Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in
preparation for removing src/third_party/libjingle in Chromium.
BUG=webrtc:4256
NOTRY=True
TBR=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/1973313002
Cr-Commit-Position: refs/heads/master@{#12731}
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index e1aeb45..374e652 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -84,6 +84,10 @@
all_dependent_configs = [ "dbus-glib" ]
}
+ if (rtc_relative_path) {
+ defines += [ "EXPAT_RELATIVE_PATH" ]
+ }
+
if (build_with_chromium) {
defines += [ "LOGGING_INSIDE_WEBRTC" ]
} else {
@@ -182,11 +186,13 @@
deps = [
":webrtc_common",
+ "api",
"audio",
"base:rtc_base",
"call",
"common_audio",
"common_video",
+ "media",
"modules/audio_coding",
"modules/audio_conference_mixer",
"modules/audio_device",
@@ -198,6 +204,8 @@
"modules/utility",
"modules/video_coding",
"modules/video_processing",
+ "p2p",
+ "pc",
"system_wrappers",
"tools",
"video",
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 6dc5217..f84010d 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -7,3 +7,130 @@
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
+
+group("api") {
+ deps = [
+ ":libjingle_peerconnection",
+ ]
+}
+
+config("libjingle_peerconnection_warnings_config") {
+ # GN orders flags on a target before flags from configs. The default config
+ # adds these flags so to cancel them out they need to come from a config and
+ # cannot be on the target directly.
+ if (!is_win) {
+ cflags = [ "-Wno-sign-compare" ]
+ if (!is_clang) {
+ cflags += [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
+ }
+ }
+}
+
+source_set("libjingle_peerconnection") {
+ cflags = []
+ sources = [
+ "audiotrack.cc",
+ "audiotrack.h",
+ "datachannel.cc",
+ "datachannel.h",
+ "datachannelinterface.h",
+ "dtlsidentitystore.cc",
+ "dtlsidentitystore.h",
+ "dtmfsender.cc",
+ "dtmfsender.h",
+ "dtmfsenderinterface.h",
+ "jsep.h",
+ "jsepicecandidate.cc",
+ "jsepicecandidate.h",
+ "jsepsessiondescription.cc",
+ "jsepsessiondescription.h",
+ "localaudiosource.cc",
+ "localaudiosource.h",
+ "mediaconstraintsinterface.cc",
+ "mediaconstraintsinterface.h",
+ "mediacontroller.cc",
+ "mediacontroller.h",
+ "mediastream.cc",
+ "mediastream.h",
+ "mediastreaminterface.h",
+ "mediastreamobserver.cc",
+ "mediastreamobserver.h",
+ "mediastreamprovider.h",
+ "mediastreamproxy.h",
+ "mediastreamtrack.h",
+ "mediastreamtrackproxy.h",
+ "notifier.h",
+ "peerconnection.cc",
+ "peerconnection.h",
+ "peerconnectionfactory.cc",
+ "peerconnectionfactory.h",
+ "peerconnectionfactoryproxy.h",
+ "peerconnectioninterface.h",
+ "peerconnectionproxy.h",
+ "proxy.h",
+ "remoteaudiosource.cc",
+ "remoteaudiosource.h",
+ "rtpparameters.h",
+ "rtpreceiver.cc",
+ "rtpreceiver.h",
+ "rtpreceiverinterface.h",
+ "rtpsender.cc",
+ "rtpsender.h",
+ "rtpsenderinterface.h",
+ "sctputils.cc",
+ "sctputils.h",
+ "statscollector.cc",
+ "statscollector.h",
+ "statstypes.cc",
+ "statstypes.h",
+ "streamcollection.h",
+ "videocapturertracksource.cc",
+ "videocapturertracksource.h",
+ "videosourceproxy.h",
+ "videotrack.cc",
+ "videotrack.h",
+ "videotracksource.cc",
+ "videotracksource.h",
+ "webrtcsdp.cc",
+ "webrtcsdp.h",
+ "webrtcsession.cc",
+ "webrtcsession.h",
+ "webrtcsessiondescriptionfactory.cc",
+ "webrtcsessiondescriptionfactory.h",
+ ]
+
+ configs += [
+ "..:common_config",
+ ":libjingle_peerconnection_warnings_config",
+ ]
+ public_configs = [ "..:common_inherited_config" ]
+
+ if (is_clang) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:extra_warnings" ]
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ if (is_win) {
+ cflags += [ "/wd4389" ] # signed/unsigned mismatch.
