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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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| 38 "base/h264_profile_level_id.h", | 38 "base/h264_profile_level_id.h", |
| 39 ] | 39 ] |
| 40 | 40 |
| 41 if (!build_with_chromium && is_clang) { | 41 if (!build_with_chromium && is_clang) { |
| 42 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 42 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 43 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 43 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 44 } | 44 } |
| 45 | 45 |
| 46 deps = [ | 46 deps = [ |
| 47 "..:webrtc_common", | 47 "..:webrtc_common", |
| 48 "../rtc_base:rtc_base", | 48 "../base:rtc_base", |
| 49 "../rtc_base:rtc_base_approved", | 49 "../base:rtc_base_approved", |
| 50 ] | 50 ] |
| 51 } | 51 } |
| 52 | 52 |
| 53 rtc_static_library("rtc_media_base") { | 53 rtc_static_library("rtc_media_base") { |
| 54 # TODO(kjellander): Remove (bugs.webrtc.org/6828) | 54 # TODO(kjellander): Remove (bugs.webrtc.org/6828) |
| 55 # Enabling GN check triggers cyclic dependency error: | 55 # Enabling GN check triggers cyclic dependency error: |
| 56 # //webrtc/media:rtc_media_base -> | 56 # //webrtc/media:rtc_media_base -> |
| 57 # //webrtc/pc:rtc_pc_base -> | 57 # //webrtc/pc:rtc_pc_base -> |
| 58 # //webrtc/media:rtc_data -> | 58 # //webrtc/media:rtc_data -> |
| 59 # //webrtc/media:rtc_media_base | 59 # //webrtc/media:rtc_media_base |
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| 108 ] | 108 ] |
| 109 } else { | 109 } else { |
| 110 # Need to add a directory normally exported by libyuv. | 110 # Need to add a directory normally exported by libyuv. |
| 111 include_dirs += [ "$rtc_libyuv_dir/include" ] | 111 include_dirs += [ "$rtc_libyuv_dir/include" ] |
| 112 } | 112 } |
| 113 | 113 |
| 114 deps += [ | 114 deps += [ |
| 115 ":rtc_h264_profile_id", | 115 ":rtc_h264_profile_id", |
| 116 "..:webrtc_common", | 116 "..:webrtc_common", |
| 117 "../api:libjingle_peerconnection_api", | 117 "../api:libjingle_peerconnection_api", |
| 118 "../base:rtc_base", |
| 119 "../base:rtc_base_approved", |
| 118 "../p2p", | 120 "../p2p", |
| 119 "../rtc_base:rtc_base", | |
| 120 "../rtc_base:rtc_base_approved", | |
| 121 ] | 121 ] |
| 122 | 122 |
| 123 if (is_nacl) { | 123 if (is_nacl) { |
| 124 deps += [ "//native_client_sdk/src/libraries/nacl_io" ] | 124 deps += [ "//native_client_sdk/src/libraries/nacl_io" ] |
| 125 } | 125 } |
| 126 } | 126 } |
| 127 | 127 |
| 128 rtc_static_library("rtc_audio_video") { | 128 rtc_static_library("rtc_audio_video") { |
| 129 defines = [] | 129 defines = [] |
| 130 libs = [] | 130 libs = [] |
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| 220 "..:video_stream_api", | 220 "..:video_stream_api", |
| 221 "..:webrtc_common", | 221 "..:webrtc_common", |
| 222 "../api:call_api", | 222 "../api:call_api", |
| 223 "../api:libjingle_peerconnection_api", | 223 "../api:libjingle_peerconnection_api", |
| 224 "../api:transport_api", | 224 "../api:transport_api", |
| 225 "../api:video_frame_api", | 225 "../api:video_frame_api", |
| 226 "../api/audio_codecs:audio_codecs_api", | 226 "../api/audio_codecs:audio_codecs_api", |
| 227 "../api/audio_codecs:builtin_audio_decoder_factory", | 227 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 228 "../api/audio_codecs:builtin_audio_encoder_factory", | 228 "../api/audio_codecs:builtin_audio_encoder_factory", |
