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Reland "Set defines for Chromium"

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This is a reland of https://codereview.webrtc.org/1847013002/
with the following changes:
* _USE_32BIT_TIME_T is no longer set: it was removed from Chromium
in https://codereview.chromium.org/1862443003/.
Setting it in target_defaults was likely the reason to
remoting_unittests failing in the previous attempt to land this.
* Added define for FreeBSD platform.
* Added corresponding GN changes.

Copy the defines from the target_defaults section of Chromium's
src/third_party/libjingle.gyp into our webrtc/build/common.gypi
in order to ensure the same defines are used for the Chromium build
when removing the source listings in src/third_party/libjingle.gyp.
With this CL landed, it should be possible to replace them with
dependencies on:
* webrtc/api/api.gyp:libjingle_peerconnections
* webrtc/media/media.gyp:rtc_media
* webrtc/pc/pc.gyp:rtc_pc
* webrtc/pp2/p2p.gyp:rtc_p2p
* webrtc/libjingle/xmpp/xmpp.gyp:rtc_xmpp

Not ported (Windows specific):
* Precompiled headers (build/win_precompile.gypi):
  since it only seems to offer a compile speedup. Will be landed
  for all of WebRTC in separate CL.

BUG=webrtc:4256
NOTRY=True

Review-Url: https://codereview.webrtc.org/1924663003
Cr-Commit-Position: refs/heads/master@{#12959}
This commit is contained in:
kjellander 2016-05-30 06:27:50 -07:00 committed by Commit bot
parent 073eb7ed1e
commit f0e174aab4
2 changed files with 112 additions and 5 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

52
webrtc/BUILD.gn
View File

@ -21,7 +21,17 @@ config("common_inherited_config") {
defines += [ "WEBRTC_MOZILLA_BUILD" ]
}
if (build_with_chromium) {
defines = [ "WEBRTC_CHROMIUM_BUILD" ]
defines = [
# TODO(kjellander): Cleanup unused ones and move defines closer to
# the source when webrtc:4256 is completed.
"FEATURE_ENABLE_SSL",
"FEATURE_ENABLE_VOICEMAIL",
"EXPAT_RELATIVE_PATH",
"GTEST_RELATIVE_PATH",
"NO_MAIN_THREAD_WRAPPING",
"NO_SOUND_SYSTEM",
"WEBRTC_CHROMIUM_BUILD",
]
include_dirs = [
# The overrides must be included first as that is the mechanism for
# selecting the override headers in Chromium.
@ -51,7 +61,10 @@ config("common_inherited_config") {
defines += [ "WEBRTC_MAC" ]
}
if (is_win) {
defines += [ "WEBRTC_WIN" ]
defines += [
"WEBRTC_WIN",
"_CRT_SECURE_NO_WARNINGS", # Suppress warnings about _vsnprinf
]
}
if (is_android) {
defines += [
@ -59,6 +72,20 @@ config("common_inherited_config") {
"WEBRTC_ANDROID",
]
}
if (is_chromeos) {
defines += [ "CHROMEOS" ]
}
# TODO(GYP): Support these in GN.
# if (is_bsd) {
# defines += [ "BSD" ]
# }
# if (is_openbsd) {
# defines += [ "OPENBSD" ]
# }
# if (is_freebsd) {
# defines += [ "FREEBSD" ]
# }
}
if (rtc_have_dbus_glib) {
@ -70,8 +97,10 @@ if (rtc_have_dbus_glib) {
config("common_config") {
cflags = []
cflags_cc = []
defines = []
if (rtc_restrict_logging) {
defines = [ "WEBRTC_RESTRICT_LOGGING" ]
defines += [ "WEBRTC_RESTRICT_LOGGING" ]
}
if (rtc_have_dbus_glib) {
@ -85,7 +114,22 @@ config("common_config") {
}
if (build_with_chromium) {
defines += [ "LOGGING_INSIDE_WEBRTC" ]
defines += [
# NOTICE: Since common_inherited_config is used in public_configs for our
# targets, there's no point including the defines in that config here.
# TODO(kjellander): Cleanup unused ones and move defines closer to the
# source when webrtc:4256 is completed.
"ENABLE_EXTERNAL_AUTH",
"HAVE_OPENSSL_SSL_H",
"HAVE_SCTP",
"HAVE_SRTP",
"HAVE_WEBRTC_VIDEO",
"HAVE_WEBRTC_VOICE",
"LOGGING_INSIDE_WEBRTC",
"SRTP_RELATIVE_PATH",
"SSL_USE_OPENSSL",
"USE_WEBRTC_DEV_BRANCH",
]
} else {
if (is_posix) {
# -Wextra is currently disabled in Chromium"s common.gypi. Enable

65
webrtc/build/common.gypi
View File

@ -280,8 +280,22 @@
['build_with_chromium==1', {
'defines': [
# Changes settings for Chromium build.
'WEBRTC_CHROMIUM_BUILD',
# TODO(kjellander): Cleanup unused ones and move defines closer to the
# source when webrtc:4256 is completed.
'ENABLE_EXTERNAL_AUTH',
'FEATURE_ENABLE_SSL',
'HAVE_OPENSSL_SSL_H',
'HAVE_SCTP',
'HAVE_SRTP',
'HAVE_WEBRTC_VIDEO',
'HAVE_WEBRTC_VOICE',
'LOGGING_INSIDE_WEBRTC',
'NO_MAIN_THREAD_WRAPPING',
'NO_SOUND_SYSTEM',
'SRTP_RELATIVE_PATH',
'SSL_USE_OPENSSL',
'USE_WEBRTC_DEV_BRANCH',
'WEBRTC_CHROMIUM_BUILD',
],
'include_dirs': [
# Include the top-level directory when building in Chrome, so we can
@ -460,6 +474,26 @@
}],
],
}],
['chromeos==1', {
'defines': [
'CHROMEOS',
],
}],
['os_bsd==1', {
'defines': [
'BSD',
],
}],
['OS=="openbsd"', {
'defines': [
'OPENBSD',
],
}],
['OS=="freebsd"', {
'defines': [
'FREEBSD',
],
}],
['include_internal_audio_device==1', {
'defines': [
'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE',
@ -482,6 +516,14 @@
['build_with_chromium==1', {
'defines': [
# Changes settings for Chromium build.
# TODO(kjellander): Cleanup unused ones and move defines closer to
# the source when webrtc:4256 is completed.
'FEATURE_ENABLE_SSL',
'FEATURE_ENABLE_VOICEMAIL',
'EXPAT_RELATIVE_PATH',
'GTEST_RELATIVE_PATH',
'NO_MAIN_THREAD_WRAPPING',
'NO_SOUND_SYSTEM',
'WEBRTC_CHROMIUM_BUILD',
],
'include_dirs': [
@ -509,6 +551,7 @@
['OS=="win"', {
'defines': [
'WEBRTC_WIN',
'_CRT_SECURE_NO_WARNINGS', # Suppress warnings about _vsnprinf
],
}],
['OS=="linux"', {
@ -529,6 +572,26 @@
'WEBRTC_POSIX',
],
}],
['chromeos==1', {
'defines': [
'CHROMEOS',
],
}],
['os_bsd==1', {
'defines': [
'BSD',
],
}],
['OS=="openbsd"', {
'defines': [
'OPENBSD',
],
}],
['OS=="freebsd"', {
'defines': [
'FREEBSD',
],
}],
],
},
}, # target_defaults