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| 1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| 2 # |
| 3 # Use of this source code is governed by a BSD-style license |
| 4 # that can be found in the LICENSE file in the root of the source |
| 5 # tree. An additional intellectual property rights grant can be found |
| 6 # in the file PATENTS. All contributing project authors may |
| 7 # be found in the AUTHORS file in the root of the source tree. |
| 8 |
| 9 import("//build/config/arm.gni") |
| 10 import("//build/config/features.gni") |
| 11 import("//build/config/mips.gni") |
| 12 import("//build/config/sanitizers/sanitizers.gni") |
| 13 import("//build_overrides/build.gni") |
| 14 import("//testing/test.gni") |
| 15 |
| 16 declare_args() { |
| 17 # Disable this to avoid building the Opus audio codec. |
| 18 rtc_include_opus = true |
| 19 |
| 20 # Enable this to let the Opus audio codec change complexity on the fly. |
| 21 rtc_opus_variable_complexity = false |
| 22 |
| 23 # Disable to use absolute header paths for some libraries. |
| 24 rtc_relative_path = true |
| 25 |
| 26 # Used to specify an external Jsoncpp include path when not compiling the |
| 27 # library that comes with WebRTC (i.e. rtc_build_json == 0). |
| 28 rtc_jsoncpp_root = "//third_party/jsoncpp/source/include" |
| 29 |
| 30 # Used to specify an external OpenSSL include path when not compiling the |
| 31 # library that comes with WebRTC (i.e. rtc_build_ssl == 0). |
| 32 rtc_ssl_root = "" |
| 33 |
| 34 # Selects fixed-point code where possible. |
| 35 rtc_prefer_fixed_point = false |
| 36 |
| 37 # Enables the use of protocol buffers for debug recordings. |
| 38 rtc_enable_protobuf = true |
| 39 |
| 40 # Disable the code for the intelligibility enhancer by default. |
| 41 rtc_enable_intelligibility_enhancer = false |
| 42 |
| 43 # Enable when an external authentication mechanism is used for performing |
| 44 # packet authentication for RTP packets instead of libsrtp. |
| 45 rtc_enable_external_auth = build_with_chromium |
| 46 |
| 47 # Selects whether debug dumps for the audio processing module |
| 48 # should be generated. |
| 49 apm_debug_dump = false |
| 50 |
| 51 # Set this to true to enable BWE test logging. |
| 52 rtc_enable_bwe_test_logging = false |
| 53 |
| 54 # Set this to disable building with support for SCTP data channels. |
| 55 rtc_enable_sctp = true |
| 56 |
| 57 # Disable these to not build components which can be externally provided. |
| 58 rtc_build_expat = true |
| 59 rtc_build_json = true |
| 60 rtc_build_libjpeg = true |
| 61 rtc_build_libsrtp = true |
| 62 rtc_build_libvpx = true |
| 63 rtc_libvpx_build_vp9 = true |
| 64 rtc_build_libyuv = true |
| 65 rtc_build_openmax_dl = true |
| 66 rtc_build_opus = true |
| 67 rtc_build_ssl = true |
| 68 rtc_build_usrsctp = true |
| 69 |
| 70 # Enable to use the Mozilla internal settings. |
| 71 build_with_mozilla = false |
| 72 |
| 73 rtc_enable_android_opensl = false |
| 74 |
| 75 # Link-Time Optimizations. |
| 76 # Executes code generation at link-time instead of compile-time. |
| 77 # https://gcc.gnu.org/wiki/LinkTimeOptimization |
| 78 rtc_use_lto = false |
| 79 |
| 80 # Set to "func", "block", "edge" for coverage generation. |
| 81 # At unit test runtime set UBSAN_OPTIONS="coverage=1". |
| 82 # It is recommend to set include_examples=0. |
| 83 # Use llvm's sancov -html-report for human readable reports. |
| 84 # See http://clang.llvm.org/docs/SanitizerCoverage.html . |
| 85 rtc_sanitize_coverage = "" |
| 86 |
| 87 # Enable libevent task queues on platforms that support it. |
| 88 if (is_win || is_mac || is_ios || is_nacl) { |
| 89 rtc_enable_libevent = false |
| 90 rtc_build_libevent = false |
| 91 } else { |
| 92 rtc_enable_libevent = true |
| 93 rtc_build_libevent = true |
| 94 } |
| 95 |
| 96 if (current_cpu == "arm" || current_cpu == "arm64") { |
| 97 rtc_prefer_fixed_point = true |
| 98 } |
| 99 |
| 100 if (!is_ios && (current_cpu != "arm" || arm_version >= 7) && |
| 101 current_cpu != "mips64el") { |
| 102 rtc_use_openmax_dl = true |
| 103 } else { |
| 104 rtc_use_openmax_dl = false |
