| # 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") { |
| deps = [] |
| if (!build_with_mozilla) { |
| deps += [ |
| ":rtc_media", |
| ":rtc_media_base", |
| ] |
| } |
| } |
| |
| config("rtc_media_defines_config") { |
| defines = [ |
| "HAVE_WEBRTC_VIDEO", |
| "HAVE_WEBRTC_VOICE", |
| ] |
| } |
| |
| rtc_source_set("rtc_h264_profile_id") { |
| visibility = [ "*" ] |
| 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", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base/system:rtc_export", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| rtc_source_set("rtc_media_config") { |
| visibility = [ "*" ] |
| sources = [ |
| "base/media_config.h", |
| "base/mediaconfig.h", |
| ] |
| } |
| |
| rtc_source_set("rtc_vp9_profile") { |
| visibility = [ "*" ] |
| sources = [ |
| "base/vp9_profile.cc", |
| "base/vp9_profile.h", |
| ] |
| |
| deps = [ |
| "..:webrtc_common", |
| "../api/video_codecs:video_codecs_api", |
| "../rtc_base:rtc_base_approved", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| rtc_static_library("rtc_media_base") { |
| visibility = [ "*" ] |
| defines = [] |
| libs = [] |
| deps = [ |
| "../api:array_view", |
| "../api:audio_options_api", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:sanitizer", |
| "../rtc_base:sequenced_task_checker", |
| "../rtc_base:stringutils", |
| ] |
| sources = [ |
| "base/adapted_video_track_source.cc", |
| "base/adapted_video_track_source.h", |
| "base/adaptedvideotracksource.h", |
| "base/audio_source.h", |
| "base/audiosource.h", |
| "base/codec.cc", |
| "base/codec.h", |
| "base/device.h", |
| "base/media_channel.cc", |
| "base/media_channel.h", |
| "base/media_constants.cc", |
| "base/media_constants.h", |
| "base/media_engine.cc", |
| "base/media_engine.h", |
| "base/mediachannel.h", |
| "base/mediaconstants.h", |
| "base/mediaengine.h", |
| "base/rid_description.cc", |
| "base/rid_description.h", |
| "base/riddescription.h", |
| "base/rtp_data_engine.cc", |
| "base/rtp_data_engine.h", |
| "base/rtp_utils.cc", |
| "base/rtp_utils.h", |
| "base/rtpdataengine.h", |
| "base/rtputils.h", |
| "base/stream_params.cc", |
| "base/stream_params.h", |
| "base/streamparams.h", |
| "base/turn_utils.cc", |
| "base/turn_utils.h", |
| "base/turnutils.h", |
| "base/video_adapter.cc", |
| "base/video_adapter.h", |
| "base/video_broadcaster.cc", |
| "base/video_broadcaster.h", |
| "base/video_capturer.cc", |
| "base/video_capturer.h", |
| "base/video_common.cc", |
| "base/video_common.h", |
| "base/video_source_base.cc", |
| "base/video_source_base.h", |
| "base/videoadapter.h", |
| "base/videobroadcaster.h", |
| "base/videocapturer.h", |
| "base/videocommon.h", |
| "base/videosourcebase.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 += [ |
| ":rtc_h264_profile_id", |
| ":rtc_media_config", |
| ":rtc_vp9_profile", |
| "..:webrtc_common", |
| "../api:libjingle_peerconnection_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video:video_frame_i420", |
| "../api/video_codecs:video_codecs_api", |
| "../call:call_interfaces", |
| "../common_video", |
| "../modules/audio_processing:audio_processing_statistics", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base/system:rtc_export", |
| "../rtc_base/third_party/sigslot", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| if (!build_with_mozilla) { |
| deps += [ "../p2p" ] |
| } |
| |
| if (is_nacl) { |
| deps += [ "//native_client_sdk/src/libraries/nacl_io" ] |
| } |
| } |
| |
| rtc_static_library("rtc_constants") { |
| defines = [] |
| libs = [] |
| deps = [] |
| sources = [ |
| "engine/constants.cc", |
| "engine/constants.h", |
| ] |
| } |
| |
| rtc_static_library("rtc_simulcast_encoder_adapter") { |
| visibility = [ "*" ] |
| defines = [] |
| libs = [] |
| sources = [ |
| "engine/simulcast_encoder_adapter.cc", |
| "engine/simulcast_encoder_adapter.h", |
| ] |
| deps = [ |
| "../api/video:video_codec_constants", |
| "../api/video:video_frame", |
| "../api/video:video_frame_i420", |
| "../api/video_codecs:video_codecs_api", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:video_coding_utility", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:sequenced_task_checker", |
| "../system_wrappers", |
| "../system_wrappers:field_trial", |
| "//third_party/abseil-cpp/absl/types:optional", |
| "//third_party/libyuv", |
| ] |
| 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" ] |
| } |
| } |
| |
| rtc_static_library("rtc_encoder_simulcast_proxy") { |
| visibility = [ "*" ] |
| defines = [] |
| libs = [] |
| sources = [ |
| "engine/encoder_simulcast_proxy.cc", |
| "engine/encoder_simulcast_proxy.h", |
| ] |
| deps = [ |
| ":rtc_simulcast_encoder_adapter", |
| "../:webrtc_common", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video_codecs:video_codecs_api", |
| "../modules/video_coding:video_codec_interface", |
| ] |
| 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" ] |
| } |
| } |
| |
| rtc_static_library("rtc_internal_video_codecs") { |
| visibility = [ "*" ] |
| allow_poison = [ "software_video_codecs" ] |
| defines = [] |
| libs = [] |