+ }
+
+ deps = [
+ "../media",
+ "../pc",
+ ]
+
+ if (rtc_use_quic) {
+ sources += [
+ "quicdatachannel.cc",
+ "quicdatachannel.h",
+ "quicdatatransport.cc",
+ "quicdatatransport.h",
+ ]
+ deps += [ "//third_party/libquic" ]
+ public_deps = [
+ "//third_party/libquic",
+ ]
+ }
+}
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 8e1b952..72664c9 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -15,6 +15,9 @@
# Disable this to avoid building the Opus audio codec.
rtc_include_opus = true
+ # Disable to use absolute header paths for some libraries.
+ rtc_relative_path = true
+
# Used to specify an external Jsoncpp include path when not compiling the
# library that comes with WebRTC (i.e. rtc_build_json == 0).
rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
@@ -37,11 +40,13 @@
rtc_build_expat = true
rtc_build_json = true
rtc_build_libjpeg = true
+ rtc_build_libsrtp = true
rtc_build_libvpx = true
rtc_build_libyuv = true
rtc_build_openmax_dl = true
rtc_build_opus = true
rtc_build_ssl = true
+ rtc_build_usrsctp = true
# Disable by default.
rtc_have_dbus_glib = false
@@ -95,12 +100,19 @@
# http://www.openh264.org, https://www.ffmpeg.org/
rtc_use_h264 = proprietary_codecs && !is_android && !is_ios
+ # Determines whether QUIC code will be built.
+ rtc_use_quic = false
+
# FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
# by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
# only be initialized once. Projects that initialize FFmpeg externally, such
# as Chromium, must turn this flag off so that WebRTC does not also
# initialize.
rtc_initialize_ffmpeg = !build_with_chromium
+
+ # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
+ # build environments, even if available for Chromium builds.
+ rtc_use_gtk = !build_with_chromium
}
# A second declare_args block, so that declarations within it can
diff --git a/webrtc/libjingle/xmllite/BUILD.gn b/webrtc/libjingle/xmllite/BUILD.gn
new file mode 100644
index 0000000..8495580
--- /dev/null
+++ b/webrtc/libjingle/xmllite/BUILD.gn
@@ -0,0 +1,54 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../build/webrtc.gni")
+
+group("xmllite") {
+ deps = [
+ ":rtc_xmllite",
+ ]
+}
+
+source_set("rtc_xmllite") {
+ sources = [
+ "qname.cc",
+ "qname.h",
+ "xmlbuilder.cc",
+ "xmlbuilder.h",
+ "xmlconstants.cc",
+ "xmlconstants.h",
+ "xmlelement.cc",
+ "xmlelement.h",
+ "xmlnsstack.cc",
+ "xmlnsstack.h",
+ "xmlparser.cc",
+ "xmlparser.h",
+ "xmlprinter.cc",
+ "xmlprinter.h",
+ ]
+
+ deps = [
+ "../../base:rtc_base",
+ ]
+
+ if (rtc_build_expat) {
+ deps += [ "//third_party/expat" ]
+ public_deps = [
+ "//third_party/expat",
+ ]
+ }
+
+ configs += [ "../..:common_config" ]
+ public_configs = [ "../..:common_inherited_config" ]
+
+ if (is_clang) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
+}
diff --git a/webrtc/libjingle/xmpp/BUILD.gn b/webrtc/libjingle/xmpp/BUILD.gn
new file mode 100644
index 0000000..a3dfc51
--- /dev/null
+++ b/webrtc/libjingle/xmpp/BUILD.gn
@@ -0,0 +1,154 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../build/webrtc.gni")
+
+group("xmpp") {
+ deps = [
+ ":rtc_xmpp",
+ ]
+}
+
+config("xmpp_warnings_config") {
+ # GN orders flags on a target before flags from configs. The default config
+ # adds these flags so to cancel them out they need to come from a config and
+ # cannot be on the target directly.