| 105 } |
| 106 |
| 107 # Determines whether NEON code will be built. |
| 108 rtc_build_with_neon = |
| 109 (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64" |
| 110 |
| 111 # Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on |
| 112 # all platforms except Android and iOS. Because FFmpeg can be built |
| 113 # with/without H.264 support, |ffmpeg_branding| has to separately be set to a |
| 114 # value that includes H.264, for example "Chrome". If FFmpeg is built without |
| 115 # H.264, compilation succeeds but |H264DecoderImpl| fails to initialize. See |
| 116 # also: |rtc_initialize_ffmpeg|. |
| 117 # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING. |
| 118 # http://www.openh264.org, https://www.ffmpeg.org/ |
| 119 rtc_use_h264 = proprietary_codecs && !is_android && !is_ios |
| 120 |
| 121 # Determines whether QUIC code will be built. |
| 122 rtc_use_quic = false |
| 123 |
| 124 # By default, use normal platform audio support or dummy audio, but don't |
| 125 # use file-based audio playout and record. |
| 126 rtc_use_dummy_audio_file_devices = false |
| 127 |
| 128 # When set to true, test targets will declare the files needed to run memcheck |
| 129 # as data dependencies. This is to enable memcheck execution on swarming bots. |
| 130 rtc_use_memcheck = false |
| 131 |
| 132 # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done |
| 133 # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must |
| 134 # only be initialized once. Projects that initialize FFmpeg externally, such |
| 135 # as Chromium, must turn this flag off so that WebRTC does not also |
| 136 # initialize. |
| 137 rtc_initialize_ffmpeg = !build_with_chromium |
| 138 |
| 139 # Build sources requiring GTK. NOTICE: This is not present in Chrome OS |
| 140 # build environments, even if available for Chromium builds. |
| 141 rtc_use_gtk = !build_with_chromium |
| 142 } |
| 143 |
| 144 # A second declare_args block, so that declarations within it can |
| 145 # depend on the possibly overridden variables in the first |
| 146 # declare_args block. |
| 147 declare_args() { |
| 148 # Include the iLBC audio codec? |
| 149 rtc_include_ilbc = !(build_with_chromium || build_with_mozilla) |
| 150 |
| 151 rtc_restrict_logging = build_with_chromium |
| 152 |
| 153 # Excluded in Chromium since its prerequisites don't require Pulse Audio. |
| 154 rtc_include_pulse_audio = !build_with_chromium |
| 155 |
| 156 # Chromium uses its own IO handling, so the internal ADM is only built for |
| 157 # standalone WebRTC. |
| 158 rtc_include_internal_audio_device = !build_with_chromium |
| 159 |
| 160 # Include tests in standalone checkout. |
| 161 rtc_include_tests = !build_with_chromium |
| 162 } |
| 163 |
| 164 # Make it possible to provide custom locations for some libraries (move these |
| 165 # up into declare_args should we need to actually use them for the GN build). |
| 166 rtc_libvpx_dir = "//third_party/libvpx" |
| 167 rtc_libyuv_dir = "//third_party/libyuv" |
| 168 rtc_opus_dir = "//third_party/opus" |
| 169 |
| 170 # Desktop capturer is supported only on Windows, OSX and Linux. |
| 171 rtc_desktop_capture_supported = is_win || is_mac || is_linux |
| 172 |
| 173 ############################################################################### |
| 174 # Templates |
| 175 # |
| 176 |
| 177 # Points to //webrtc/ in webrtc stand-alone or to //third_party/webrtc/ in |
| 178 # chromium. |
| 179 # We need absolute paths for all configs in templates as they are shared in |
| 180 # different subdirectories. |
| 181 webrtc_root = get_path_info("../", "abspath") |
| 182 |
| 183 # Global configuration that should be applied to all WebRTC targets. |
| 184 # You normally shouldn't need to include this in your target as it's |
| 185 # automatically included when using the rtc_* templates. |
| 186 # It sets defines, include paths and compilation warnings accordingly, |
| 187 # both for WebRTC stand-alone builds and for the scenario when WebRTC |
| 188 # native code is built as part of Chromium. |
| 189 rtc_common_configs = [ webrtc_root + ":common_config" ] |
