webrtc/build/common.gypi - src - Git at Google

 # 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,

     # 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 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,

     # 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/.
           '../..',
         ],
       }, {
          # 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=="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
 }
	# 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,

	# 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 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,

	# 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/.
	'../..',
	],
	}, {
	# 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=="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
	}