+ if (is_android) {
+ cflags = [ "-Wno-error" ]
+ }
+}
+
+config("xmpp_inherited_config") {
+ defines = [
+ "FEATURE_ENABLE_SSL",
+ "FEATURE_ENABLE_VOICEMAIL",
+ ]
+}
+
+source_set("rtc_xmpp") {
+ cflags = []
+ sources = [
+ "asyncsocket.h",
+ "chatroommodule.h",
+ "chatroommoduleimpl.cc",
+ "constants.cc",
+ "constants.h",
+ "discoitemsquerytask.cc",
+ "discoitemsquerytask.h",
+ "hangoutpubsubclient.cc",
+ "hangoutpubsubclient.h",
+ "iqtask.cc",
+ "iqtask.h",
+ "jid.cc",
+ "jid.h",
+ "module.h",
+ "moduleimpl.cc",
+ "moduleimpl.h",
+ "mucroomconfigtask.cc",
+ "mucroomconfigtask.h",
+ "mucroomdiscoverytask.cc",
+ "mucroomdiscoverytask.h",
+ "mucroomlookuptask.cc",
+ "mucroomlookuptask.h",
+ "mucroomuniquehangoutidtask.cc",
+ "mucroomuniquehangoutidtask.h",
+ "pingtask.cc",
+ "pingtask.h",
+ "plainsaslhandler.h",
+ "presenceouttask.cc",
+ "presenceouttask.h",
+ "presencereceivetask.cc",
+ "presencereceivetask.h",
+ "presencestatus.cc",
+ "presencestatus.h",
+ "prexmppauth.h",
+ "pubsub_task.cc",
+ "pubsub_task.h",
+ "pubsubclient.cc",
+ "pubsubclient.h",
+ "pubsubstateclient.cc",
+ "pubsubstateclient.h",
+ "pubsubtasks.cc",
+ "pubsubtasks.h",
+ "receivetask.cc",
+ "receivetask.h",
+ "rostermodule.h",
+ "rostermoduleimpl.cc",
+ "rostermoduleimpl.h",
+ "saslcookiemechanism.h",
+ "saslhandler.h",
+ "saslmechanism.cc",
+ "saslmechanism.h",
+ "saslplainmechanism.h",
+ "xmppauth.cc",
+ "xmppauth.h",
+ "xmppclient.cc",
+ "xmppclient.h",
+ "xmppclientsettings.h",
+ "xmppengine.h",
+ "xmppengineimpl.cc",
+ "xmppengineimpl.h",
+ "xmppengineimpl_iq.cc",
+ "xmpplogintask.cc",
+ "xmpplogintask.h",
+ "xmpppump.cc",
+ "xmpppump.h",
+ "xmppsocket.cc",
+ "xmppsocket.h",
+ "xmppstanzaparser.cc",
+ "xmppstanzaparser.h",
+ "xmpptask.cc",
+ "xmpptask.h",
+ "xmppthread.cc",
+ "xmppthread.h",
+ ]
+
+ defines = [ "FEATURE_ENABLE_SSL" ]
+
+ deps = [
+ "../../base:rtc_base",
+ "../xmllite",
+ ]
+
+ if (rtc_build_expat) {
+ deps += [ "//third_party/expat" ]
+ public_deps = [
+ "//third_party/expat",
+ ]
+ }
+
+ configs += [
+ "../..:common_config",
+ ":xmpp_warnings_config",
+ ]
+
+ public_configs = [
+ "../..:common_inherited_config",
+ ":xmpp_inherited_config",
+ ]
+
+ if (!build_with_chromium) {
+ defines += [
+ "FEATURE_ENABLE_VOICEMAIL",
+ "FEATURE_ENABLE_PSTN",
+ ]
+ }
+
+ if (is_posix && is_debug) {
+ # The Chromium build/common.gypi defines this for all posix
+ # _except_ for ios & mac. We want it there as well, e.g.
+ # because ASSERT and friends trigger off of it.
