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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") |
| (...skipping 15 matching lines...) Expand all Loading... |
| 26 config("rtc_media_warnings_config") { | 26 config("rtc_media_warnings_config") { |
| 27 # GN orders flags on a target before flags from configs. The default config | 27 # GN orders flags on a target before flags from configs. The default config |
| 28 # adds these flags so to cancel them out they need to come from a config and | 28 # adds these flags so to cancel them out they need to come from a config and |
| 29 # cannot be on the target directly. | 29 # cannot be on the target directly. |
| 30 if (!is_win) { | 30 if (!is_win) { |
| 31 cflags = [ "-Wno-deprecated-declarations" ] | 31 cflags = [ "-Wno-deprecated-declarations" ] |
| 32 } | 32 } |
| 33 } | 33 } |
| 34 | 34 |
| 35 rtc_static_library("rtc_media_base") { | 35 rtc_static_library("rtc_media_base") { |
| 36 # TODO(kjellander): Remove (bugs.webrtc.org/6828) |
| 37 # Enabling GN check triggers cyclic dependency error: |
| 38 # //webrtc/media:rtc_media_base -> |
| 39 # //webrtc/pc:rtc_pc -> |
| 40 # //webrtc/media:media -> |
| 41 # //webrtc/media:rtc_media_base |
| 42 check_includes = false |
| 36 defines = [] | 43 defines = [] |
| 37 libs = [] | 44 libs = [] |
| 38 deps = [] | 45 deps = [] |
| 39 sources = [ | 46 sources = [ |
| 40 "base/adaptedvideotracksource.cc", | 47 "base/adaptedvideotracksource.cc", |
| 41 "base/adaptedvideotracksource.h", | 48 "base/adaptedvideotracksource.h", |
| 42 "base/audiosource.h", | 49 "base/audiosource.h", |
| 43 "base/codec.cc", | 50 "base/codec.cc", |
| 44 "base/codec.h", | 51 "base/codec.h", |
| 45 "base/cryptoparams.h", | 52 "base/cryptoparams.h", |
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| 84 public_deps = [ | 91 public_deps = [ |
| 85 "$rtc_libyuv_dir", | 92 "$rtc_libyuv_dir", |
| 86 ] | 93 ] |
| 87 } else { | 94 } else { |
| 88 # Need to add a directory normally exported by libyuv. | 95 # Need to add a directory normally exported by libyuv. |
| 89 include_dirs += [ "$rtc_libyuv_dir/include" ] | 96 include_dirs += [ "$rtc_libyuv_dir/include" ] |
| 90 } | 97 } |
| 91 | 98 |
| 92 deps += [ | 99 deps += [ |
| 93 "..:webrtc_common", | 100 "..:webrtc_common", |
| 101 "../api:libjingle_peerconnection_api", |
| 102 "../api:video_frame_api", |
| 103 "../api/audio_codecs:audio_codecs_api", |
| 104 "../base:rtc_base", |
| 94 "../base:rtc_base_approved", | 105 "../base:rtc_base_approved", |
| 106 "../call:call_interfaces", |
| 107 "../common_video:common_video", |
| 95 "../p2p", | 108 "../p2p", |
| 96 ] | 109 ] |
| 97 } | 110 } |
| 98 | 111 |
| 99 rtc_static_library("rtc_media") { | 112 rtc_static_library("rtc_media") { |
| 113 # TODO(kjellander): Remove (bugs.webrtc.org/6828) |
| 114 # Enabling GN check triggers cyclic dependency error: |
| 115 # //webrtc/media:media -> |
| 116 # //webrtc/media:rtc_media -> |
| 117 # //webrtc/pc:rtc_pc -> |
| 118 # //webrtc/media:media |
| 119 check_includes = false |
| 100 defines = [] | 120 defines = [] |
| 101 libs = [] | 121 libs = [] |
| 102 deps = [] | 122 deps = [] |
| 103 sources = [ | 123 sources = [ |
| 104 "engine/apm_helpers.cc", | 124 "engine/apm_helpers.cc", |
| 105 "engine/apm_helpers.h", | 125 "engine/apm_helpers.h", |
| 106 "engine/internaldecoderfactory.cc", | 126 "engine/internaldecoderfactory.cc", |
| 107 "engine/internaldecoderfactory.h", | 127 "engine/internaldecoderfactory.h", |
| 108 "engine/internalencoderfactory.cc", | 128 "engine/internalencoderfactory.cc", |
| 109 "engine/internalencoderfactory.h", | 129 "engine/internalencoderfactory.h", |
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| 194 deps += [ "../modules/video_capture:video_capture" ] | 214 deps += [ "../modules/video_capture:video_capture" ] |
| 195 } else { | 215 } else { |
| 196 public_configs += [ ":rtc_media_defines_config" ] | 216 public_configs += [ ":rtc_media_defines_config" ] |
| 197 deps += [ "../modules/video_capture:video_capture_internal_impl" ] | 217 deps += [ "../modules/video_capture:video_capture_internal_impl" ] |
