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
Review-Url: https://codereview.webrtc.org/1929633002
Cr-Commit-Position: refs/heads/master@{#12724}
This commit is contained in:
kjellander
@ -84,6 +84,10 @@ config("common_config") {
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all_dependent_configs = [ "dbus-glib" ]
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}
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if (rtc_relative_path) {
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defines += [ "EXPAT_RELATIVE_PATH" ]
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}
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if (build_with_chromium) {
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defines += [ "LOGGING_INSIDE_WEBRTC" ]
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} else {
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@ -182,11 +186,13 @@ source_set("webrtc") {
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deps = [
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":webrtc_common",
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"api",
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"audio",
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"base:rtc_base",
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"call",
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"common_audio",
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"common_video",
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"media",
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"modules/audio_coding",
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"modules/audio_conference_mixer",
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"modules/audio_device",
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@ -198,6 +204,8 @@ source_set("webrtc") {
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"modules/utility",
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"modules/video_coding",
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"modules/video_processing",
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"p2p",
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"pc",
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"system_wrappers",
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"tools",
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"video",
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@ -7,3 +7,130 @@
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# be found in the AUTHORS file in the root of the source tree.
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import("../build/webrtc.gni")
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group("api") {
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deps = [
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":libjingle_peerconnection",
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]
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}
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config("libjingle_peerconnection_warnings_config") {
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# GN orders flags on a target before flags from configs. The default config
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# adds these flags so to cancel them out they need to come from a config and
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# cannot be on the target directly.
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if (!is_win) {
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cflags = [ "-Wno-sign-compare" ]
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if (!is_clang) {
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cflags += [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
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}
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}
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}
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source_set("libjingle_peerconnection") {
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cflags = []
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sources = [
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"audiotrack.cc",
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"audiotrack.h",
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"datachannel.cc",
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"datachannel.h",
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"datachannelinterface.h",
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"dtlsidentitystore.cc",
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"dtlsidentitystore.h",
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"dtmfsender.cc",
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"dtmfsender.h",
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"dtmfsenderinterface.h",
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"jsep.h",
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"jsepicecandidate.cc",
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"jsepicecandidate.h",
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"jsepsessiondescription.cc",
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"jsepsessiondescription.h",
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"localaudiosource.cc",
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"localaudiosource.h",
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"mediaconstraintsinterface.cc",
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"mediaconstraintsinterface.h",
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"mediacontroller.cc",
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"mediacontroller.h",
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"mediastream.cc",
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"mediastream.h",
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"mediastreaminterface.h",
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"mediastreamobserver.cc",
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"mediastreamobserver.h",
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"mediastreamprovider.h",
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"mediastreamproxy.h",
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"mediastreamtrack.h",
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"mediastreamtrackproxy.h",
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"notifier.h",
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"peerconnection.cc",
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"peerconnection.h",
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"peerconnectionfactory.cc",
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"peerconnectionfactory.h",
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"peerconnectionfactoryproxy.h",
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"peerconnectioninterface.h",
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"peerconnectionproxy.h",
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"proxy.h",
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"remoteaudiosource.cc",
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"remoteaudiosource.h",
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"rtpparameters.h",
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"rtpreceiver.cc",
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"rtpreceiver.h",
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"rtpreceiverinterface.h",
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"rtpsender.cc",
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"rtpsender.h",
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"rtpsenderinterface.h",
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"sctputils.cc",
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"sctputils.h",
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"statscollector.cc",
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"statscollector.h",
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"statstypes.cc",
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"statstypes.h",
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"streamcollection.h",
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"videocapturertracksource.cc",
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"videocapturertracksource.h",
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"videosourceproxy.h",
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"videotrack.cc",
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"videotrack.h",
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"videotracksource.cc",
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"videotracksource.h",
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"webrtcsdp.cc",
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"webrtcsdp.h",
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"webrtcsession.cc",
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"webrtcsession.h",
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"webrtcsessiondescriptionfactory.cc",
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"webrtcsessiondescriptionfactory.h",
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]
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configs += [
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"..:common_config",
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":libjingle_peerconnection_warnings_config",
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]
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public_configs = [ "..:common_inherited_config" ]
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if (is_clang) {
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# Suppress warnings from Chrome's Clang plugins.
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# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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configs -= [ "//build/config/clang:extra_warnings" ]
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configs -= [ "//build/config/clang:find_bad_constructs" ]
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}
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if (is_win) {
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cflags += [ "/wd4389" ] # signed/unsigned mismatch.
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}
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deps = [
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"../media",
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"../pc",
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]
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if (rtc_use_quic) {
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sources += [
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"quicdatachannel.cc",
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"quicdatachannel.h",
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"quicdatatransport.cc",
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"quicdatatransport.h",
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]
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deps += [ "//third_party/libquic" ]
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public_deps = [
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"//third_party/libquic",
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]
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}
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}
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@ -15,6 +15,9 @@ declare_args() {
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# Disable this to avoid building the Opus audio codec.
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rtc_include_opus = true
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# Disable to use absolute header paths for some libraries.
