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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
2 # | 2 # |
3 # Use of this source code is governed by a BSD-style license | 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 | 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 | 5 # tree. An additional intellectual property rights grant can be found |
6 # in the file PATENTS. All contributing project authors may | 6 # in the file PATENTS. All contributing project authors may |
7 # be found in the AUTHORS file in the root of the source tree. | 7 # be found in the AUTHORS file in the root of the source tree. |
8 | 8 |
9 import("//build/config/arm.gni") | 9 import("../webrtc.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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