From 4d167e5ccde9e748488179b6dd3f26ccf35d7365 Mon Sep 17 00:00:00 2001 From: kjellander Date: Wed, 1 Jun 2016 04:45:05 -0700 Subject: [PATCH] Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #5 id:80001 of https://codereview.webrtc.org/1979933002/ ) Reason for revert: Too many errors to address showed up when trying to land this with Chromium changes in https://codereview.chromium.org/2022833002/. Will address them separately before relanding. 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. > > Changes from previous attempt: > * Added libstunprober target > * Adjusted warnings for Chromium's clang plugins > * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. > > As soon this has landed a roll including the changes in > https://codereview.chromium.org/2022833002/ is needed to make > Chromium build cleanly. > > BUG=webrtc:4256 > NOTRY=True > NOPRESUBMIT=True > > Committed: https://crrev.com/164e978f981c7810c4260c4184f41e26bae90230 > Cr-Commit-Position: refs/heads/master@{#12983} TBR=perkj@webrtc.org,tommi@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/2023233002 Cr-Commit-Position: refs/heads/master@{#12988} --- webrtc/BUILD.gn | 8 -- webrtc/api/BUILD.gn | 127 ------------------ webrtc/build/webrtc.gni | 13 -- webrtc/libjingle/xmllite/BUILD.gn | 54 -------- webrtc/libjingle/xmpp/BUILD.gn | 154 ---------------------- webrtc/media/BUILD.gn | 206 ------------------------------ webrtc/p2p/BUILD.gn | 155 ---------------------- webrtc/pc/BUILD.gn | 76 ----------- webrtc/pc/pc.gyp | 6 - 9 files changed, 799 deletions(-) delete mode 100644 webrtc/libjingle/xmllite/BUILD.gn delete mode 100644 webrtc/libjingle/xmpp/BUILD.gn delete mode 100644 webrtc/media/BUILD.gn delete mode 100644 webrtc/p2p/BUILD.gn delete mode 100644 webrtc/pc/BUILD.gn diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 37f74dd47b..07f2fe998d 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -113,10 +113,6 @@ config("common_config") { all_dependent_configs = [ "dbus-glib" ] } - if (rtc_relative_path) { - defines += [ "EXPAT_RELATIVE_PATH" ] - } - if (build_with_chromium) { defines += [ # NOTICE: Since common_inherited_config is used in public_configs for our @@ -237,13 +233,11 @@ source_set("webrtc") { deps = [ ":webrtc_common", - "api", "audio", "base:rtc_base", "call", "common_audio", "common_video", - "media", "modules/audio_coding", "modules/audio_conference_mixer", "modules/audio_device", @@ -255,8 +249,6 @@ source_set("webrtc") { "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 f84010d309..6dc52174ec 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/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/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni index aa86c99efa..1dc1bc488a 100644 --- a/webrtc/build/webrtc.gni +++ b/webrtc/build/webrtc.gni @@ -15,9 +15,6 @@ declare_args() { # 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" @@ -39,14 +36,11 @@ declare_args() { # Disable these to not build components which can be externally provided. 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 @@ -106,19 +100,12 @@ declare_args() { # 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 deleted file mode 100644 index c35dde69e0..0000000000 --- a/webrtc/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 (!build_with_chromium && 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 deleted file mode 100644 index c27c70bfc7..0000000000 --- a/webrtc/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_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" ] - } - } - - 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" ] - } -} diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn deleted file mode 100644 index c245d6ee3e..0000000000 --- a/webrtc/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/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn deleted file mode 100644 index 4940e34c98..0000000000 --- a/webrtc/p2p/BUILD.gn +++ /dev/null @@ -1,155 +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 (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" ] - } - } - - 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" ] - } -} - -source_set("libstunprober") { - sources = [ - "stunprober/stunprober.cc", - ] - - if (!build_with_chromium && 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" ] - } - - deps = [ - "..:webrtc_common", - "../base:rtc_base", - ] -} diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn deleted file mode 100644 index bf0f2b53f8..0000000000 --- a/webrtc/pc/BUILD.gn +++ /dev/null @@ -1,76 +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 (build_with_chromium) { - sources += [ - "externalhmac.cc", - "externalhmac.h", - ] - } - 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" ] - } -} diff --git a/webrtc/pc/pc.gyp b/webrtc/pc/pc.gyp index 8c10989037..25ebc5de78 100755 --- a/webrtc/pc/pc.gyp +++ b/webrtc/pc/pc.gyp @@ -24,12 +24,6 @@ '<(webrtc_root)/media/media.gyp:rtc_media', ], 'conditions': [ - ['build_with_chromium==1', { - 'sources': [ - 'externalhmac.h', - 'externalhmac.cc', - ], - }], ['build_libsrtp==1', { 'dependencies': [ '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',