| # 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', |
| 'android_tests_path%': '<(DEPTH)/third_party/webrtc/build/android_tests_noop.gyp', |
| }, { |
| 'webrtc_root%': '<(DEPTH)/webrtc', |
| 'android_tests_path%': '<(DEPTH)/webrtc/build/android_tests.gyp', |
| }], |
| |
| # 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)', |
| 'android_tests_path%': '<(android_tests_path)', |
| '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)', |
| 'android_tests_path%': '<(android_tests_path)', |
| '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, |
| |
| # 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 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 use HW H.264 encoder/decoder on iOS/Mac PeerConnections. |
| # Enabling this may break interop with Android clients that support H264. |
| 'use_objc_h264%': 0, |
| |
| # 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, |
| |
| '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/. |
| '../..', |
| ], |
| }, { |
| # 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', |
| ], |
| }], |
| ], |
| }], |
| ['enable_libevent==1', { |
| 'defines': [ |
| 'WEBRTC_BUILD_LIBEVENT', |
| ], |
| }], |
| ['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' ] }, |
| }], |
| ['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=="ios" and use_objc_h264==1', { |
| 'defines': [ |
| 'WEBRTC_OBJC_H264', |
| ], |
| }], |
| ['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 |
| } |