blob: 849b71f8003ae4a626848832a7cdf8a74c93f609 [file] [log] [blame]
# Copyright (c) 2012 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.
# This file contains common settings for building WebRTC components.
{
# Nesting is required in order to use variables for setting other variables.
'variables': {
'variables': {
'variables': {
'variables': {
# This will already be set to zero by supplement.gypi
'build_with_chromium%': 1,
# Enable to use the Mozilla internal settings.
'build_with_mozilla%': 0,
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla%)',
'include_opus%': 1,
'conditions': [
# Include the iLBC audio codec?
['build_with_chromium==1 or build_with_mozilla==1', {
'include_ilbc%': 0,
}, {
'include_ilbc%': 1,
}],
['build_with_chromium==1', {
'webrtc_root%': '<(DEPTH)/third_party/webrtc',
}, {
'webrtc_root%': '<(DEPTH)/webrtc',
}],
# Controls whether we use libevent on posix platforms.
# TODO(phoglund): should arguably be controlled by platform #ifdefs
# in the code instead.
['OS=="win" or OS=="mac" or OS=="ios"', {
'build_libevent%': 0,
'enable_libevent%': 0,
}, {
'build_libevent%': 1,
'enable_libevent%': 1,
}],
],
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla)',
'build_libevent%': '<(build_libevent)',
'enable_libevent%': '<(enable_libevent)',
'webrtc_root%': '<(webrtc_root)',
'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
'webrtc_vp9_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp9',
'include_ilbc%': '<(include_ilbc)',
'include_opus%': '<(include_opus)',
'opus_dir%': '<(DEPTH)/third_party/opus',
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla)',
'build_libevent%': '<(build_libevent)',
'enable_libevent%': '<(enable_libevent)',
'webrtc_root%': '<(webrtc_root)',
'test_runner_path': '<(DEPTH)/webrtc/build/android/test_runner.py',
'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
'webrtc_vp9_dir%': '<(webrtc_vp9_dir)',
'include_ilbc%': '<(include_ilbc)',
'include_opus%': '<(include_opus)',
'rtc_relative_path%': 1,
'external_libraries%': '0',
'json_root%': '<(DEPTH)/third_party/jsoncpp/source/include/',
# openssl needs to be defined or gyp will complain. Is is only used when
# when providing external libraries so just use current directory as a
# placeholder.
'ssl_root%': '.',
# The Chromium common.gypi we use treats all gyp files without
# chromium_code==1 as third party code. This disables many of the
# preferred warning settings.
#
# We can set this here to have WebRTC code treated as Chromium code. Our
# third party code will still have the reduced warning settings.
'chromium_code': 1,
# Targets are by default not NaCl untrusted code. Use this variable exclude
# code that uses libraries that aren't available in the NaCl sandbox.
'nacl_untrusted_build%': 0,
# Set to 1 to enable code coverage on Linux using the gcov library.
'coverage%': 0,
# Set to "func", "block", "edge" for coverage generation.
# At unit test runtime set UBSAN_OPTIONS="coverage=1".
# It is recommend to set include_examples=0.
# Use llvm's sancov -html-report for human readable reports.
# See http://clang.llvm.org/docs/SanitizerCoverage.html .
'webrtc_sanitize_coverage%': "",
# Remote bitrate estimator logging/plotting.
'enable_bwe_test_logging%': 0,
# Selects fixed-point code where possible.
'prefer_fixed_point%': 0,
# Enable data logging. Produces text files with data logged within engines
# which can be easily parsed for offline processing.
'enable_data_logging%': 0,
# Enables the use of protocol buffers for debug recordings.
'enable_protobuf%': 1,
# Disable the code for the intelligibility enhancer by default.
'enable_intelligibility_enhancer%': 0,
# Selects whether debug dumps for the audio processing module
# should be generated.
'apm_debug_dump%': 0,
