| Index: third_party/libjingle/BUILD.gn
|
| diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn
|
| index 82db9c7404ea17a4f6bcbc91c2bd033592bb5612..4ec99353f4891e660862d005f6b16d9ddf833b50 100644
|
| --- a/third_party/libjingle/BUILD.gn
|
| +++ b/third_party/libjingle/BUILD.gn
|
| @@ -6,25 +6,6 @@ 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_SCTP",
|
| - "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 = [
|
| "../../third_party/webrtc_overrides",
|
| "../../testing/gtest/include",
|
| @@ -41,46 +22,6 @@ config("jingle_public_configs") {
|
| "../../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",
|
| - ]
|
| -
|
| - 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" ]
|
| - }
|
| - if (is_chromeos) {
|
| - defines += [ "CHROMEOS" ]
|
| - }
|
| }
|
|
|
| # From third_party/libjingle/libjingle.gyp's target_defaults.
|
| @@ -97,115 +38,19 @@ group("jingle_deps") {
|
|
|
| # 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/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",
|
| +
|
| + # TODO(kjellander): Start cleaning up this target as soon as
|
| + # https://codereview.chromium.org/2022833002/ is landed. The target should
|
| + # be removed entirely if possible.
|
| + "//third_party/webrtc/base:rtc_base",
|
| ]
|
| deps = [
|
| - "//third_party/webrtc/base:rtc_base",
|
| + "//third_party/webrtc/p2p:rtc_p2p",
|
| ]
|
|
|
| # From libjingle_common.gypi's conditions list.
|
| @@ -221,11 +66,19 @@ static_library("libjingle") {
|
| ]
|
| } else {
|
| # Otherwise, we just add the field_trial which redirects to base.
|
| - sources += [ "../webrtc_overrides/field_trial.cc" ]
|
| + sources = [
|
| + "../webrtc_overrides/field_trial.cc",
|
| + ]
|
| }
|
|
|
| - configs += [ ":jingle_unexported_configs" ]
|
| - public_configs = [ ":jingle_public_configs" ]
|
| + configs += [
|
| + ":jingle_unexported_configs",
|
| + "//third_party/webrtc:common_config",
|
| + ]
|
| + public_configs = [
|
| + ":jingle_public_configs",
|
| + "//third_party/webrtc:common_inherited_config",
|
| + ]
|
| }
|
|
|
| if (enable_webrtc) {
|
| @@ -237,162 +90,39 @@ if (enable_webrtc) {
|
| "../webrtc_overrides/init_webrtc.cc",
|
| "../webrtc_overrides/init_webrtc.h",
|
| ]
|
| - configs += [ ":jingle_unexported_configs" ]
|
| - public_configs = [ ":jingle_public_configs" ]
|
| + configs += [
|
| + ":jingle_unexported_configs",
|
| + "//third_party/webrtc:common_config",
|
| + ]
|
| + public_configs = [
|
| + ":jingle_public_configs",
|
| + "//third_party/webrtc:common_inherited_config",
|
| + ]
|
| public_deps = [
|
| ":libjingle_webrtc_common",
|
| ]
|
| }
|
|
|
| - static_library("libjingle_webrtc_common") {
|
| - sources = [
|
| - "../webrtc/api/audiotrack.cc",
|
| - "../webrtc/api/audiotrack.h",
|
| - "../webrtc/api/datachannel.cc",
|
| - "../webrtc/api/datachannel.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/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/remoteaudiosource.cc",
|
| - "../webrtc/api/remoteaudiosource.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/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/simulcast.cc",
|
| - "../webrtc/media/engine/simulcast.h",
|
| - "../webrtc/media/engine/webrtccommon.h",
|
| - "../webrtc/media/engine/webrtcmediaengine.cc",
|
| - "../webrtc/media/engine/webrtcmediaengine.h",
|
| - "../webrtc/media/engine/webrtcvideoengine2.cc",
|
| - "../webrtc/media/engine/webrtcvideoengine2.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/media/engine/webrtcvoiceengine.cc",
|
| - "../webrtc/media/engine/webrtcvoiceengine.h",
|
| - "../webrtc/media/sctp/sctpdataengine.cc",
|
| - "../webrtc/media/sctp/sctpdataengine.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",
|
| + source_set("libjingle_webrtc_common") {
|
| + configs += [
|
| + ":jingle_unexported_configs",
|
| + "//third_party/webrtc:common_config",
|
| + ]
|
| + public_configs = [
|
| + ":jingle_public_configs",
|
| + "//third_party/webrtc:common_inherited_config",
|
| ]
|
| -
|
| - 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/usrsctp",
|
| "//third_party/webrtc",
|
| + "//third_party/webrtc/api:libjingle_peerconnection",
|
| + "//third_party/webrtc/media:rtc_media",
|
| "//third_party/webrtc/modules/media_file",
|
| "//third_party/webrtc/modules/video_capture",
|
| + "//third_party/webrtc/pc:rtc_pc",
|
| "//third_party/webrtc/system_wrappers",
|
| "//third_party/webrtc/voice_engine",
|
| ]
|
|
|