jsepicecandidate.h", "source/talk/app/webrtc/jsepsessiondescription.cc", "source/talk/app/webrtc/jsepsessiondescription.h", "source/talk/app/webrtc/localaudiosource.cc", "source/talk/app/webrtc/localaudiosource.h", "source/talk/app/webrtc/mediaconstraintsinterface.cc", "source/talk/app/webrtc/mediaconstraintsinterface.h", "source/talk/app/webrtc/mediacontroller.cc", "source/talk/app/webrtc/mediacontroller.h", "source/talk/app/webrtc/mediastream.cc", "source/talk/app/webrtc/mediastream.h", "source/talk/app/webrtc/mediastreamhandler.cc", "source/talk/app/webrtc/mediastreamhandler.h", "source/talk/app/webrtc/mediastreaminterface.h", "source/talk/app/webrtc/mediastreamobserver.cc", "source/talk/app/webrtc/mediastreamobserver.h", "source/talk/app/webrtc/mediastreamprovider.h", "source/talk/app/webrtc/mediastreamproxy.h", "source/talk/app/webrtc/mediastreamtrack.h", "source/talk/app/webrtc/mediastreamtrackproxy.h", "source/talk/app/webrtc/notifier.h", "source/talk/app/webrtc/peerconnection.cc", "source/talk/app/webrtc/peerconnection.h", "source/talk/app/webrtc/peerconnectionfactory.cc", "source/talk/app/webrtc/peerconnectionfactory.h", "source/talk/app/webrtc/peerconnectioninterface.h", "source/talk/app/webrtc/portallocatorfactory.cc", "source/talk/app/webrtc/portallocatorfactory.h", "source/talk/app/webrtc/remoteaudiosource.cc", "source/talk/app/webrtc/remoteaudiosource.h", "source/talk/app/webrtc/remoteaudiotrack.cc", "source/talk/app/webrtc/remoteaudiotrack.h", "source/talk/app/webrtc/remotevideocapturer.cc", "source/talk/app/webrtc/remotevideocapturer.h", "source/talk/app/webrtc/rtpreceiver.cc", "source/talk/app/webrtc/rtpreceiver.h", "source/talk/app/webrtc/rtpreceiverinterface.h", "source/talk/app/webrtc/rtpsender.cc", "source/talk/app/webrtc/rtpsender.h", "source/talk/app/webrtc/rtpsenderinterface.h", "source/talk/app/webrtc/sctputils.cc", "source/talk/app/webrtc/sctputils.h", "source/talk/app/webrtc/statscollector.cc", "source/talk/app/webrtc/statscollector.h", "source/talk/app/webrtc/statstypes.cc", "source/talk/app/webrtc/statstypes.h", "source/talk/app/webrtc/streamcollection.h", "source/talk/app/webrtc/umametrics.h", "source/talk/app/webrtc/videosource.cc", "source/talk/app/webrtc/videosource.h", "source/talk/app/webrtc/videosourceinterface.h", "source/talk/app/webrtc/videosourceproxy.h", "source/talk/app/webrtc/videotrack.cc", "source/talk/app/webrtc/videotrack.h", "source/talk/app/webrtc/videotrackrenderers.cc", "source/talk/app/webrtc/videotrackrenderers.h", "source/talk/app/webrtc/webrtcsdp.cc", "source/talk/app/webrtc/webrtcsdp.h", "source/talk/app/webrtc/webrtcsession.cc", "source/talk/app/webrtc/webrtcsession.h", "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc", "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h", "source/talk/session/media/audiomonitor.cc", "source/talk/session/media/audiomonitor.h", "source/talk/session/media/bundlefilter.cc", "source/talk/session/media/bundlefilter.h", "source/talk/session/media/channel.cc", "source/talk/session/media/channel.h", "source/talk/session/media/channelmanager.cc", "source/talk/session/media/channelmanager.h", "source/talk/session/media/currentspeakermonitor.cc", "source/talk/session/media/currentspeakermonitor.h", "source/talk/session/media/externalhmac.cc", "source/talk/session/media/externalhmac.h", "source/talk/session/media/mediamonitor.cc", "source/talk/session/media/mediamonitor.h", "source/talk/session/media/mediasession.cc", "source/talk/session/media/mediasession.h", "source/talk/session/media/mediasink.h", "source/talk/session/media/rtcpmuxfilter.cc", "source/talk/session/media/rtcpmuxfilter.h", "source/talk/session/media/srtpfilter.cc", "source/talk/session/media/srtpfilter.h", "source/talk/session/media/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" ] } } # Note: this does not support the shared library build of libpeerconnection # as is supported in the GYP build. It's not clear what this is used for. source_set("libpeerconnection") { sources = [ "../webrtc/media/webrtc/simulcast.cc", "../webrtc/media/webrtc/simulcast.h", "../webrtc/media/webrtc/webrtcmediaengine.cc", "../webrtc/media/webrtc/webrtcmediaengine.h", "../webrtc/media/webrtc/webrtcvideoengine2.cc", "../webrtc/media/webrtc/webrtcvideoengine2.h", "../webrtc/media/webrtc/webrtcvoiceengine.cc", "../webrtc/media/webrtc/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.