# Disable these to not build components which can be externally provided.
'build_expat%': 1,
'build_json%': 1,
'build_libsrtp%': 1,
'build_libvpx%': 1,
'libvpx_build_vp9%': 1,
'build_libyuv%': 1,
'build_openmax_dl%': 1,
'build_opus%': 1,
'build_protobuf%': 1,
'build_ssl%': 1,
'build_usrsctp%': 1,
# Disable by default
'have_dbus_glib%': 0,
# Make it possible to provide custom locations for some libraries.
'libvpx_dir%': '<(DEPTH)/third_party/libvpx',
'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
'opus_dir%': '<(opus_dir)',
# Use Java based audio layer as default for Android.
# Change this setting to 1 to use Open SL audio instead.
# TODO(henrika): add support for Open SL ES.
'enable_android_opensl%': 0,
# Link-Time Optimizations
# Executes code generation at link-time instead of compile-time
# https://gcc.gnu.org/wiki/LinkTimeOptimization
'use_lto%': 0,
# Defer ssl perference to that specified through sslconfig.h instead of
# choosing openssl or nss directly. In practice, this can be used to
# enable schannel on windows.
'use_legacy_ssl_defaults%': 0,
# Determines whether NEON code will be built.
'build_with_neon%': 0,
# Disable this to skip building source requiring GTK.
'use_gtk%': 1,
# Enable this to prevent extern symbols from being hidden on iOS builds.
# The chromium settings we inherit hide symbols by default on Release
# builds. We want our symbols to be visible when distributing WebRTC via
# static libraries to avoid linker warnings.
'ios_override_visibility%': 0,
# Determines whether QUIC code will be built.
'use_quic%': 0,
# By default, use normal platform audio support or dummy audio, but don't
# use file-based audio playout and record.
'use_dummy_audio_file_devices%': 0,
'conditions': [
# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported
# on all platforms except Android and iOS. Because FFmpeg can be built
# with/without H.264 support, |ffmpeg_branding| has to separately be set
# to a value that includes H.264, for example "Chrome". If FFmpeg is built
# without H.264, compilation succeeds but |H264DecoderImpl| fails to
# initialize. See also: |rtc_initialize_ffmpeg|.
# CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
# http://www.openh264.org, https://www.ffmpeg.org/
['proprietary_codecs==1 and OS!="android" and OS!="ios"', {
'rtc_use_h264%': 1,
}, {
'rtc_use_h264%': 0,
}],
# 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.
['build_with_chromium==0', {
'rtc_initialize_ffmpeg%': 1,
}, {
'rtc_initialize_ffmpeg%': 0,
}],
['build_with_chromium==1', {
# Build sources requiring GTK. NOTICE: This is not present in Chrome OS
# build environments, even if available for Chromium builds.
'use_gtk%': 0,
# Exclude pulse audio on Chromium since its prerequisites don't require
# pulse audio.
'include_pulse_audio%': 0,
# Exclude internal ADM since Chromium uses its own IO handling.
'include_internal_audio_device%': 0,
# Remove tests for Chromium to avoid slowing down GYP generation.
'include_tests%': 0,
'restrict_webrtc_logging%': 1,
}, { # Settings for the standalone (not-in-Chromium) build.
'use_gtk%': 1,
# TODO(andrew): For now, disable the Chrome plugins, which causes a
# flood of chromium-style warnings. Investigate enabling them:
# http://code.google.com/p/webrtc/issues/detail?id=163
'clang_use_chrome_plugins%': 0,
'include_pulse_audio%': 1,
'include_internal_audio_device%': 1,
'include_tests%': 1,
'restrict_webrtc_logging%': 0,
}],
['target_arch=="arm" or target_arch=="arm64" or target_arch=="mipsel"', {
'prefer_fixed_point%': 1,
}],
['(target_arch=="arm" and arm_neon==1) or target_arch=="arm64"', {
'build_with_neon%': 1,
}],
['OS!="ios" and (target_arch!="arm" or arm_version>=7) and target_arch!="mips64el"', {
