| # 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 = [ |
| "overrides/talk/media/webrtc/webrtcexport.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/media/base/audiorenderer.h", |
| "source/talk/media/base/capturemanager.cc", |
| "source/talk/media/base/capturemanager.h", |
| "source/talk/media/base/capturerenderadapter.cc", |
| "source/talk/media/base/capturerenderadapter.h", |
| "source/talk/media/base/codec.cc", |
| "source/talk/media/base/codec.h", |
| "source/talk/media/base/constants.cc", |
| "source/talk/media/base/constants.h", |
| "source/talk/media/base/cryptoparams.h", |
| "source/talk/media/base/hybriddataengine.h", |
| "source/talk/media/base/mediachannel.h", |
| "source/talk/media/base/mediaengine.cc", |
| "source/talk/media/base/mediaengine.h", |
| "source/talk/media/base/rtpdataengine.cc", |
| "source/talk/media/base/rtpdataengine.h", |
| "source/talk/media/base/rtpdump.cc", |
| "source/talk/media/base/rtpdump.h", |
| "source/talk/media/base/rtputils.cc", |
| "source/talk/media/base/rtputils.h", |
| "source/talk/media/base/streamparams.cc", |
| "source/talk/media/base/streamparams.h", |
| "source/talk/media/base/videoadapter.cc", |
| "source/talk/media/base/videoadapter.h", |
| "source/talk/media/base/videocapturer.cc", |
| "source/talk/media/base/videocapturer.h", |
| "source/talk/media/base/videocommon.cc", |
| "source/talk/media/base/videocommon.h", |
| "source/talk/media/base/videoframe.cc", |
| "source/talk/media/base/videoframe.h", |
| "source/talk/media/base/videoframefactory.cc", |
| "source/talk/media/base/videoframefactory.h", |
| "source/talk/media/devices/dummydevicemanager.cc", |
| "source/talk/media/devices/dummydevicemanager.h", |
| "source/talk/media/devices/filevideocapturer.cc", |
| "source/talk/media/devices/filevideocapturer.h", |
| "source/talk/media/webrtc/webrtccommon.h", |
| "source/talk/media/webrtc/webrtcvideoframe.cc", |
| "source/talk/media/webrtc/webrtcvideoframe.h", |
| "source/talk/media/webrtc/webrtcvideoframefactory.cc", |
| "source/talk/media/webrtc/webrtcvideoframefactory.h", |
| "source/talk/media/webrtc/webrtcvoe.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 += [ |
| "source/talk/media/sctp/sctpdataengine.cc", |
| "source/talk/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 = [ |
| "source/talk/media/webrtc/simulcast.cc", |
| "source/talk/media/webrtc/simulcast.h", |
| "source/talk/media/webrtc/webrtcmediaengine.cc", |
| "source/talk/media/webrtc/webrtcmediaengine.h", |
| "source/talk/media/webrtc/webrtcvideoengine2.cc", |
| "source/talk/media/webrtc/webrtcvideoengine2.h", |
| "source/talk/media/webrtc/webrtcvoiceengine.cc", |
| "source/talk/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. |
