OLD | NEW |
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("../webrtc.gni") | 10 import("../webrtc.gni") |
(...skipping 208 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
219 deps += [ "../modules/video_capture:video_capture_internal_impl" ] | 219 deps += [ "../modules/video_capture:video_capture_internal_impl" ] |
220 } | 220 } |
221 deps += [ | 221 deps += [ |
222 ":rtc_media_base", | 222 ":rtc_media_base", |
223 "..:webrtc_common", | 223 "..:webrtc_common", |
224 "../api:call_api", | 224 "../api:call_api", |
225 "../api:transport_api", | 225 "../api:transport_api", |
226 "../api:video_frame_api", | 226 "../api:video_frame_api", |
227 "../api/audio_codecs:audio_codecs_api", | 227 "../api/audio_codecs:audio_codecs_api", |
228 "../api/audio_codecs:builtin_audio_decoder_factory", | 228 "../api/audio_codecs:builtin_audio_decoder_factory", |
229 "../api/video_codecs:video_codecs_api", | |
230 "../base:rtc_base", | 229 "../base:rtc_base", |
231 "../base:rtc_base_approved", | 230 "../base:rtc_base_approved", |
232 "../call", | 231 "../call", |
233 "../common_video:common_video", | 232 "../common_video:common_video", |
234 "../modules/audio_coding:rent_a_codec", | 233 "../modules/audio_coding:rent_a_codec", |
235 "../modules/audio_device:audio_device", | 234 "../modules/audio_device:audio_device", |
236 "../modules/audio_mixer:audio_mixer_impl", | 235 "../modules/audio_mixer:audio_mixer_impl", |
237 "../modules/audio_processing:audio_processing", | 236 "../modules/audio_processing:audio_processing", |
238 "../modules/video_capture:video_capture_module", | 237 "../modules/video_capture:video_capture_module", |
239 "../modules/video_coding", | 238 "../modules/video_coding", |
(...skipping 59 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
299 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 298 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
300 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 299 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
301 } | 300 } |
302 | 301 |
303 deps += [ | 302 deps += [ |
304 ":rtc_media", | 303 ":rtc_media", |
305 ":rtc_media_base", | 304 ":rtc_media_base", |
306 "..:webrtc_common", | 305 "..:webrtc_common", |
307 "../api:call_api", | 306 "../api:call_api", |
308 "../api:video_frame_api", | 307 "../api:video_frame_api", |
309 "../api/video_codecs:video_codecs_api", | |
310 "../base:rtc_base", | 308 "../base:rtc_base", |
311 "../base:rtc_base_approved", | 309 "../base:rtc_base_approved", |
312 "../base:rtc_base_tests_main", | 310 "../base:rtc_base_tests_main", |
313 "../base:rtc_base_tests_utils", | 311 "../base:rtc_base_tests_utils", |
314 "../call:call_interfaces", | 312 "../call:call_interfaces", |
315 "../test:test_support", | 313 "../test:test_support", |
316 "//testing/gtest", | 314 "//testing/gtest", |
317 ] | 315 ] |
318 public_deps += [ "//testing/gmock" ] | 316 public_deps += [ "//testing/gmock" ] |
319 } | 317 } |
(...skipping 113 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
433 if (is_ios) { | 431 if (is_ios) { |
434 deps += [ ":rtc_media_unittests_bundle_data" ] | 432 deps += [ ":rtc_media_unittests_bundle_data" ] |
435 } | 433 } |
436 | 434 |
437 deps += [ | 435 deps += [ |
438 ":rtc_media", | 436 ":rtc_media", |
439 ":rtc_media_base", | 437 ":rtc_media_base", |
440 ":rtc_unittest_main", | 438 ":rtc_unittest_main", |
441 "../api:video_frame_api", | 439 "../api:video_frame_api", |
442 "../api/audio_codecs:builtin_audio_decoder_factory", | 440 "../api/audio_codecs:builtin_audio_decoder_factory", |
443 "../api/video_codecs:video_codecs_api", | |
444 "../audio", | 441 "../audio", |
445 "../base:rtc_base", | 442 "../base:rtc_base", |
446 "../base:rtc_base_approved", | 443 "../base:rtc_base_approved", |
447 "../base:rtc_base_tests_utils", | 444 "../base:rtc_base_tests_utils", |
448 "../call:call_interfaces", | 445 "../call:call_interfaces", |
449 "../common_video:common_video", | 446 "../common_video:common_video", |
450 "../logging:rtc_event_log_api", | 447 "../logging:rtc_event_log_api", |
451 "../modules/audio_device:mock_audio_device", | 448 "../modules/audio_device:mock_audio_device", |
452 "../modules/audio_processing:audio_processing", | 449 "../modules/audio_processing:audio_processing", |
453 "../modules/video_coding:video_coding_utility", | 450 "../modules/video_coding:video_coding_utility", |
454 "../modules/video_coding:webrtc_vp8", | 451 "../modules/video_coding:webrtc_vp8", |
455 "../p2p:rtc_p2p_unittests", | 452 "../p2p:rtc_p2p_unittests", |
456 "../system_wrappers:metrics_default", | 453 "../system_wrappers:metrics_default", |
457 "../test:test_support", | 454 "../test:test_support", |
458 "../voice_engine:voice_engine", | 455 "../voice_engine:voice_engine", |
459 ] | 456 ] |
460 } | 457 } |
461 } | 458 } |
OLD | NEW |