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GN: Update webrtc/base to recent GYP changes.

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Update the webrtc/base/BUILD.gn file to reflect
webrtc/base/base.gyp changes between r6438 and r7011.

BUG=3441
TESTED= Trybots + compilation with a standalone WebRTC checkout:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default

Compilation of Chromium's 'all' target with src/third_party/webrtc
symlinked to the WebRTC checkout with this CL applied, both
with the default GN settings and using
--args="is_debug=false os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7022 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kjellander@webrtc.org
2014-09-02 11:22:06 +00:00
parent 468516c959
commit e281f7fba3
2 changed files with 78 additions and 30 deletions
Download Patch File Download Diff File Expand all files Collapse all files

100
webrtc/base/BUILD.gn
View File

@@ -6,8 +6,6 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# TODO(kjellander): Rebase this to base.gyp changes after r6438.
import("//build/config/crypto.gni")
import("//build/config/ui.gni")
import("../build/webrtc.gni")
@@ -20,7 +18,8 @@ config("webrtc_base_config") {
defines = [
"FEATURE_ENABLE_SSL",
"GTEST_RELATIVE_PATH",
"LOGGING=1",
"USE_WEBRTC_DEV_BRANCH",
]
# TODO(henrike): issue 3307, make webrtc_base build without disabling
@@ -103,10 +102,18 @@ if (is_linux && !build_with_chromium) {
}
}
if (build_ssl == 0) {
config("external_ssl_library") {
assert(webrtc_ssl_root != "",
"You must specify webrtc_ssl_root when build_ssl==0.")
include_dirs = [ webrtc_ssl_root ]
}
}
static_library("webrtc_base") {
cflags = []
cflags_cc = []
libs = []
deps= []
configs += [
@@ -264,6 +271,9 @@ static_library("webrtc_base") {
"taskrunner.h",
"thread.cc",
"thread.h",
"thread_checker.h",
"thread_checker_impl.cc",
"thread_checker_impl.h",
"timeutils.cc",
"timeutils.h",
"timing.cc",
@@ -315,6 +325,8 @@ static_library("webrtc_base") {
"callback.h",
"callback.h.pump",
"constructormagic.h",
"exp_filter.cc",
"exp_filter.h",
"filelock.cc",
"filelock.h",
"fileutils_mock.h",
@@ -388,13 +400,6 @@ static_library("webrtc_base") {
]
}
if (use_x11) {
sources += [
"x11windowpicker.cc",
"x11windowpicker.h",
]
}
if (is_mac) {
sources += [
"macasyncsocket.cc",
@@ -420,12 +425,19 @@ static_library("webrtc_base") {
"win32socketserver.h",
]
}
if (build_json) {
deps += [ "//third_party/jsoncpp" ]
} else {
include_dirs += [ webrtc_jsoncpp_root ]
deps += [ "//third_party/jsoncpp" ]
# When defined changes the include path for json.h to where it is
# expected to be when building json outside of the standalone build.
defines += [ "WEBRTC_EXTERNAL_JSON" ]
}
} # !build_with_chromium
if (is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# Suppress warnings from the Chrome Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
@@ -439,8 +451,11 @@ static_library("webrtc_base") {
if (use_openssl) {
direct_dependent_configs += [ ":openssl_config" ]
deps += [ "//third_party/boringssl" ]
if (build_ssl) {
deps += [ "//third_party/boringssl" ]
} else {
configs += [ "external_ssl_library" ]
}
} else {
direct_dependent_configs += [ ":no_openssl_config" ]
}
@@ -453,7 +468,7 @@ static_library("webrtc_base") {
direct_dependent_configs += [ ":android_config" ]
libs = [
libs += [
"log",
"GLESv2"
]
@@ -464,12 +479,19 @@ static_library("webrtc_base") {
if (is_ios) {
all_dependent_configs += [ ":ios_config" ]
deps += [ "//net/third_party/nss/ssl:libssl" ]
if (build_ssl) {
deps += [ "//net/third_party/nss/ssl:libssl" ]
} else {
configs += [ "external_ssl_library" ]
}
}
if (is_linux) {
libs = [
"crypto",
if (use_x11) {
sources += [
"x11windowpicker.cc",
"x11windowpicker.h",
]
libs += [
"dl",
"rt",
"Xext",
@@ -477,7 +499,17 @@ static_library("webrtc_base") {
"Xcomposite",
"Xrender",
]
configs += [ "//third_party/nss:system_nss_no_ssl_config" ]
}
if (is_linux) {
libs += [
"crypto",
"dl",
"rt",
]
if (build_ssl) {
configs += [ "//third_party/nss:system_nss_no_ssl_config" ]
}
}
if (is_mac) {
@@ -492,7 +524,7 @@ static_library("webrtc_base") {
all_dependent_configs = [ ":mac_config" ]
libs = [
libs += [
"crypto", # $(SDKROOT)/usr/lib/libcrypto.dylib
"ssl", # $(SDKROOT)/usr/lib/libssl.dylib
]
@@ -520,7 +552,7 @@ static_library("webrtc_base") {
"winping.h",
]
libs = [
libs += [
"crypt32.lib",
"iphlpapi.lib",
"secur32.lib",
@@ -536,7 +568,7 @@ static_library("webrtc_base") {
}
if (is_posix && is_debug) {
# Chromium's build/common.gypi defines this for all posix
# 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" ]
@@ -546,7 +578,7 @@ static_library("webrtc_base") {
defines += [ "CARBON_DEPRECATED=YES" ]
}
if (!is_ios && is_posix) {
if (!is_ios) {
sources += [
"openssl.h",
"openssladapter.cc",
@@ -568,18 +600,26 @@ static_library("webrtc_base") {
}
if (is_mac || is_ios || is_win) {
deps += [
"//net/third_party/nss/ssl:libssl",
"//third_party/nss:nspr",
"//third_party/nss:nss",
]
if (build_ssl) {
deps += [
"//net/third_party/nss/ssl:libssl",
"//third_party/nss:nspr",
"//third_party/nss:nss",
]
} else {
configs += [ "external_ssl_library" ]
}
}
if (is_posix && !is_mac && !is_ios && !is_android) {
if (build_with_chromium) {
deps += [ "//crypto:platform" ]
} else {
deps += [ ":linux_system_ssl" ]
if (build_ssl) {
deps += [ ":linux_system_ssl" ]
} else {
configs += [ "external_ssl_library" ]
}
}
}
}

8
webrtc/build/webrtc.gni
View File

@@ -17,6 +17,14 @@ declare_args() {
# Disable this to avoid building the Opus audio codec.
include_opus = true
# Used to specify an external Jsoncpp include path when not compiling the
# library that comes with WebRTC (i.e. build_json == 0).
webrtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
# Used to specify an external OpenSSL include path when not compiling the
# library that comes with WebRTC (i.e. build_ssl == 0).
webrtc_ssl_root = ""
# Adds video support to dependencies shared by voice and video engine.
# This should normally be enabled; the intended use is to disable only
# when building voice engine exclusively.