| 190 |
| 191 # Global public configuration that should be applied to all WebRTC targets. You |
| 192 # normally shouldn't need to include this in your target as it's automatically |
| 193 # included when using the rtc_* templates. It set the defines, include paths and |
| 194 # compilation warnings that should be propagated to dependents of the targets |
| 195 # depending on the target having this config. |
| 196 rtc_common_inherited_config = webrtc_root + ":common_inherited_config" |
| 197 |
| 198 # Common configs to remove or add in all rtc targets. |
| 199 rtc_remove_configs = [] |
| 200 rtc_add_configs = rtc_common_configs |
| 201 |
| 202 set_defaults("rtc_test") { |
| 203 configs = rtc_add_configs |
| 204 suppressed_configs = [] |
| 205 } |
| 206 |
| 207 set_defaults("rtc_source_set") { |
| 208 configs = rtc_add_configs |
| 209 suppressed_configs = [] |
| 210 } |
| 211 |
| 212 set_defaults("rtc_executable") { |
| 213 configs = rtc_add_configs |
| 214 suppressed_configs = [] |
| 215 } |
| 216 |
| 217 set_defaults("rtc_static_library") { |
| 218 configs = rtc_add_configs |
| 219 suppressed_configs = [] |
| 220 } |
| 221 |
| 222 set_defaults("rtc_shared_library") { |
| 223 configs = rtc_add_configs |
| 224 suppressed_configs = [] |
| 225 } |
| 226 |
| 227 template("rtc_test") { |
| 228 test(target_name) { |
| 229 forward_variables_from(invoker, |
| 230 "*", |
| 231 [ |
| 232 "configs", |
| 233 "public_configs", |
| 234 "suppressed_configs", |
| 235 ]) |
| 236 configs += invoker.configs |
| 237 configs -= rtc_remove_configs |
| 238 configs -= invoker.suppressed_configs |
| 239 public_configs = [ rtc_common_inherited_config ] |
| 240 if (defined(invoker.public_configs)) { |
| 241 public_configs += invoker.public_configs |
| 242 } |
| 243 } |
| 244 } |
| 245 |
| 246 template("rtc_source_set") { |
| 247 source_set(target_name) { |
| 248 forward_variables_from(invoker, |
| 249 "*", |
| 250 [ |
| 251 "configs", |
| 252 "public_configs", |
| 253 "suppressed_configs", |
| 254 ]) |
| 255 configs += invoker.configs |
| 256 configs -= rtc_remove_configs |
| 257 configs -= invoker.suppressed_configs |
| 258 public_configs = [ rtc_common_inherited_config ] |
| 259 if (defined(invoker.public_configs)) { |
| 260 public_configs += invoker.public_configs |
| 261 } |
| 262 } |
| 263 } |
| 264 |
| 265 template("rtc_executable") { |
| 266 executable(target_name) { |
| 267 forward_variables_from(invoker, |
| 268 "*", |
| 269 [ |
| 270 "deps", |
| 271 "configs", |
| 272 "public_configs", |
| 273 "suppressed_configs", |
| 274 ]) |
| 275 configs += invoker.configs |
| 276 configs -= rtc_remove_configs |
| 277 configs -= invoker.suppressed_configs |
| 278 deps = [ |
| 279 "//build/config/sanitizers:deps", |
| 280 ] |
| 281 deps += invoker.deps |
| 282 public_configs = [ rtc_common_inherited_config ] |
| 283 if (defined(invoker.public_configs)) { |
| 284 public_configs += invoker.public_configs |
| 285 } |
| 286 } |
| 287 } |
| 288 |
| 289 template("rtc_static_library") { |
| 290 static_library(target_name) { |
| 291 forward_variables_from(invoker, |
| 292 "*", |
| 293 [ |
| 294 "configs", |
| 295 "public_configs", |
| 296 "suppressed_configs", |
| 297 ]) |
| 298 configs += invoker.configs |
| 299 configs -= rtc_remove_configs |
| 300 configs -= invoker.suppressed_configs |
| 301 public_configs = [ rtc_common_inherited_config ] |
| 302 if (defined(invoker.public_configs)) { |
| 303 public_configs += invoker.public_configs |
| 304 } |
| 305 } |
| 306 } |
| 307 |
| 308 template("rtc_shared_library") { |
| 309 shared_library(target_name) { |
| 310 forward_variables_from(invoker, |
| 311 "*", |
| 312 [ |
| 313 "configs", |
| 314 "public_configs", |
| 315 "suppressed_configs", |
| 316 ]) |
| 317 configs += invoker.configs |
| 318 configs -= rtc_remove_configs |
| 319 configs -= invoker.suppressed_configs |
| 320 public_configs = [ rtc_common_inherited_config ] |
| 321 if (defined(invoker.public_configs)) { |
| 322 public_configs += invoker.public_configs |
| 323 } |
| 324 } |
| 325 } |
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