+ defines += [ "_DEBUG" ]
+ }
+
+ if (is_clang) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
+}
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
new file mode 100644
index 0000000..c245d6e
--- /dev/null
+++ b/webrtc/media/BUILD.gn
@@ -0,0 +1,206 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/linux/pkg_config.gni")
+import("../build/webrtc.gni")
+
+group("media") {
+ deps = [
+ ":rtc_media",
+ ]
+}
+
+config("rtc_media_defines_config") {
+ defines = [
+ "HAVE_WEBRTC_VIDEO",
+ "HAVE_WEBRTC_VOICE",
+ ]
+}
+
+config("rtc_media_warnings_config") {
+ # GN orders flags on a target before flags from configs. The default config
+ # adds these flags so to cancel them out they need to come from a config and
+ # cannot be on the target directly.
+ if (!is_win) {
+ cflags = [ "-Wno-deprecated-declarations" ]
+ cflags_cc = [ "-Wno-overloaded-virtual" ]
+ }
+}
+
+if (is_linux && rtc_use_gtk) {
+ pkg_config("gtk-lib") {
+ packages = [
+ "gobject-2.0",
+ "gthread-2.0",
+ "gtk+-2.0",
+ ]
+ }
+}
+
+source_set("rtc_media") {
+ defines = []
+ libs = []
+ deps = []
+ sources = [
+ "base/audiosource.h",
+ "base/codec.cc",
+ "base/codec.h",
+ "base/cpuid.cc",
+ "base/cpuid.h",
+ "base/cryptoparams.h",
+ "base/device.h",
+ "base/fakescreencapturerfactory.h",
+ "base/hybriddataengine.h",
+ "base/mediachannel.h",
+ "base/mediacommon.h",
+ "base/mediaconstants.cc",
+ "base/mediaconstants.h",
+ "base/mediaengine.cc",
+ "base/mediaengine.h",
+ "base/rtpdataengine.cc",
+ "base/rtpdataengine.h",
+ "base/rtpdump.cc",
+ "base/rtpdump.h",
+ "base/rtputils.cc",
+ "base/rtputils.h",
+ "base/screencastid.h",
+ "base/streamparams.cc",
+ "base/streamparams.h",
+ "base/turnutils.cc",
+ "base/turnutils.h",
+ "base/videoadapter.cc",
+ "base/videoadapter.h",
+ "base/videobroadcaster.cc",
+ "base/videobroadcaster.h",
+ "base/videocapturer.cc",
+ "base/videocapturer.h",
+ "base/videocapturerfactory.h",
+ "base/videocommon.cc",
+ "base/videocommon.h",
+ "base/videoframe.cc",
+ "base/videoframe.h",
+ "base/videoframefactory.cc",
+ "base/videoframefactory.h",
+ "base/videorenderer.h",
+ "base/videosourcebase.cc",
+ "base/videosourcebase.h",
+ "base/yuvframegenerator.cc",
+ "base/yuvframegenerator.h",
+ "devices/videorendererfactory.h",
+ "engine/nullwebrtcvideoengine.h",
+ "engine/simulcast.cc",
+ "engine/simulcast.h",
+ "engine/webrtccommon.h",
+ "engine/webrtcmediaengine.cc",
+ "engine/webrtcmediaengine.h",
+ "engine/webrtcvideocapturer.cc",
+ "engine/webrtcvideocapturer.h",
+ "engine/webrtcvideocapturerfactory.cc",
+ "engine/webrtcvideocapturerfactory.h",
+ "engine/webrtcvideodecoderfactory.h",
+ "engine/webrtcvideoencoderfactory.h",
+ "engine/webrtcvideoengine2.cc",
+ "engine/webrtcvideoengine2.h",
+ "engine/webrtcvideoframe.cc",
+ "engine/webrtcvideoframe.h",
+ "engine/webrtcvideoframefactory.cc",
+ "engine/webrtcvideoframefactory.h",
+ "engine/webrtcvoe.h",
+ "engine/webrtcvoiceengine.cc",
+ "engine/webrtcvoiceengine.h",
+ "sctp/sctpdataengine.cc",
+ "sctp/sctpdataengine.h",
+ ]
+
+ configs += [
+ "..:common_config",
+ ":rtc_media_warnings_config",
+ ]
+
+ public_configs = [ "..:common_inherited_config" ]
+
+ if (is_clang) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:extra_warnings" ]
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ if (is_win) {
+ cflags = [
+ "/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch.
