Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:20001 of https://codereview.webrtc.org/1973313002/ )
Reason for revert:
Breaks GN in Chromium (again), even though I tested this configuration: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/6000/steps/generate_build_files/logs/stdio
Original issue's description:
> 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
>
> Committed: https://crrev.com/c8d848b1049d8b9e8e33e023d13bec1180dd4926
> Cr-Commit-Position: refs/heads/master@{#12731}
TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review-Url: https://codereview.webrtc.org/1975223002
Cr-Original-Commit-Position: refs/heads/master@{#12733}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: fb1dd43ac137e50c0c26e97f5e9866135da24e2c
diff --git a/BUILD.gn b/BUILD.gn
index 374e652..e1aeb45 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -84,10 +84,6 @@
all_dependent_configs = [ "dbus-glib" ]
}
- if (rtc_relative_path) {
- defines += [ "EXPAT_RELATIVE_PATH" ]
- }
-
if (build_with_chromium) {
defines += [ "LOGGING_INSIDE_WEBRTC" ]
} else {
@@ -186,13 +182,11 @@
deps = [
":webrtc_common",
- "api",
"audio",
"base:rtc_base",
"call",
"common_audio",
"common_video",
- "media",
"modules/audio_coding",
"modules/audio_conference_mixer",
"modules/audio_device",
@@ -204,8 +198,6 @@
"modules/utility",
"modules/video_coding",
"modules/video_processing",
- "p2p",
- "pc",
"system_wrappers",
"tools",
"video",
diff --git a/api/BUILD.gn b/api/BUILD.gn
index f84010d..6dc5217 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -7,130 +7,3 @@
# 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/build/webrtc.gni b/build/webrtc.gni
index 72664c9..8e1b952 100644
--- a/build/webrtc.gni
+++ b/build/webrtc.gni
@@ -15,9 +15,6 @@
# 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"
@@ -40,13 +37,11 @@
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
@@ -100,19 +95,12 @@
# 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/libjingle/xmllite/BUILD.gn b/libjingle/xmllite/BUILD.gn
deleted file mode 100644
index 8495580..0000000
--- a/libjingle/xmllite/BUILD.gn
+++ /dev/null
@@ -1,54 +0,0 @@
-# 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/libjingle/xmpp/BUILD.gn b/libjingle/xmpp/BUILD.gn
deleted file mode 100644
index a3dfc51..0000000
--- a/libjingle/xmpp/BUILD.gn
+++ /dev/null
@@ -1,154 +0,0 @@
-# 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/media/BUILD.gn b/media/BUILD.gn
deleted file mode 100644
index c245d6e..0000000
--- a/media/BUILD.gn
+++ /dev/null
@@ -1,206 +0,0 @@
-# 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/p2p/BUILD.gn b/p2p/BUILD.gn
deleted file mode 100644
index 538781b..0000000
--- a/p2p/BUILD.gn
+++ /dev/null
@@ -1,138 +0,0 @@
-# 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/pc/BUILD.gn b/pc/BUILD.gn
deleted file mode 100644
index 50bb26a..0000000
--- a/pc/BUILD.gn
+++ /dev/null
@@ -1,70 +0,0 @@
-# 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" ]
- }
-}
