diff options
author | kjellander@webrtc.org <kjellander@webrtc.org> | 2014-09-07 17:36:10 +0000 |
---|---|---|
committer | kjellander@webrtc.org <kjellander@webrtc.org> | 2014-09-07 17:36:10 +0000 |
commit | 4ad10d659f2e27b129aa510c35a49165ba348a03 (patch) | |
tree | 5b3776d28c56b708739dd32e91b1f092dfeb1ab7 | |
parent | 729d5a92e805a22c51c6cfd928275852e7d17a99 (diff) | |
download | webrtc-4ad10d659f2e27b129aa510c35a49165ba348a03.tar.gz |
GN: Prefix WebRTC specific variables with "rtc_"
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/27379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
-rw-r--r-- | BUILD.gn | 10 | ||||
-rw-r--r-- | base/BUILD.gn | 22 | ||||
-rw-r--r-- | build/webrtc.gni | 66 | ||||
-rw-r--r-- | common_audio/BUILD.gn | 4 | ||||
-rw-r--r-- | common_video/BUILD.gn | 2 | ||||
-rw-r--r-- | modules/audio_coding/BUILD.gn | 10 | ||||
-rw-r--r-- | modules/audio_device/BUILD.gn | 4 | ||||
-rw-r--r-- | modules/audio_processing/BUILD.gn | 12 | ||||
-rw-r--r-- | modules/utility/BUILD.gn | 2 | ||||
-rw-r--r-- | modules/video_capture/BUILD.gn | 2 | ||||
-rw-r--r-- | modules/video_coding/BUILD.gn | 2 | ||||
-rw-r--r-- | modules/video_render/BUILD.gn | 2 | ||||
-rw-r--r-- | system_wrappers/BUILD.gn | 2 |
13 files changed, 70 insertions, 70 deletions
@@ -57,13 +57,13 @@ config("common_inherited_config") { "WEBRTC_LINUX", "WEBRTC_ANDROID", ] - if (enable_android_opensl) { + if (rtc_enable_android_opensl) { defines += [ "WEBRTC_ANDROID_OPENSLES" ] } } } -if (have_dbus_glib) { +if (rtc_have_dbus_glib) { pkg_config("dbus-glib") { packages = [ "dbus-glib-1" ] } @@ -72,11 +72,11 @@ if (have_dbus_glib) { config("common_config") { cflags = [] cflags_cc = [] - if (restrict_webrtc_logging) { + if (rtc_restrict_logging) { defines = [ "WEBRTC_RESTRICT_LOGGING" ] } - if (have_dbus_glib) { + if (rtc_have_dbus_glib) { defines += [ "HAVE_DBUS_GLIB" ] # TODO(kjellander): Investigate this, it seems like include <dbus/dbus.h> # is still not found even if the execution of @@ -85,7 +85,7 @@ config("common_config") { all_dependent_configs = [ "dbus-glib" ] } - if (enable_video) { + if (rtc_enable_video) { defines += [ "WEBRTC_MODULE_UTILITY_VIDEO" ] } diff --git a/base/BUILD.gn b/base/BUILD.gn index b0da45af..4e14ab1c 100644 --- a/base/BUILD.gn +++ b/base/BUILD.gn @@ -102,11 +102,11 @@ if (is_linux && !build_with_chromium) { } } -if (build_ssl == 0) { +if (rtc_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 ] + assert(rtc_ssl_root != "", + "You must specify rtc_ssl_root when rtc_build_ssl==0.") + include_dirs = [ rtc_ssl_root ] } } @@ -425,10 +425,10 @@ static_library("webrtc_base") { "win32socketserver.h", ] } - if (build_json) { + if (rtc_build_json) { deps += [ "//third_party/jsoncpp" ] } else { - include_dirs += [ webrtc_jsoncpp_root ] + include_dirs += [ rtc_jsoncpp_root ] # When defined changes the include path for json.h to where it is # expected to be when building json outside of the standalone build. @@ -451,7 +451,7 @@ static_library("webrtc_base") { if (use_openssl) { direct_dependent_configs += [ ":openssl_config" ] - if (build_ssl) { + if (rtc_build_ssl) { deps += [ "//third_party/boringssl" ] } else { configs += [ "external_ssl_library" ] @@ -479,7 +479,7 @@ static_library("webrtc_base") { if (is_ios) { all_dependent_configs += [ ":ios_config" ] - if (build_ssl) { + if (rtc_build_ssl) { deps += [ "//net/third_party/nss/ssl:libssl" ] } else { configs += [ "external_ssl_library" ] @@ -507,7 +507,7 @@ static_library("webrtc_base") { "dl", "rt", ] - if (build_ssl) { + if (rtc_build_ssl) { configs += [ "//third_party/nss:system_nss_no_ssl_config" ] } } @@ -600,7 +600,7 @@ static_library("webrtc_base") { } if (is_mac || is_ios || is_win) { - if (build_ssl) { + if (rtc_build_ssl) { deps += [ "//net/third_party/nss/ssl:libssl", "//third_party/nss:nspr", @@ -615,7 +615,7 @@ static_library("webrtc_base") { if (build_with_chromium) { deps += [ "//crypto:platform" ] } else { - if (build_ssl) { + if (rtc_build_ssl) { deps += [ ":linux_system_ssl" ] } else { configs += [ "external_ssl_library" ] diff --git a/build/webrtc.gni b/build/webrtc.gni index 346a0622..d46b8c64 100644 --- a/build/webrtc.gni +++ b/build/webrtc.gni @@ -15,40 +15,40 @@ declare_args() { build_with_libjingle = true # Disable this to avoid building the Opus audio codec. - include_opus = true + rtc_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" + # library that comes with WebRTC (i.e. rtc_build_json == 0). + rtc_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 = "" + # library that comes with WebRTC (i.e. rtc_build_ssl == 0). + rtc_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. - enable_video = true + rtc_enable_video = true # Selects fixed-point code where possible. - prefer_fixed_point = false + rtc_prefer_fixed_point = false # Enable data logging. Produces text files with data logged within engines # which can be easily parsed for offline processing. - enable_data_logging = false + rtc_enable_data_logging = false # Enables the use of protocol buffers for debug recordings. - enable_protobuf = true + rtc_enable_protobuf = true # Disable these to not build components which can be externally provided. - build_json = true - build_libjpeg = true - build_libyuv = true - build_libvpx = true - build_ssl = true + rtc_build_json = true + rtc_build_libjpeg = true + rtc_build_libyuv = true + rtc_build_libvpx = true + rtc_build_ssl = true # Disable by default. - have_dbus_glib = false + rtc_have_dbus_glib = false # Enable to use the Mozilla internal settings. build_with_mozilla = false @@ -59,26 +59,26 @@ declare_args() { mips_dsp_rev = 0 mips_fpu = true - enable_android_opensl = true + rtc_enable_android_opensl = true # Link-Time Optimizations. # Executes code generation at link-time instead of compile-time. # https://gcc.gnu.org/wiki/LinkTimeOptimization - use_lto = false + rtc_use_lto = false if (build_with_chromium) { # Exclude pulse audio on Chromium since its prerequisites don't require # pulse audio. - include_pulse_audio = false + rtc_include_pulse_audio = false # Exclude internal ADM since Chromium uses its own IO handling. - include_internal_audio_device = false + rtc_include_internal_audio_device = false # Exclude internal VCM in Chromium build. - include_internal_video_capture = false + rtc_include_internal_video_capture = false # Exclude internal video render module in Chromium build. - include_internal_video_render = false + rtc_include_internal_video_render = false } else { # Settings for the standalone (not-in-Chromium) build. @@ -87,31 +87,31 @@ declare_args() { # http://code.google.com/p/webrtc/issues/detail?id=163 clang_use_chrome_plugins = false - include_pulse_audio = true - include_internal_audio_device = true - include_internal_video_capture = true - include_internal_video_render = true + rtc_include_pulse_audio = true + rtc_include_internal_audio_device = true + rtc_include_internal_video_capture = true + rtc_include_internal_video_render = true } if (build_with_libjingle) { - include_tests = false - restrict_webrtc_logging = true + rtc_include_tests = false + rtc_restrict_logging = true } else { - include_tests = true - restrict_webrtc_logging = false + rtc_include_tests = true + rtc_restrict_logging = false } if (is_ios) { - build_libjpeg = false - enable_protobuf = false + rtc_build_libjpeg = false + rtc_enable_protobuf = false } if (cpu_arch == "arm") { - prefer_fixed_point = true + rtc_prefer_fixed_point = true } # WebRTC builds ARM v7 Neon instruction set optimized code for both iOS and # Android, which is why we currently cannot use the variables in # //build/config/arm.gni (since it disables Neon for Android). - build_armv7_neon = (cpu_arch == "arm" && arm_version == 7) + rtc_build_armv7_neon = (cpu_arch == "arm" && arm_version == 7) } diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn index 6b5fe9de..f9bbd6a1 100644 --- a/common_audio/BUILD.gn +++ b/common_audio/BUILD.gn @@ -170,7 +170,7 @@ if (cpu_arch == "x86" || cpu_arch == "x64") { } } -if (build_armv7_neon) { +if (rtc_build_armv7_neon) { source_set("common_audio_neon") { sources = [ "fir_filter_neon.cc", @@ -199,7 +199,7 @@ if (build_armv7_neon) { ] # Disable LTO in audio_processing_neon target due to compiler bug. - if (use_lto) { + if (rtc_use_lto) { cflags -= [ "-flto", "-ffat-lto-objects", diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn index d4325432..1b9ad8c7 100644 --- a/common_video/BUILD.gn +++ b/common_video/BUILD.gn @@ -42,7 +42,7 @@ source_set("common_video") { deps = [ "../system_wrappers" ] - if (build_libyuv) { + if (rtc_build_libyuv) { deps += [ "//third_party/libyuv" ] } else { # Need to add a directory normally exported by libyuv. diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 6e5a4ef8..8972ff9b 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -102,7 +102,7 @@ source_set("audio_coding") { "../../system_wrappers", ] - if (include_opus) { + if (rtc_include_opus) { defines += [ "WEBRTC_CODEC_OPUS" ] deps += [ ":webrtc_opus" ] } @@ -453,7 +453,7 @@ source_set("isacfix") { "../../system_wrappers", ] - if (build_armv7_neon) { + if (rtc_build_armv7_neon) { deps += [ ":isac_neon" ] # Enable compilation for the ARM v7 Neon instruction set. This is needed @@ -497,7 +497,7 @@ source_set("isacfix") { } } - if (build_armv7_neon) { + if (rtc_build_armv7_neon) { sources += [ "codecs/isac/fix/source/lattice_c.c", "codecs/isac/fix/source/pitch_estimator_c.c", @@ -505,7 +505,7 @@ source_set("isacfix") { } } -if (build_armv7_neon) { +if (rtc_build_armv7_neon) { source_set("isac_neon") { sources = [ "codecs/isac/fix/source/entropy_coding_neon.c", @@ -521,7 +521,7 @@ if (build_armv7_neon) { ] # Disable LTO in audio_processing_neon target due to compiler bug. - if (use_lto) { + if (rtc_use_lto) { cflags -= [ "-flto", "-ffat-lto-objects", diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index afc58856..60a83dc3 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -55,7 +55,7 @@ source_set("audio_device") { if (is_android) { include_dirs += [ "android" ] } - if (include_internal_audio_device) { + if (rtc_include_internal_audio_device) { sources += [ "linux/alsasymboltable_linux.cc", "linux/alsasymboltable_linux.h", @@ -118,7 +118,7 @@ source_set("audio_device") { "X11", ] - if (include_pulse_audio) { + if (rtc_include_pulse_audio) { sources += [ "linux/audio_device_pulse_linux.cc", "linux/audio_device_pulse_linux.h", diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index 57b721bd..950dfff3 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -89,12 +89,12 @@ source_set("audio_processing") { defines += [ "WEBRTC_UNTRUSTED_DELAY" ] } - if (enable_protobuf) { + if (rtc_enable_protobuf) { defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ] deps += [ ":audioproc_debug_proto" ] } - if (prefer_fixed_point) { + if (rtc_prefer_fixed_point) { defines += [ "WEBRTC_NS_FIXED" ] sources += [ "ns/include/noise_suppression_x.h", @@ -124,7 +124,7 @@ source_set("audio_processing") { deps += [ ":audio_processing_sse2" ] } - if (build_armv7_neon) { + if (rtc_build_armv7_neon) { deps += [ ":audio_processing_neon" ] } @@ -159,7 +159,7 @@ source_set("audio_processing") { ] } -if (enable_protobuf) { +if (rtc_enable_protobuf) { proto_library("audioproc_debug_proto") { sources = [ "debug.proto" ] @@ -180,7 +180,7 @@ if (cpu_arch == "x86" || cpu_arch == "x64") { } } -if (build_armv7_neon) { +if (rtc_build_armv7_neon) { source_set("audio_processing_neon") { sources = [ "aec/aec_core_neon.c", @@ -217,7 +217,7 @@ if (build_armv7_neon) { ] # Disable LTO in audio_processing_neon target due to compiler bug. - if (use_lto) { + if (rtc_use_lto) { cflags -= [ "-flto", "-ffat-lto-objects", diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn index 9856d2da..ff321127 100644 --- a/modules/utility/BUILD.gn +++ b/modules/utility/BUILD.gn @@ -42,7 +42,7 @@ source_set("utility") { "../audio_coding", "../media_file", ] - if (enable_video) { + if (rtc_enable_video) { sources += [ "source/frame_scaler.cc", "source/video_coder.cc", diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index 9c314aec..7d6ddc0e 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -35,7 +35,7 @@ source_set("video_capture") { libs = [] deps = [] - if (include_internal_video_capture) { + if (rtc_include_internal_video_capture) { if (is_linux) { sources += [ "linux/device_info_linux.cc", diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index 49df4009..42d5d5ce 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -137,7 +137,7 @@ source_set("webrtc_vp8") { "../../common_video", "../../system_wrappers", ] -# if (build_libvpx) { +# if (rtc_build_libvpx) { # deps += [ # "//third_party/libvpx", # ] diff --git a/modules/video_render/BUILD.gn b/modules/video_render/BUILD.gn index fe9259c2..2c14b546 100644 --- a/modules/video_render/BUILD.gn +++ b/modules/video_render/BUILD.gn @@ -35,7 +35,7 @@ source_set("video_render") { libs = [] deps = [] - if (include_internal_video_render) { + if (rtc_include_internal_video_render) { defines += [ "WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER" ] if (is_linux) { diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index 5b51e6f1..67c47545 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -119,7 +119,7 @@ static_library("system_wrappers") { ":system_wrappers_inherited_config", ] - if (enable_data_logging) { + if (rtc_enable_data_logging) { sources += [ "source/data_log.cc" ] } else { sources += [ "source/data_log_no_op.cc" ] |