| 198 } | 218 } |
| 199 deps += [ | 219 deps += [ |
| 200 ":rtc_media_base", | 220 ":rtc_media_base", |
| 201 "..:webrtc_common", | 221 "..:webrtc_common", |
| 202 "../api:call_api", | 222 "../api:call_api", |
| 203 "../api:transport_api", | 223 "../api:transport_api", |
| 224 "../api:video_frame_api", |
| 225 "../api/audio_codecs:audio_codecs_api", |
| 226 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 227 "../base:rtc_base", |
| 204 "../base:rtc_base_approved", | 228 "../base:rtc_base_approved", |
| 205 "../call", | 229 "../call", |
| 230 "../common_video:common_video", |
| 231 "../modules/audio_coding:rent_a_codec", |
| 232 "../modules/audio_device:audio_device", |
| 206 "../modules/audio_mixer:audio_mixer_impl", | 233 "../modules/audio_mixer:audio_mixer_impl", |
| 234 "../modules/audio_processing:audio_processing", |
| 235 "../modules/video_capture:video_capture_module", |
| 207 "../modules/video_coding", | 236 "../modules/video_coding", |
| 237 "../modules/video_coding:webrtc_h264", |
| 238 "../modules/video_coding:webrtc_vp8", |
| 239 "../modules/video_coding:webrtc_vp9", |
| 240 "../p2p:rtc_p2p", |
| 208 "../system_wrappers", | 241 "../system_wrappers", |
| 242 "../video", |
| 209 "../voice_engine", | 243 "../voice_engine", |
| 210 ] | 244 ] |
| 211 } | 245 } |
| 212 | 246 |
| 213 if (rtc_include_tests) { | 247 if (rtc_include_tests) { |
| 214 config("rtc_unittest_main_config") { | 248 config("rtc_unittest_main_config") { |
| 215 # GN orders flags on a target before flags from configs. The default config | 249 # GN orders flags on a target before flags from configs. The default config |
| 216 # adds -Wall, and this flag have to be after -Wall -- so they need to | 250 # adds -Wall, and this flag have to be after -Wall -- so they need to |
| 217 # come from a config and can"t be on the target directly. | 251 # come from a config and can"t be on the target directly. |
| 218 if (is_clang && is_ios) { | 252 if (is_clang && is_ios) { |
| 219 cflags = [ "-Wno-unused-variable" ] | 253 cflags = [ "-Wno-unused-variable" ] |
| 220 } | 254 } |
| 221 } | 255 } |
| 222 | 256 |
| 223 rtc_source_set("rtc_unittest_main") { | 257 rtc_source_set("rtc_unittest_main") { |
| 224 testonly = true | 258 testonly = true |
| 225 | 259 |
| 226 include_dirs = [] | 260 include_dirs = [] |
| 227 public_deps = [] | 261 public_deps = [] |
| 228 deps = [] | 262 deps = [ |
| 263 "../modules/audio_coding:rent_a_codec", |
| 264 "../modules/audio_processing:audio_processing", |
| 265 "../p2p:rtc_p2p", |
| 266 ] |
| 229 sources = [ | 267 sources = [ |
| 230 "base/fakemediaengine.h", | 268 "base/fakemediaengine.h", |
| 231 "base/fakenetworkinterface.h", | 269 "base/fakenetworkinterface.h", |
| 232 "base/fakertp.h", | 270 "base/fakertp.h", |
| 233 "base/fakevideocapturer.h", | 271 "base/fakevideocapturer.h", |
| 234 "base/fakevideorenderer.h", | 272 "base/fakevideorenderer.h", |
| 235 "base/test/mock_mediachannel.h", | 273 "base/test/mock_mediachannel.h", |
| 236 "base/testutils.cc", | 274 "base/testutils.cc", |
| 237 "base/testutils.h", | 275 "base/testutils.h", |
| 238 "engine/fakewebrtccall.cc", | 276 "engine/fakewebrtccall.cc", |
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| 253 # Need to add a directory normally exported by libyuv. | 291 # Need to add a directory normally exported by libyuv. |
| 254 include_dirs += [ "$rtc_libyuv_dir/include" ] | 292 include_dirs += [ "$rtc_libyuv_dir/include" ] |
| 255 } | 293 } |
| 256 | 294 |
| 257 if (!build_with_chromium && is_clang) { | 295 if (!build_with_chromium && is_clang) { |
| 258 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 296 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 259 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 297 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 260 } | 298 } |
| 261 | 299 |
| 262 deps += [ | 300 deps += [ |
| 301 ":rtc_media", |
| 302 ":rtc_media_base", |
| 303 "..:webrtc_common", |
| 304 "../api:call_api", |
| 305 "../api:video_frame_api", |
| 306 "../base:rtc_base", |
| 307 "../base:rtc_base_approved", |