| 229 "../api/video_codecs:video_codecs_api", | 229 "../api/video_codecs:video_codecs_api", |
| 230 "../base:rtc_base", |
| 231 "../base:rtc_base_approved", |
| 232 "../base:rtc_task_queue", |
| 233 "../base:sequenced_task_checker", |
| 230 "../call", | 234 "../call", |
| 231 "../common_video:common_video", | 235 "../common_video:common_video", |
| 232 "../modules/audio_coding:rent_a_codec", | 236 "../modules/audio_coding:rent_a_codec", |
| 233 "../modules/audio_device:audio_device", | 237 "../modules/audio_device:audio_device", |
| 234 "../modules/audio_mixer:audio_mixer_impl", | 238 "../modules/audio_mixer:audio_mixer_impl", |
| 235 "../modules/audio_processing:audio_processing", | 239 "../modules/audio_processing:audio_processing", |
| 236 "../modules/audio_processing/aec_dump", | 240 "../modules/audio_processing/aec_dump", |
| 237 "../modules/video_capture:video_capture_module", | 241 "../modules/video_capture:video_capture_module", |
| 238 "../modules/video_coding", | 242 "../modules/video_coding", |
| 239 "../modules/video_coding:webrtc_h264", | 243 "../modules/video_coding:webrtc_h264", |
| 240 "../modules/video_coding:webrtc_vp8", | 244 "../modules/video_coding:webrtc_vp8", |
| 241 "../modules/video_coding:webrtc_vp9", | 245 "../modules/video_coding:webrtc_vp9", |
| 242 "../p2p:rtc_p2p", | 246 "../p2p:rtc_p2p", |
| 243 "../pc:rtc_pc_base", | 247 "../pc:rtc_pc_base", |
| 244 "../rtc_base:rtc_base", | |
| 245 "../rtc_base:rtc_base_approved", | |
| 246 "../rtc_base:rtc_task_queue", | |
| 247 "../rtc_base:sequenced_task_checker", | |
| 248 "../system_wrappers", | 248 "../system_wrappers", |
| 249 "../video", | 249 "../video", |
| 250 "../voice_engine", | 250 "../voice_engine", |
| 251 ] | 251 ] |
| 252 } | 252 } |
| 253 | 253 |
| 254 rtc_static_library("rtc_data") { | 254 rtc_static_library("rtc_data") { |
| 255 defines = [] | 255 defines = [] |
| 256 deps = [] | 256 deps = [] |
| 257 | 257 |
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| 285 "//third_party/usrsctp/usrsctplib", | 285 "//third_party/usrsctp/usrsctplib", |
| 286 ] | 286 ] |
| 287 deps += [ "//third_party/usrsctp" ] | 287 deps += [ "//third_party/usrsctp" ] |
| 288 } | 288 } |
| 289 | 289 |
| 290 deps += [ | 290 deps += [ |
| 291 ":rtc_media_base", | 291 ":rtc_media_base", |
| 292 "..:webrtc_common", | 292 "..:webrtc_common", |
| 293 "../api:call_api", | 293 "../api:call_api", |
| 294 "../api:transport_api", | 294 "../api:transport_api", |
| 295 "../base:rtc_base", |
| 296 "../base:rtc_base_approved", |
| 295 "../p2p:rtc_p2p", | 297 "../p2p:rtc_p2p", |
| 296 "../rtc_base:rtc_base", | |
| 297 "../rtc_base:rtc_base_approved", | |
| 298 "../system_wrappers", | 298 "../system_wrappers", |
| 299 ] | 299 ] |
| 300 } | 300 } |
| 301 | 301 |
| 302 rtc_source_set("rtc_media") { | 302 rtc_source_set("rtc_media") { |
| 303 public_deps = [ | 303 public_deps = [ |
| 304 ":rtc_audio_video", | 304 ":rtc_audio_video", |
| 305 ":rtc_data", | 305 ":rtc_data", |
| 306 ] | 306 ] |
| 307 } | 307 } |
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| 361 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 361 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 362 } | 362 } |
| 363 | 363 |
| 364 deps += [ | 364 deps += [ |
| 365 ":rtc_media", | 365 ":rtc_media", |
| 366 ":rtc_media_base", | 366 ":rtc_media_base", |
| 367 "..:webrtc_common", | 367 "..:webrtc_common", |
| 368 "../api:call_api", | 368 "../api:call_api", |