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rtc_relative_path = true
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# Used to specify an external Jsoncpp include path when not compiling the
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# library that comes with WebRTC (i.e. rtc_build_json == 0).
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rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
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@ -37,11 +40,13 @@ declare_args() {
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rtc_build_expat = true
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rtc_build_json = true
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rtc_build_libjpeg = true
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rtc_build_libsrtp = true
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rtc_build_libvpx = true
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rtc_build_libyuv = true
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rtc_build_openmax_dl = true
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rtc_build_opus = true
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rtc_build_ssl = true
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rtc_build_usrsctp = true
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# Disable by default.
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rtc_have_dbus_glib = false
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@ -95,12 +100,19 @@ declare_args() {
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# http://www.openh264.org, https://www.ffmpeg.org/
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rtc_use_h264 = proprietary_codecs && !is_android && !is_ios
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# Determines whether QUIC code will be built.
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rtc_use_quic = false
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# FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
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# by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
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# only be initialized once. Projects that initialize FFmpeg externally, such
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# as Chromium, must turn this flag off so that WebRTC does not also
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# initialize.
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rtc_initialize_ffmpeg = !build_with_chromium
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# Build sources requiring GTK. NOTICE: This is not present in Chrome OS
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# build environments, even if available for Chromium builds.
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rtc_use_gtk = !build_with_chromium
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}
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# A second declare_args block, so that declarations within it can
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54
webrtc/libjingle/xmllite/BUILD.gn
Normal file
54
webrtc/libjingle/xmllite/BUILD.gn
Normal file
@ -0,0 +1,54 @@
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# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../build/webrtc.gni")
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group("xmllite") {
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deps = [
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":rtc_xmllite",
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]
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}
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source_set("rtc_xmllite") {
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sources = [
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"qname.cc",
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"qname.h",
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"xmlbuilder.cc",
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"xmlbuilder.h",
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"xmlconstants.cc",
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"xmlconstants.h",
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"xmlelement.cc",
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"xmlelement.h",
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"xmlnsstack.cc",
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"xmlnsstack.h",
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"xmlparser.cc",
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"xmlparser.h",
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"xmlprinter.cc",
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"xmlprinter.h",
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]
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deps = [
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"../../base:rtc_base",
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]
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if (rtc_build_expat) {
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deps += [ "//third_party/expat" ]
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public_deps = [
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"//third_party/expat",
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]
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}
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configs += [ "../..:common_config" ]
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public_configs = [ "../..:common_inherited_config" ]
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if (is_clang) {
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# Suppress warnings from Chrome's Clang plugins.
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# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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configs -= [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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154
webrtc/libjingle/xmpp/BUILD.gn
Normal file
154
webrtc/libjingle/xmpp/BUILD.gn
Normal file
@ -0,0 +1,154 @@
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# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
|
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# be found in the AUTHORS file in the root of the source tree.
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import("../../build/webrtc.gni")
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group("xmpp") {
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deps = [
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":rtc_xmpp",
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]
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}
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config("xmpp_warnings_config") {
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# GN orders flags on a target before flags from configs. The default config
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# adds these flags so to cancel them out they need to come from a config and
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# cannot be on the target directly.
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if (is_android) {
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cflags = [ "-Wno-error" ]
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}
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}
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config("xmpp_inherited_config") {
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defines = [
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"FEATURE_ENABLE_SSL",
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"FEATURE_ENABLE_VOICEMAIL",
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]
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}
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source_set("rtc_xmpp") {
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cflags = []
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sources = [
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"asyncsocket.h",
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"chatroommodule.h",
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"chatroommoduleimpl.cc",
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"constants.cc",
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"constants.h",
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"discoitemsquerytask.cc",
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"discoitemsquerytask.h",
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"hangoutpubsubclient.cc",
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"hangoutpubsubclient.h",
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"iqtask.cc",
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"iqtask.h",
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"jid.cc",
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"jid.h",
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"module.h",
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"moduleimpl.cc",
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"moduleimpl.h",
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"mucroomconfigtask.cc",
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||||
"mucroomconfigtask.h",
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"mucroomdiscoverytask.cc",
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"mucroomdiscoverytask.h",
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"mucroomlookuptask.cc",
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||||
"mucroomlookuptask.h",
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||||
"mucroomuniquehangoutidtask.cc",
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||||
"mucroomuniquehangoutidtask.h",
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"pingtask.cc",
|
||||
"pingtask.h",
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||||
"plainsaslhandler.h",
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||||
"presenceouttask.cc",
|
||||
"presenceouttask.h",
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"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" ]
|
||||
}
|
||||
}
|
||||
206
webrtc/media/BUILD.gn
Normal file
206
webrtc/media/BUILD.gn
Normal file
@ -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" ]
|
||||
} 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",
|
||||
]
|
||||
}
|
||||
138
webrtc/p2p/BUILD.gn
Normal file
138
webrtc/p2p/BUILD.gn
Normal file
@ -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" ]
|
||||
}
|
||||
}
|
||||
70
webrtc/pc/BUILD.gn
Normal file
70
webrtc/pc/BUILD.gn
Normal file
@ -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" ]
|
||||
}
|
||||
}
|
||||
Reference in New Issue
Block a user