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1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2016 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/linux/pkg_config.gni") | 9 import("//build/config/linux/pkg_config.gni") |
10 import("../build/webrtc.gni") | 10 import("../build/webrtc.gni") |
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136 "engine/webrtcvideocapturerfactory.h", | 136 "engine/webrtcvideocapturerfactory.h", |
137 "engine/webrtcvideodecoderfactory.h", | 137 "engine/webrtcvideodecoderfactory.h", |
138 "engine/webrtcvideoencoderfactory.cc", | 138 "engine/webrtcvideoencoderfactory.cc", |
139 "engine/webrtcvideoencoderfactory.h", | 139 "engine/webrtcvideoencoderfactory.h", |
140 "engine/webrtcvideoengine2.cc", | 140 "engine/webrtcvideoengine2.cc", |
141 "engine/webrtcvideoengine2.h", | 141 "engine/webrtcvideoengine2.h", |
142 "engine/webrtcvideoframe.h", | 142 "engine/webrtcvideoframe.h", |
143 "engine/webrtcvoe.h", | 143 "engine/webrtcvoe.h", |
144 "engine/webrtcvoiceengine.cc", | 144 "engine/webrtcvoiceengine.cc", |
145 "engine/webrtcvoiceengine.h", | 145 "engine/webrtcvoiceengine.h", |
146 "sctp/sctpdataengine.cc", | |
147 "sctp/sctpdataengine.h", | |
148 ] | 146 ] |
149 | 147 |
| 148 if (rtc_enable_sctp) { |
| 149 sources += [ |
| 150 "sctp/sctpdataengine.cc", |
| 151 "sctp/sctpdataengine.h", |
| 152 ] |
| 153 } |
| 154 |
150 configs += [ ":rtc_media_warnings_config" ] | 155 configs += [ ":rtc_media_warnings_config" ] |
151 | 156 |
152 if (!build_with_chromium && is_clang) { | 157 if (!build_with_chromium && is_clang) { |
153 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
154 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
155 } | 160 } |
156 | 161 |
157 if (is_win) { | 162 if (is_win) { |
158 cflags = [ | 163 cflags = [ |
159 "/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch. | 164 "/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch. |
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172 if (rtc_build_libyuv) { | 177 if (rtc_build_libyuv) { |
173 deps += [ "$rtc_libyuv_dir" ] | 178 deps += [ "$rtc_libyuv_dir" ] |
174 public_deps = [ | 179 public_deps = [ |
175 "$rtc_libyuv_dir", | 180 "$rtc_libyuv_dir", |
176 ] | 181 ] |
177 } else { | 182 } else { |
178 # Need to add a directory normally exported by libyuv. | 183 # Need to add a directory normally exported by libyuv. |
179 include_dirs += [ "$rtc_libyuv_dir/include" ] | 184 include_dirs += [ "$rtc_libyuv_dir/include" ] |
180 } | 185 } |
181 | 186 |
182 if (rtc_build_usrsctp) { | 187 if (rtc_enable_sctp && rtc_build_usrsctp) { |
183 include_dirs += [ | 188 include_dirs += [ |
184 # TODO(jiayl): move this into the public_configs of | 189 # TODO(jiayl): move this into the public_configs of |
185 # //third_party/usrsctp/BUILD.gn. | 190 # //third_party/usrsctp/BUILD.gn. |
186 "//third_party/usrsctp/usrsctplib", | 191 "//third_party/usrsctp/usrsctplib", |
187 ] | 192 ] |
188 deps += [ "//third_party/usrsctp" ] | 193 deps += [ "//third_party/usrsctp" ] |
189 } | 194 } |
190 | 195 |
191 public_configs = [] | 196 public_configs = [] |
192 if (build_with_chromium) { | 197 if (build_with_chromium) { |
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331 "engine/internaldecoderfactory_unittest.cc", | 336 "engine/internaldecoderfactory_unittest.cc", |
332 "engine/nullwebrtcvideoengine_unittest.cc", | 337 "engine/nullwebrtcvideoengine_unittest.cc", |
333 "engine/payload_type_mapper_unittest.cc", | 338 "engine/payload_type_mapper_unittest.cc", |
334 "engine/simulcast_unittest.cc", | 339 "engine/simulcast_unittest.cc", |
335 "engine/videodecodersoftwarefallbackwrapper_unittest.cc", | 340 "engine/videodecodersoftwarefallbackwrapper_unittest.cc", |
336 "engine/videoencodersoftwarefallbackwrapper_unittest.cc", | 341 "engine/videoencodersoftwarefallbackwrapper_unittest.cc", |
337 "engine/webrtcmediaengine_unittest.cc", | 342 "engine/webrtcmediaengine_unittest.cc", |
338 "engine/webrtcvideocapturer_unittest.cc", | 343 "engine/webrtcvideocapturer_unittest.cc", |
339 "engine/webrtcvideoengine2_unittest.cc", | 344 "engine/webrtcvideoengine2_unittest.cc", |
340 "engine/webrtcvoiceengine_unittest.cc", | 345 "engine/webrtcvoiceengine_unittest.cc", |
341 "sctp/sctpdataengine_unittest.cc", | |
342 ] | 346 ] |
343 | 347 |
| 348 if (rtc_enable_sctp) { |
| 349 sources += [ "sctp/sctpdataengine_unittest.cc" ] |
| 350 } |
| 351 |
344 configs += [ ":rtc_media_unittests_config" ] | 352 configs += [ ":rtc_media_unittests_config" ] |
345 | 353 |
346 if (rtc_use_h264) { | 354 if (rtc_use_h264) { |
347 defines += [ "WEBRTC_USE_H264" ] | 355 defines += [ "WEBRTC_USE_H264" ] |
348 } | 356 } |
349 if (is_win) { | 357 if (is_win) { |
350 cflags = [ | 358 cflags = [ |
351 "/wd4245", # conversion from int to size_t, signed/unsigned mismatch. | 359 "/wd4245", # conversion from int to size_t, signed/unsigned mismatch. |
352 "/wd4373", # virtual function override. | 360 "/wd4373", # virtual function override. |
353 "/wd4389", # signed/unsigned mismatch. | 361 "/wd4389", # signed/unsigned mismatch. |
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378 # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243. | 386 # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243. |
379 ":rtc_media", | 387 ":rtc_media", |
380 ":rtc_unittest_main", | 388 ":rtc_unittest_main", |
381 "../audio", | 389 "../audio", |
382 "../base:rtc_base_tests_utils", | 390 "../base:rtc_base_tests_utils", |
383 "../modules/audio_device:mock_audio_device", | 391 "../modules/audio_device:mock_audio_device", |
384 "../system_wrappers:metrics_default", | 392 "../system_wrappers:metrics_default", |
385 ] | 393 ] |
386 } | 394 } |
387 } | 395 } |
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