| 369 "../api:video_frame_api", | 369 "../api:video_frame_api", |
| 370 "../api/video_codecs:video_codecs_api", | 370 "../api/video_codecs:video_codecs_api", |
| 371 "../base:rtc_base", |
| 372 "../base:rtc_base_approved", |
| 373 "../base:rtc_base_tests_utils", |
| 371 "../call:call_interfaces", | 374 "../call:call_interfaces", |
| 372 "../rtc_base:rtc_base", | |
| 373 "../rtc_base:rtc_base_approved", | |
| 374 "../rtc_base:rtc_base_tests_utils", | |
| 375 "../test:test_support", | 375 "../test:test_support", |
| 376 "//testing/gtest", | 376 "//testing/gtest", |
| 377 ] | 377 ] |
| 378 public_deps += [ "//testing/gmock" ] | 378 public_deps += [ "//testing/gmock" ] |
| 379 } | 379 } |
| 380 | 380 |
| 381 config("rtc_media_unittests_config") { | 381 config("rtc_media_unittests_config") { |
| 382 # GN orders flags on a target before flags from configs. The default config | 382 # GN orders flags on a target before flags from configs. The default config |
| 383 # adds -Wall, and this flag have to be after -Wall -- so they need to | 383 # adds -Wall, and this flag have to be after -Wall -- so they need to |
| 384 # come from a config and can"t be on the target directly. | 384 # come from a config and can"t be on the target directly. |
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| 501 | 501 |
| 502 deps += [ | 502 deps += [ |
| 503 ":rtc_media", | 503 ":rtc_media", |
| 504 ":rtc_media_base", | 504 ":rtc_media_base", |
| 505 ":rtc_media_tests_utils", | 505 ":rtc_media_tests_utils", |
| 506 "../api:video_frame_api", | 506 "../api:video_frame_api", |
| 507 "../api/audio_codecs:builtin_audio_decoder_factory", | 507 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 508 "../api/audio_codecs:builtin_audio_encoder_factory", | 508 "../api/audio_codecs:builtin_audio_encoder_factory", |
| 509 "../api/video_codecs:video_codecs_api", | 509 "../api/video_codecs:video_codecs_api", |
| 510 "../audio", | 510 "../audio", |
| 511 "../base:rtc_base", |
| 512 "../base:rtc_base_approved", |
| 513 "../base:rtc_base_tests_main", |
| 514 "../base:rtc_base_tests_utils", |
| 511 "../call:call_interfaces", | 515 "../call:call_interfaces", |
| 512 "../common_video:common_video", | 516 "../common_video:common_video", |
| 513 "../logging:rtc_event_log_api", | 517 "../logging:rtc_event_log_api", |
| 514 "../modules/audio_device:mock_audio_device", | 518 "../modules/audio_device:mock_audio_device", |
| 515 "../modules/audio_processing:audio_processing", | 519 "../modules/audio_processing:audio_processing", |
| 516 "../modules/video_coding:simulcast_test_utility", | 520 "../modules/video_coding:simulcast_test_utility", |
| 517 "../modules/video_coding:video_coding_utility", | 521 "../modules/video_coding:video_coding_utility", |
| 518 "../modules/video_coding:webrtc_vp8", | 522 "../modules/video_coding:webrtc_vp8", |
| 519 "../p2p:p2p_test_utils", | 523 "../p2p:p2p_test_utils", |
| 520 "../rtc_base:rtc_base", | |
| 521 "../rtc_base:rtc_base_approved", | |
| 522 "../rtc_base:rtc_base_tests_main", | |
| 523 "../rtc_base:rtc_base_tests_utils", | |
| 524 "../system_wrappers:metrics_default", | 524 "../system_wrappers:metrics_default", |
| 525 "../test:audio_codec_mocks", | 525 "../test:audio_codec_mocks", |
| 526 "../test:test_support", | 526 "../test:test_support", |
| 527 "../voice_engine:voice_engine", | 527 "../voice_engine:voice_engine", |
| 528 ] | 528 ] |
| 529 } | 529 } |
| 530 } | 530 } |
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