'rtc_use_openmax_dl%': 1,
}, {
'rtc_use_openmax_dl%': 0,
}],
], # conditions
},
'target_defaults': {
'conditions': [
['restrict_webrtc_logging==1', {
'defines': ['WEBRTC_RESTRICT_LOGGING',],
}],
['build_with_mozilla==1', {
'defines': [
# Changes settings for Mozilla build.
'WEBRTC_MOZILLA_BUILD',
],
}],
['have_dbus_glib==1', {
'defines': [
'HAVE_DBUS_GLIB',
],
'cflags': [
'<!@(pkg-config --cflags dbus-glib-1)',
],
}],
['rtc_relative_path==1', {
'defines': ['EXPAT_RELATIVE_PATH',],
}],
['os_posix==1', {
'configurations': {
'Debug_Base': {
'defines': [
# Chromium's build/common.gypi defines _DEBUG for all posix
# _except_ for ios & mac. The size of data types such as
# pthread_mutex_t varies between release and debug builds
# and is controlled via this flag. Since we now share code
# between base/base.gyp and build/common.gypi (this file),
# both gyp(i) files, must consistently set this flag uniformly
# or else we'll run in to hard-to-figure-out problems where
# one compilation unit uses code from another but expects
# differently laid out types.
# For WebRTC, we want it there as well, because ASSERT and
# friends trigger off of it.
'_DEBUG',
],
},
},
}],
['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.