| 263 "../base:rtc_base_tests_main", | 308 "../base:rtc_base_tests_main", |
| 309 "../base:rtc_base_tests_utils", |
| 310 "../call:call_interfaces", |
| 311 "../test:test_support", |
| 264 "//testing/gtest", | 312 "//testing/gtest", |
| 265 ] | 313 ] |
| 266 public_deps += [ "//testing/gmock" ] | 314 public_deps += [ "//testing/gmock" ] |
| 267 } | 315 } |
| 268 | 316 |
| 269 config("rtc_media_unittests_config") { | 317 config("rtc_media_unittests_config") { |
| 270 # GN orders flags on a target before flags from configs. The default config | 318 # GN orders flags on a target before flags from configs. The default config |
| 271 # adds -Wall, and this flag have to be after -Wall -- so they need to | 319 # adds -Wall, and this flag have to be after -Wall -- so they need to |
| 272 # come from a config and can"t be on the target directly. | 320 # come from a config and can"t be on the target directly. |
| 273 # TODO(kjellander): Make the code compile without disabling these flags. | 321 # TODO(kjellander): Make the code compile without disabling these flags. |
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| 302 outputs = [ | 350 outputs = [ |
| 303 "{{bundle_resources_dir}}/{{source_file_part}}", | 351 "{{bundle_resources_dir}}/{{source_file_part}}", |
| 304 ] | 352 ] |
| 305 } | 353 } |
| 306 } | 354 } |
| 307 | 355 |
| 308 rtc_test("rtc_media_unittests") { | 356 rtc_test("rtc_media_unittests") { |
| 309 testonly = true | 357 testonly = true |
| 310 | 358 |
| 311 defines = [] | 359 defines = [] |
| 312 deps = [] | 360 deps = [ |
| 361 "../pc:rtc_pc", |
| 362 "../test:field_trial", |
| 363 ] |
| 313 sources = [ | 364 sources = [ |
| 314 "base/codec_unittest.cc", | 365 "base/codec_unittest.cc", |
| 315 "base/rtpdataengine_unittest.cc", | 366 "base/rtpdataengine_unittest.cc", |
| 316 "base/rtputils_unittest.cc", | 367 "base/rtputils_unittest.cc", |
| 317 "base/streamparams_unittest.cc", | 368 "base/streamparams_unittest.cc", |
| 318 "base/turnutils_unittest.cc", | 369 "base/turnutils_unittest.cc", |
| 319 "base/videoadapter_unittest.cc", | 370 "base/videoadapter_unittest.cc", |
| 320 "base/videobroadcaster_unittest.cc", | 371 "base/videobroadcaster_unittest.cc", |
| 321 "base/videocapturer_unittest.cc", | 372 "base/videocapturer_unittest.cc", |
| 322 "base/videocommon_unittest.cc", | 373 "base/videocommon_unittest.cc", |
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| 373 if (is_android) { | 424 if (is_android) { |
| 374 deps += [ "//testing/android/native_test:native_test_support" ] | 425 deps += [ "//testing/android/native_test:native_test_support" ] |
| 375 shard_timeout = 900 | 426 shard_timeout = 900 |
| 376 } | 427 } |
| 377 | 428 |
| 378 if (is_ios) { | 429 if (is_ios) { |
| 379 deps += [ ":rtc_media_unittests_bundle_data" ] | 430 deps += [ ":rtc_media_unittests_bundle_data" ] |
| 380 } | 431 } |
| 381 | 432 |
| 382 deps += [ | 433 deps += [ |
| 383 # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243. | |
| 384 ":rtc_media", | 434 ":rtc_media", |
| 435 ":rtc_media_base", |
| 385 ":rtc_unittest_main", | 436 ":rtc_unittest_main", |
| 437 "../api:video_frame_api", |
| 438 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 386 "../audio", | 439 "../audio", |
| 440 "../base:rtc_base", |
| 441 "../base:rtc_base_approved", |
| 387 "../base:rtc_base_tests_utils", | 442 "../base:rtc_base_tests_utils", |
| 443 "../call:call_interfaces", |
| 444 "../common_video:common_video", |
| 445 "../logging:rtc_event_log_api", |
| 388 "../modules/audio_device:mock_audio_device", | 446 "../modules/audio_device:mock_audio_device", |
| 447 "../modules/audio_processing:audio_processing", |
| 448 "../modules/video_coding:video_coding_utility", |
| 449 "../modules/video_coding:webrtc_vp8", |
| 450 "../p2p:rtc_p2p_unittests", |
| 389 "../system_wrappers:metrics_default", | 451 "../system_wrappers:metrics_default", |
| 452 "../test:test_support", |
| 453 "../voice_engine:voice_engine", |
| 390 ] | 454 ] |
| 391 } | 455 } |
| 392 } | 456 } |
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