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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("../webrtc.gni") | 10 import("../webrtc.gni") |
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216 deps += [ "../modules/video_capture:video_capture" ] | 216 deps += [ "../modules/video_capture:video_capture" ] |
217 } else { | 217 } else { |
218 public_configs += [ ":rtc_media_defines_config" ] | 218 public_configs += [ ":rtc_media_defines_config" ] |
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_decoder_api", |
| 227 "../api:video_encoder_api", |
226 "../api:video_frame_api", | 228 "../api:video_frame_api", |
227 "../api/audio_codecs:audio_codecs_api", | 229 "../api/audio_codecs:audio_codecs_api", |
228 "../api/audio_codecs:builtin_audio_decoder_factory", | 230 "../api/audio_codecs:builtin_audio_decoder_factory", |
229 "../base:rtc_base", | 231 "../base:rtc_base", |
230 "../base:rtc_base_approved", | 232 "../base:rtc_base_approved", |
231 "../call", | 233 "../call", |
232 "../common_video:common_video", | 234 "../common_video:common_video", |
233 "../modules/audio_coding:rent_a_codec", | 235 "../modules/audio_coding:rent_a_codec", |
234 "../modules/audio_device:audio_device", | 236 "../modules/audio_device:audio_device", |
235 "../modules/audio_mixer:audio_mixer_impl", | 237 "../modules/audio_mixer:audio_mixer_impl", |
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297 if (!build_with_chromium && is_clang) { | 299 if (!build_with_chromium && is_clang) { |
298 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 300 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
299 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 301 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
300 } | 302 } |
301 | 303 |
302 deps += [ | 304 deps += [ |
303 ":rtc_media", | 305 ":rtc_media", |
304 ":rtc_media_base", | 306 ":rtc_media_base", |
305 "..:webrtc_common", | 307 "..:webrtc_common", |
306 "../api:call_api", | 308 "../api:call_api", |
| 309 "../api:video_decoder_api", |
| 310 "../api:video_encoder_api", |
307 "../api:video_frame_api", | 311 "../api:video_frame_api", |
308 "../base:rtc_base", | 312 "../base:rtc_base", |
309 "../base:rtc_base_approved", | 313 "../base:rtc_base_approved", |
310 "../base:rtc_base_tests_main", | 314 "../base:rtc_base_tests_main", |
311 "../base:rtc_base_tests_utils", | 315 "../base:rtc_base_tests_utils", |
312 "../call:call_interfaces", | 316 "../call:call_interfaces", |
313 "../test:test_support", | 317 "../test:test_support", |
314 "//testing/gtest", | 318 "//testing/gtest", |
315 ] | 319 ] |
316 public_deps += [ "//testing/gmock" ] | 320 public_deps += [ "//testing/gmock" ] |
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429 } | 433 } |
430 | 434 |
431 if (is_ios) { | 435 if (is_ios) { |
432 deps += [ ":rtc_media_unittests_bundle_data" ] | 436 deps += [ ":rtc_media_unittests_bundle_data" ] |
433 } | 437 } |
434 | 438 |
435 deps += [ | 439 deps += [ |
436 ":rtc_media", | 440 ":rtc_media", |
437 ":rtc_media_base", | 441 ":rtc_media_base", |
438 ":rtc_unittest_main", | 442 ":rtc_unittest_main", |
| 443 "../api:video_decoder_api", |
| 444 "../api:video_encoder_api", |
439 "../api:video_frame_api", | 445 "../api:video_frame_api", |
440 "../api/audio_codecs:builtin_audio_decoder_factory", | 446 "../api/audio_codecs:builtin_audio_decoder_factory", |
441 "../audio", | 447 "../audio", |
442 "../base:rtc_base", | 448 "../base:rtc_base", |
443 "../base:rtc_base_approved", | 449 "../base:rtc_base_approved", |
444 "../base:rtc_base_tests_utils", | 450 "../base:rtc_base_tests_utils", |
445 "../call:call_interfaces", | 451 "../call:call_interfaces", |
446 "../common_video:common_video", | 452 "../common_video:common_video", |
447 "../logging:rtc_event_log_api", | 453 "../logging:rtc_event_log_api", |
448 "../modules/audio_device:mock_audio_device", | 454 "../modules/audio_device:mock_audio_device", |
449 "../modules/audio_processing:audio_processing", | 455 "../modules/audio_processing:audio_processing", |
450 "../modules/video_coding:video_coding_utility", | 456 "../modules/video_coding:video_coding_utility", |
451 "../modules/video_coding:webrtc_vp8", | 457 "../modules/video_coding:webrtc_vp8", |
452 "../p2p:rtc_p2p_unittests", | 458 "../p2p:rtc_p2p_unittests", |
453 "../system_wrappers:metrics_default", | 459 "../system_wrappers:metrics_default", |
454 "../test:test_support", | 460 "../test:test_support", |
455 "../voice_engine:voice_engine", | 461 "../voice_engine:voice_engine", |
456 ] | 462 ] |
457 } | 463 } |
458 } | 464 } |
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