'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
# use full paths (e.g. headers inside testing/ or third_party/).
'<(DEPTH)',
# The overrides must be included before the WebRTC root as that's the
# mechanism for selecting the override headers in Chromium.
'../../webrtc_overrides',
# The WebRTC root is needed to allow includes in the WebRTC code base
# to be prefixed with webrtc/.
'../..',
],
}, {
'includes': [
# Rules for excluding e.g. foo_win.cc from the build on non-Windows.
'filename_rules.gypi',
],
# Include the top-level dir so the WebRTC code can use full paths.
'include_dirs': [
'../..',
],
'conditions': [
['os_posix==1', {
'conditions': [
# -Wextra is currently disabled in Chromium's common.gypi. Enable
# for targets that can handle it. For Android/arm64 right now
# there will be an 'enumeral and non-enumeral type in conditional
# expression' warning in android_tools/ndk_experimental's version
# of stlport.
# See: https://code.google.com/p/chromium/issues/detail?id=379699
['target_arch!="arm64" or OS!="android"', {
'cflags': [
'-Wextra',
# We need to repeat some flags from Chromium's common.gypi
# here that get overridden by -Wextra.
'-Wno-unused-parameter',
'-Wno-missing-field-initializers',
'-Wno-strict-overflow',
],
}],
],
'cflags_cc': [
'-Wnon-virtual-dtor',
# This is enabled for clang; enable for gcc as well.
'-Woverloaded-virtual',
],
}],
['clang==1', {
'cflags': [
'-Wimplicit-fallthrough',
'-Wthread-safety',
'-Winconsistent-missing-override',
],
}],
],
}],
['target_arch=="arm64"', {
'defines': [
'WEBRTC_ARCH_ARM64',
'WEBRTC_HAS_NEON',
],
}],
['target_arch=="arm"', {
'defines': [
'WEBRTC_ARCH_ARM',
],
'conditions': [
['arm_version>=7', {
'defines': ['WEBRTC_ARCH_ARM_V7',],
'conditions': [
['arm_neon==1', {
'defines': ['WEBRTC_HAS_NEON',],
}],
],
}],
],
}],
['target_arch=="mipsel" and mips_arch_variant!="r6"', {
'defines': [
'MIPS32_LE',
],
'conditions': [
['mips_float_abi=="hard"', {
'defines': [
'MIPS_FPU_LE',
],
}],
['mips_arch_variant=="r2"', {
'defines': [
'MIPS32_R2_LE',
],
}],
['mips_dsp_rev==1', {
'defines': [
'MIPS_DSP_R1_LE',
],
}],
['mips_dsp_rev==2', {
'defines': [
'MIPS_DSP_R1_LE',
'MIPS_DSP_R2_LE',
],
}],
],
}],
['coverage==1 and OS=="linux"', {
'cflags': [ '-ftest-coverage',
'-fprofile-arcs' ],
'ldflags': [ '--coverage' ],
'link_settings': { 'libraries': [ '-lgcov' ] },
}],
['webrtc_sanitize_coverage!=""', {
'cflags': [ '-fsanitize-coverage=<(webrtc_sanitize_coverage)' ],
'ldflags': [ '-fsanitize-coverage=<(webrtc_sanitize_coverage)' ],
}],
['webrtc_sanitize_coverage!="" and OS=="mac"', {
'xcode_settings': {
'OTHER_CFLAGS': [
'-fsanitize-coverage=func',
],
},
}],
['os_posix==1', {
# For access to standard POSIXish features, use WEBRTC_POSIX instead of
# a more specific macro.
'defines': [
'WEBRTC_POSIX',
],
}],
['OS=="ios"', {
'defines': [
'WEBRTC_MAC',
'WEBRTC_IOS',
],
}],
['OS=="ios" and ios_override_visibility==1', {
'xcode_settings': {
'GCC_INLINES_ARE_PRIVATE_EXTERN': 'NO',
'GCC_SYMBOLS_PRIVATE_EXTERN': 'NO',
}
}],
['OS=="linux"', {
'defines': [
'WEBRTC_LINUX',
],
}],
['OS=="mac"', {
'defines': [
'WEBRTC_MAC',
],
}],
['OS=="win"', {
'defines': [
'WEBRTC_WIN',
],
# TODO(andrew): enable all warnings when possible.
# TODO(phoglund): get rid of 4373 supression when
# http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
'msvs_disabled_warnings': [
4373, # legacy warning for ignoring const / volatile in signatures.
4389, # Signed/unsigned mismatch.
],
# Re-enable some warnings that Chromium disables.
'msvs_disabled_warnings!': [4189,],
}],
['OS=="android"', {
'defines': [
'WEBRTC_LINUX',
'WEBRTC_ANDROID',
],
'conditions': [
['clang==0', {
# The Android NDK doesn't provide optimized versions of these
# functions. Ensure they are disabled for all compilers.
'cflags': [
'-fno-builtin-cos',
'-fno-builtin-sin',
'-fno-builtin-cosf',
'-fno-builtin-sinf',
],
}],
],
}],
['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',
],
}],
['libvpx_build_vp9==0', {
'defines': [
'RTC_DISABLE_VP9',
],
}],
], # conditions
'direct_dependent_settings': {
'conditions': [
['build_with_mozilla==1', {
'defines': [
# Changes settings for Mozilla build.
'WEBRTC_MOZILLA_BUILD',
],
}],
['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': [
# The overrides must be included first as that is the mechanism for
# selecting the override headers in Chromium.
'../../webrtc_overrides',
'../..',
],
}, {
'include_dirs': [
'../..',
],
}],
['OS=="mac"', {
'defines': [
'WEBRTC_MAC',
],
}],
['OS=="ios"', {
'defines': [
'WEBRTC_MAC',
'WEBRTC_IOS',
],
}],
['OS=="win"', {
'defines': [
'WEBRTC_WIN',
'_CRT_SECURE_NO_WARNINGS', # Suppress warnings about _vsnprinf
],
}],
['OS=="linux"', {
'defines': [
'WEBRTC_LINUX',
],
}],
['OS=="android"', {
'defines': [
'WEBRTC_LINUX',
'WEBRTC_ANDROID',
],
}],
['os_posix==1', {
# For access to standard POSIXish features, use WEBRTC_POSIX instead
# of a more specific macro.
'defines': [
'WEBRTC_POSIX',
],
}],
['chromeos==1', {
'defines': [
'CHROMEOS',
],
}],
['os_bsd==1', {
'defines': [
'BSD',
],
}],
['OS=="openbsd"', {
'defines': [
'OPENBSD',
],
}],
['OS=="freebsd"', {
'defines': [
'FREEBSD',
],
}],
],
},
}, # target_defaults
}