diff options
Diffstat (limited to '')
| -rw-r--r-- | third_party/libwebrtc/webrtc/media/BUILD.gn | 586 |
1 files changed, 586 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/media/BUILD.gn b/third_party/libwebrtc/webrtc/media/BUILD.gn new file mode 100644 index 0000000000..385163deaf --- /dev/null +++ b/third_party/libwebrtc/webrtc/media/BUILD.gn @@ -0,0 +1,586 @@ +# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("//build/config/linux/pkg_config.gni") +import("../webrtc.gni") + +group("media") { + public_deps = [ + ":rtc_media", + ":rtc_media_base", + ] +} + +config("rtc_media_defines_config") { + defines = [ + "HAVE_WEBRTC_VIDEO", + "HAVE_WEBRTC_VOICE", + ] +} + +config("rtc_media_warnings_config") { + # GN orders flags on a target before flags from configs. The default config + # adds these flags so to cancel them out they need to come from a config and + # cannot be on the target directly. + if (!is_win) { + cflags = [ "-Wno-deprecated-declarations" ] + } +} + +rtc_source_set("rtc_h264_profile_id") { + sources = [ + "base/h264_profile_level_id.cc", + "base/h264_profile_level_id.h", + ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } + + deps = [ + "..:webrtc_common", + "../api:optional", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + ] +} + +rtc_static_library("rtc_media_base") { + defines = [] + libs = [] + deps = [] + public_deps = [] + sources = [ + "base/adaptedvideotracksource.cc", + "base/adaptedvideotracksource.h", + "base/audiosource.h", + "base/codec.cc", + "base/codec.h", + "base/cryptoparams.h", + "base/device.h", + "base/mediachannel.h", + "base/mediaconstants.cc", + "base/mediaconstants.h", + "base/mediaengine.cc", + "base/mediaengine.h", + "base/streamparams.cc", + "base/streamparams.h", + "base/videoadapter.cc", + "base/videoadapter.h", + "base/videobroadcaster.cc", + "base/videobroadcaster.h", + "base/videocapturer.cc", + "base/videocapturer.h", + "base/videocapturerfactory.h", + "base/videocommon.cc", + "base/videocommon.h", + "base/videosourcebase.cc", + "base/videosourcebase.h", + + # TODO(aleloi): add "base/videosinkinterface.h" + "base/videosourceinterface.cc", + + # TODO(aleloi): add "base/videosourceinterface.h" + ] + + if (!build_with_mozilla) { + srcs += [ + "base/rtpdataengine.cc", + "base/rtpdataengine.h", + "base/rtputils.cc", + "base/rtputils.h", + "base/turnutils.cc", + "base/turnutils.h", + ] + } + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } + + include_dirs = [] + if (rtc_build_libyuv) { + deps += [ "$rtc_libyuv_dir" ] + public_deps += [ "$rtc_libyuv_dir" ] + } else { + # Need to add a directory normally exported by libyuv. + include_dirs += [ "$rtc_libyuv_dir/include" ] + } + + deps += [ + "..:webrtc_common", + "../api:optional", + "../api:video_frame_api", + "../api:video_frame_api_i420", + "../api/audio_codecs:audio_codecs_api", + "../api/video_codecs:video_codecs_api", + "../call:call_interfaces", + "../call:video_stream_api", + "../modules/audio_processing:audio_processing_statistics", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../system_wrappers:field_trial_api", + ] + + public_deps += [ ":rtc_h264_profile_id" ] + + if (!build_with_mozilla) { + deps += [ + "../api:libjingle_peerconnection_api", + "../p2p" + ] + } + + if (is_nacl) { + deps += [ "//native_client_sdk/src/libraries/nacl_io" ] + } +} + +rtc_static_library("rtc_audio_video") { + defines = [] + libs = [] + deps = [ + "../api:video_frame_api_i420", + "../modules/video_coding:video_coding_utility", + ] + sources = [ + "engine/adm_helpers.cc", + "engine/adm_helpers.h", + "engine/apm_helpers.cc", + "engine/apm_helpers.h", + "engine/constants.cc", + "engine/constants.h", + "engine/convert_legacy_video_factory.cc", + "engine/convert_legacy_video_factory.h", + "engine/encoder_simulcast_proxy.cc", + "engine/encoder_simulcast_proxy.h", + "engine/internaldecoderfactory.cc", + "engine/internaldecoderfactory.h", + "engine/internalencoderfactory.cc", + "engine/internalencoderfactory.h", + "engine/nullwebrtcvideoengine.h", + "engine/payload_type_mapper.cc", + "engine/payload_type_mapper.h", + "engine/scopedvideodecoder.cc", + "engine/scopedvideodecoder.h", + "engine/scopedvideoencoder.cc", + "engine/scopedvideoencoder.h", + "engine/simulcast.cc", + "engine/simulcast.h", + "engine/simulcast_encoder_adapter.cc", + "engine/simulcast_encoder_adapter.h", + "engine/videodecodersoftwarefallbackwrapper.cc", + "engine/videodecodersoftwarefallbackwrapper.h", + "engine/videoencodersoftwarefallbackwrapper.cc", + "engine/videoencodersoftwarefallbackwrapper.h", + "engine/webrtcvideodecoderfactory.cc", + "engine/webrtcvideodecoderfactory.h", + "engine/webrtcvideoencoderfactory.cc", + "engine/webrtcvideoencoderfactory.h", + ] + + if (!build_with_mozilla) { + sources += [ + "engine/webrtcvideocapturer.cc", + "engine/webrtcvideocapturer.h", + "engine/webrtcvideocapturerfactory.cc", + "engine/webrtcvideocapturerfactory.h", + "engine/webrtcmediaengine.cc", + "engine/webrtcmediaengine.h", + "engine/webrtcvideoengine.cc", + "engine/webrtcvideoengine.h", + "engine/webrtcvoe.h", + "engine/webrtcvoiceengine.cc", + "engine/webrtcvoiceengine.h", + ] + } + + configs += [ ":rtc_media_warnings_config" ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } + + if (is_win) { + cflags = [ + "/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch. + "/wd4267", # conversion from "size_t" to "int", possible loss of data. + "/wd4389", # signed/unsigned mismatch. + ] + } + + if (rtc_enable_intelligibility_enhancer) { + defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] + } else { + defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ] + } + + if (rtc_opus_support_120ms_ptime) { + defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ] + } else { + defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] + } + + include_dirs = [] + if (rtc_build_libyuv) { + deps += [ "$rtc_libyuv_dir" ] + public_deps = [ + "$rtc_libyuv_dir", + ] + } else { + # Need to add a directory normally exported by libyuv. + include_dirs += [ "$rtc_libyuv_dir/include" ] + } + + public_configs = [] + if (build_with_chromium) { + deps += [ "../modules/video_capture:video_capture" ] + } else { + public_configs += [ ":rtc_media_defines_config" ] + deps += [ "../modules/video_capture:video_capture_internal_impl" ] + } + if (rtc_enable_protobuf) { + deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ] + } else { + deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ] + } + deps += [ + ":rtc_media_base", + "..:webrtc_common", + "../api:call_api", + "../api:optional", + "../api:transport_api", + "../api:video_frame_api", + "../api/audio_codecs:audio_codecs_api", + "../api/video_codecs:video_codecs_api", + "../call", + "../call:video_stream_api", + "../common_video:common_video", + "../modules/audio_coding:rent_a_codec", + "../modules/audio_device:audio_device", + "../modules/audio_mixer:audio_mixer_impl", + "../modules/audio_processing:audio_processing", + "../modules/audio_processing/aec_dump", + "../modules/video_capture:video_capture_module", + "../modules/video_coding", + "../modules/video_coding:webrtc_h264", + "../modules/video_coding:webrtc_vp8", + "../modules/video_coding:webrtc_vp9", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_task_queue", + "../rtc_base:sequenced_task_checker", + "../system_wrappers", + "../video", + "../voice_engine", + ] + + if (!build_with_mozilla) { + deps += [ + "../api:libjingle_peerconnection_api", + "../p2p:rtc_p2p", + "../pc:rtc_pc_base", + ] + } +} + +rtc_static_library("rtc_data") { + defines = [] + deps = [] + + if (rtc_enable_sctp) { + sources = [ + "sctp/sctptransport.cc", + "sctp/sctptransport.h", + "sctp/sctptransportinternal.h", + ] + } + + configs += [ ":rtc_media_warnings_config" ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } + + if (is_win) { + cflags = [ + "/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch. + "/wd4267", # conversion from "size_t" to "int", possible loss of data. + "/wd4389", # signed/unsigned mismatch. + ] + } + + if (rtc_enable_sctp && rtc_build_usrsctp) { + include_dirs = [ + # TODO(jiayl): move this into the public_configs of + # //third_party/usrsctp/BUILD.gn. + "//third_party/usrsctp/usrsctplib", + ] + deps += [ "//third_party/usrsctp" ] + } + + deps += [ + ":rtc_media_base", + "..:webrtc_common", + "../api:call_api", + "../api:transport_api", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../system_wrappers", + ] + + if (!build_with_mozilla) { + deps += [ "../p2p:rtc_p2p" ] + } +} + +rtc_source_set("rtc_media") { + public_deps = [ + ":rtc_audio_video", + ] + + if (!build_with_mozilla) { + public_deps += [ + ":rtc_data", + ] + } +} + +if (rtc_include_tests) { + config("rtc_unittest_main_config") { + # GN orders flags on a target before flags from configs. The default config + # adds -Wall, and this flag have to be after -Wall -- so they need to + # come from a config and can"t be on the target directly. + if (is_clang && is_ios) { + cflags = [ "-Wno-unused-variable" ] + } + } + + rtc_source_set("rtc_media_tests_utils") { + testonly = true + + include_dirs = [] + public_deps = [] + deps = [ + "../api:video_frame_api_i420", + "../call:video_stream_api", + "../modules/audio_coding:rent_a_codec", + "../modules/audio_processing:audio_processing", + "../modules/rtp_rtcp:rtp_rtcp", + "../modules/video_coding:video_coding_utility", + "../p2p:rtc_p2p", + ] + sources = [ + "base/fakemediaengine.h", + "base/fakenetworkinterface.h", + "base/fakertp.cc", + "base/fakertp.h", + "base/fakevideocapturer.h", + "base/fakevideorenderer.h", + "base/test/mock_mediachannel.h", + "base/testutils.cc", + "base/testutils.h", + "engine/fakewebrtccall.cc", + "engine/fakewebrtccall.h", + "engine/fakewebrtcdeviceinfo.h", + "engine/fakewebrtcvcmfactory.h", + "engine/fakewebrtcvideocapturemodule.h", + "engine/fakewebrtcvideoengine.h", + "engine/fakewebrtcvoiceengine.h", + ] + + configs += [ ":rtc_unittest_main_config" ] + + if (rtc_build_libyuv) { + deps += [ "$rtc_libyuv_dir" ] + public_deps += [ "$rtc_libyuv_dir" ] + } else { + # Need to add a directory normally exported by libyuv. + include_dirs += [ "$rtc_libyuv_dir/include" ] + } + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } + + deps += [ + ":rtc_media", + ":rtc_media_base", + "..:webrtc_common", + "../api:call_api", + "../api:video_frame_api", + "../api/video_codecs:video_codecs_api", + "../call:call_interfaces", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_utils", + "../test:test_support", + "//testing/gtest", + ] + public_deps += [ "//testing/gmock" ] + } + + config("rtc_media_unittests_config") { + # GN orders flags on a target before flags from configs. The default config + # adds -Wall, and this flag have to be after -Wall -- so they need to + # come from a config and can"t be on the target directly. + # TODO(kjellander): Make the code compile without disabling these flags. + # See https://bugs.webrtc.org/3307. + if (is_clang && is_win) { + cflags = [ + # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6266 + # for -Wno-sign-compare + "-Wno-sign-compare", + ] + } + if (!is_win) { + cflags = [ "-Wno-sign-compare" ] + } + } + + rtc_media_unittests_resources = [ + "../resources/media/captured-320x240-2s-48.frames", + "../resources/media/faces.1280x720_P420.yuv", + "../resources/media/faces_I420.jpg", + "../resources/media/faces_I422.jpg", + "../resources/media/faces_I444.jpg", + "../resources/media/faces_I411.jpg", + "../resources/media/faces_I400.jpg", + ] + + if (is_ios) { + bundle_data("rtc_media_unittests_bundle_data") { + testonly = true + sources = rtc_media_unittests_resources + outputs = [ + "{{bundle_resources_dir}}/{{source_file_part}}", + ] + } + } + + rtc_test("rtc_media_unittests") { + testonly = true + + defines = [] + deps = [ + "../api:video_frame_api_i420", + "../pc:rtc_pc", + "../test:field_trial", + ] + sources = [ + "base/codec_unittest.cc", + "base/rtpdataengine_unittest.cc", + "base/rtputils_unittest.cc", + "base/streamparams_unittest.cc", + "base/turnutils_unittest.cc", + "base/videoadapter_unittest.cc", + "base/videobroadcaster_unittest.cc", + "base/videocapturer_unittest.cc", + "base/videocommon_unittest.cc", + "base/videoengine_unittest.h", + "engine/apm_helpers_unittest.cc", + "engine/encoder_simulcast_proxy_unittest.cc", + "engine/internaldecoderfactory_unittest.cc", + "engine/nullwebrtcvideoengine_unittest.cc", + "engine/payload_type_mapper_unittest.cc", + "engine/simulcast_encoder_adapter_unittest.cc", + "engine/simulcast_unittest.cc", + "engine/videodecodersoftwarefallbackwrapper_unittest.cc", + "engine/videoencodersoftwarefallbackwrapper_unittest.cc", + "engine/webrtcmediaengine_unittest.cc", + "engine/webrtcvideocapturer_unittest.cc", + "engine/webrtcvideoencoderfactory_unittest.cc", + "engine/webrtcvideoengine_unittest.cc", + ] + + # TODO(kthelgason): Reenable this test on iOS. + # See bugs.webrtc.org/5569 + if (!is_ios) { + sources += [ "engine/webrtcvoiceengine_unittest.cc" ] + } + + if (rtc_enable_sctp) { + sources += [ "sctp/sctptransport_unittest.cc" ] + } + + configs += [ ":rtc_media_unittests_config" ] + + if (rtc_use_h264) { + defines += [ "WEBRTC_USE_H264" ] + } + + if (rtc_opus_support_120ms_ptime) { + defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ] + } else { + defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] + } + + if (is_win) { + cflags = [ + "/wd4245", # conversion from int to size_t, signed/unsigned mismatch. + "/wd4373", # virtual function override. + "/wd4389", # signed/unsigned mismatch. + ] + } + + if (!build_with_chromium && is_clang) { + suppressed_configs += [ + "//build/config/clang:extra_warnings", + + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + "//build/config/clang:find_bad_constructs", + ] + } + + data = rtc_media_unittests_resources + + if (is_android) { + deps += [ "//testing/android/native_test:native_test_support" ] + shard_timeout = 900 + } + + if (is_ios) { + deps += [ ":rtc_media_unittests_bundle_data" ] + } + + deps += [ + ":rtc_media", + ":rtc_media_base", + ":rtc_media_tests_utils", + "../api:mock_video_codec_factory", + "../api:video_frame_api", + "../api/audio_codecs:builtin_audio_decoder_factory", + "../api/audio_codecs:builtin_audio_encoder_factory", + "../api/video_codecs:video_codecs_api", + "../audio", + "../call:call_interfaces", + "../common_video:common_video", + "../logging:rtc_event_log_api", + "../modules/audio_device:mock_audio_device", + "../modules/audio_processing:audio_processing", + "../modules/video_coding:simulcast_test_utility", + "../modules/video_coding:video_coding_utility", + "../modules/video_coding:webrtc_vp8", + "../p2p:p2p_test_utils", + "../rtc_base:rtc_base", + "../rtc_base:rtc_base_approved", + "../rtc_base:rtc_base_tests_main", + "../rtc_base:rtc_base_tests_utils", + "../system_wrappers:metrics_default", + "../test:audio_codec_mocks", + "../test:test_support", + "../test:video_test_common", + "../voice_engine:voice_engine", + ] + } +} |