+ "/wd4267", # conversion from "size_t" to "int", possible loss of data.
+ "/wd4389", # signed/unsigned mismatch.
+ ]
+ }
+
+ if (rtc_build_libyuv) {
+ deps += [ "$rtc_libyuv_dir" ]
+ public_deps = [
+ "$rtc_libyuv_dir",
+ ]
+ } else {
+ # Need to add a directory normally exported by libyuv.
+ include_dirs += [ "$rtc_libyuv_dir/include" ]
+ }
+
+ if (rtc_build_usrsctp) {
+ include_dirs = [
+ # TODO(jiayl): move this into the public_configs of
+ # //third_party/usrsctp/BUILD.gn.
+ "//third_party/usrsctp/usrsctplib",
+ ]
+ deps += [ "//third_party/usrsctp" ]
+ }
+
+ if (build_with_chromium) {
+ deps += [ "../modules/video_capture:video_capture" ]
+ } else {
+ configs += [ ":rtc_media_defines_config" ]
+ public_configs += [ ":rtc_media_defines_config" ]
+ deps += [ "../modules/video_capture:video_capture_internal_impl" ]
+ }
+ if (is_linux && rtc_use_gtk) {
+ sources += [
+ "devices/gtkvideorenderer.cc",
+ "devices/gtkvideorenderer.h",
+ ]
+ public_configs += [ ":gtk-lib" ]
+ }
+ if (is_win) {
+ sources += [
+ "devices/gdivideorenderer.cc",
+ "devices/gdivideorenderer.h",
+ ]
+ libs += [
+ "d3d9.lib",
+ "gdi32.lib",
+ "strmiids.lib",
+ ]
+ }
+ if (is_mac && current_cpu == "x86") {
+ sources += [
+ "devices/carbonvideorenderer.cc",
+ "devices/carbonvideorenderer.h",
+ ]
+ libs += [ "Carbon.framework" ]
+ }
+ if (is_ios || (is_mac && current_cpu != "x86")) {
+ defines += [ "CARBON_DEPRECATED=YES" ]
+ }
+
+ deps += [
+ "..:webrtc_common",
+ "../base:rtc_base_approved",
+ "../libjingle/xmllite",
+ "../libjingle/xmpp",
+ "../p2p",
+ "../system_wrappers",
+ "../voice_engine",
+ ]
+}
diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn
new file mode 100644
index 0000000..538781b
--- /dev/null
+++ b/webrtc/p2p/BUILD.gn
@@ -0,0 +1,138 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+
+group("p2p") {
+ deps = [
+ ":rtc_p2p",
+ ]
+}
+
+config("rtc_p2p_inherited_config") {
+ defines = [ "FEATURE_ENABLE_VOICEMAIL" ]
+}
+
+source_set("rtc_p2p") {
+ sources = [
+ "base/asyncstuntcpsocket.cc",
+ "base/asyncstuntcpsocket.h",
+ "base/basicpacketsocketfactory.cc",
+ "base/basicpacketsocketfactory.h",
+ "base/candidate.h",
+ "base/common.h",
+ "base/dtlstransportchannel.cc",
+ "base/dtlstransportchannel.h",
+ "base/p2pconstants.cc",
+ "base/p2pconstants.h",
+ "base/p2ptransport.cc",
+ "base/p2ptransport.h",
+ "base/p2ptransportchannel.cc",
+ "base/p2ptransportchannel.h",
+ "base/packetsocketfactory.h",
+ "base/port.cc",
+ "base/port.h",
+ "base/portallocator.cc",
+ "base/portallocator.h",
+ "base/portinterface.h",
+ "base/pseudotcp.cc",
+ "base/pseudotcp.h",
+ "base/relayport.cc",
+ "base/relayport.h",
+ "base/relayserver.cc",
+ "base/relayserver.h",
+ "base/sessiondescription.cc",
+ "base/sessiondescription.h",
+ "base/sessionid.h",
+ "base/stun.cc",
+ "base/stun.h",
+ "base/stunport.cc",
+ "base/stunport.h",
+ "base/stunrequest.cc",
+ "base/stunrequest.h",
+ "base/stunserver.cc",
+ "base/stunserver.h",
+ "base/tcpport.cc",
+ "base/tcpport.h",
+ "base/transport.cc",
+ "base/transport.h",
+ "base/transportchannel.cc",
+ "base/transportchannel.h",
+ "base/transportchannelimpl.h",
+ "base/transportcontroller.cc",
+ "base/transportcontroller.h",
+ "base/transportdescription.cc",
+ "base/transportdescription.h",
+ "base/transportdescriptionfactory.cc",
+ "base/transportdescriptionfactory.h",
+ "base/transportinfo.h",
+ "base/turnport.cc",
+ "base/turnport.h",
+ "base/turnserver.cc",
+ "base/turnserver.h",
+ "base/udpport.h",
+ "client/basicportallocator.cc",
+ "client/basicportallocator.h",
+ "client/httpportallocator.cc",
+ "client/httpportallocator.h",
+ "client/socketmonitor.cc",
+ "client/socketmonitor.h",
+ ]
+
+ defines = [ "FEATURE_ENABLE_SSL" ]
+
+ deps = [
+ "../base:rtc_base",
+ "../libjingle/xmllite",
+ ]
+
+ if (rtc_build_expat) {
+ deps += [ "//third_party/expat" ]
+ public_deps = [
+ "//third_party/expat",
+ ]
+ }
+
+ configs += [ "..:common_config" ]
+ public_configs = [
+ "..:common_inherited_config",
+ ":rtc_p2p_inherited_config",
+ ]
+
+ if (!build_with_chromium) {
+ defines += [
+ "FEATURE_ENABLE_VOICEMAIL",
+ "FEATURE_ENABLE_PSTN",
+ ]
+ }
+
+ if (rtc_use_quic) {
+ deps = [
+ "//third_party/libquic",
+ ]
+ sources += [
+ "quic/quicconnectionhelper.cc",
+ "quic/quicconnectionhelper.h",
+ "quic/quicsession.cc",
+ "quic/quicsession.h",
+ "quic/quictransport.cc",
+ "quic/quictransport.h",
+ "quic/quictransportchannel.cc",
+ "quic/quictransportchannel.h",
+ "quic/reliablequicstream.cc",
+ "quic/reliablequicstream.h",
+ ]
+ public_deps += [ "//third_party/libquic" ]
+ }
+
+ if (is_clang) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
+}
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
new file mode 100644
index 0000000..50bb26a
--- /dev/null
+++ b/webrtc/pc/BUILD.gn
@@ -0,0 +1,70 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+
+group("pc") {
+ deps = [
+ ":rtc_pc",
+ ]
+}
+
+config("rtc_pc_config") {
+ defines = [
+ "SRTP_RELATIVE_PATH",
+ "HAVE_SCTP",
+ "HAVE_SRTP",
+ ]
+}
+
+source_set("rtc_pc") {
+ defines = []
+ sources = [
+ "audiomonitor.cc",
+ "audiomonitor.h",
+ "bundlefilter.cc",
+ "bundlefilter.h",
+ "channel.cc",
+ "channel.h",
+ "channelmanager.cc",
+ "channelmanager.h",
+ "currentspeakermonitor.cc",
+ "currentspeakermonitor.h",
+ "mediamonitor.cc",
+ "mediamonitor.h",
+ "mediasession.cc",
+ "mediasession.h",
+ "mediasink.h",
+ "rtcpmuxfilter.cc",
+ "rtcpmuxfilter.h",
+ "srtpfilter.cc",
+ "srtpfilter.h",
+ "voicechannel.h",
+ ]
+
+ deps = [
+ "../base:rtc_base",
+ "../media",
+ ]
+
+ if (rtc_build_libsrtp) {
+ deps += [ "//third_party/libsrtp" ]
+ }
+
+ configs += [ "..:common_config" ]
+ public_configs = [
+ "..:common_inherited_config",
+ ":rtc_pc_config",
+ ]
+
+ if (is_clang) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
+}