| deps = [ |
| ":rtc_encoder_simulcast_proxy", |
| ":rtc_h264_profile_id", |
| ":rtc_simulcast_encoder_adapter", |
| "../:webrtc_common", |
| "../api/video:encoded_image", |
| "../api/video:video_frame", |
| "../modules/video_coding:video_codec_interface", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| sources = [ |
| "engine/fake_video_codec_factory.cc", |
| "engine/fake_video_codec_factory.h", |
| "engine/internal_decoder_factory.cc", |
| "engine/internal_decoder_factory.h", |
| "engine/internal_encoder_factory.cc", |
| "engine/internal_encoder_factory.h", |
| "engine/internaldecoderfactory.h", |
| "engine/internalencoderfactory.h", |
| "engine/multiplex_codec_factory.cc", |
| "engine/multiplex_codec_factory.h", |
| "engine/multiplexcodecfactory.h", |
| |
| # TODO(bugs.webrtc.org/7925): stop exporting this header once downstream |
| # targets depend on :rtc_encoder_simulcast_proxy directly. |
| "engine/encoder_simulcast_proxy.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 = [] |
| |
| public_configs = [] |
| deps += [ |
| ":rtc_constants", |
| ":rtc_media_base", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video_codecs:rtc_software_fallback_wrappers", |
| "../api/video_codecs:video_codecs_api", |
| "../call:call_interfaces", |
| "../call:video_stream_api", |
| "../modules:module_api", |
| "../modules/video_coding:webrtc_h264", |
| "../modules/video_coding:webrtc_multiplex", |
| "../modules/video_coding:webrtc_vp8", |
| "../modules/video_coding:webrtc_vp9", |
| "../rtc_base:checks", |
| "../rtc_base:deprecation", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base/system:rtc_export", |
| "../test:fake_video_codecs", |
| "//third_party/abseil-cpp/absl/strings", |
| ] |
| } |
| |
| rtc_static_library("rtc_audio_video") { |
| visibility = [ "*" ] |
| allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove. |
| defines = [] |
| libs = [] |
| deps = [ |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_bitrate_allocator_factory", |
| "../modules/audio_processing:api", |
| "../modules/audio_processing:gain_control_interface", |
| "../modules/audio_processing/aec_dump:aec_dump", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:video_coding", |
| "../modules/video_coding:video_coding_utility", |
| "../rtc_base:audio_format_to_string", |
| "../rtc_base:checks", |
| "../rtc_base/system:rtc_export", |
| "../rtc_base/third_party/base64", |
| "../system_wrappers:field_trial", |
| "../system_wrappers:metrics", |
| ] |
| |
| sources = [ |
| "engine/adm_helpers.cc", |
| "engine/adm_helpers.h", |
| "engine/apm_helpers.cc", |
| "engine/apm_helpers.h", |
| "engine/null_webrtc_video_engine.h", |
| "engine/nullwebrtcvideoengine.h", |
| "engine/payload_type_mapper.cc", |
| "engine/payload_type_mapper.h", |
| "engine/simulcast.cc", |
| "engine/simulcast.h", |
| "engine/webrtc_media_engine.cc", |
| "engine/webrtc_media_engine.h", |
| "engine/webrtc_video_engine.cc", |
| "engine/webrtc_video_engine.h", |
| "engine/webrtc_voice_engine.cc", |
| "engine/webrtc_voice_engine.h", |
| "engine/webrtcmediaengine.h", |
| "engine/webrtcvideoengine.h", |
| "engine/webrtcvoiceengine.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" ] |
| } |
| |
| if (rtc_opus_support_120ms_ptime) { |
| defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ] |
| } else { |
| defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] |
| } |
| |
| include_dirs = [] |
| |
| 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_constants", |
| ":rtc_media_base", |
| "..:webrtc_common", |
| "../api:call_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:transport_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/video:builtin_video_bitrate_allocator_factory", |
| "../api/video:video_codec_constants", |
| "../api/video:video_frame", |
| "../api/video:video_frame_i420", |
| "../api/video_codecs:rtc_software_fallback_wrappers", |
| "../api/video_codecs:video_codecs_api", |
| "../call", |
| "../call:call_interfaces", |
| "../call:video_stream_api", |
| "../common_video:common_video", |
| "../modules/audio_device:audio_device", |
| "../modules/audio_device:audio_device_impl", |
| "../modules/audio_mixer:audio_mixer_impl", |
| "../modules/audio_processing:audio_processing", |
| "../pc:rtc_pc_base", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:stringutils", |
| "../rtc_base/experiments:normalize_simulcast_size_experiment", |
| "../system_wrappers", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| rtc_static_library("rtc_data") { |
| defines = [] |
| deps = [] |
| |
| if (rtc_enable_sctp) { |
| sources = [ |
| "sctp/sctp_transport.cc", |
| "sctp/sctp_transport.h", |
| "sctp/sctp_transport_internal.h", |
| "sctp/sctptransport.h", |
| "sctp/sctptransportinternal.h", |
| ] |
| } else { |
| # libtool on mac does not like empty targets. |
| sources = [ |
| "sctp/noop.cc", |
| ] |
| } |
| |
| 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 (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", |
| "../p2p:rtc_p2p", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base/third_party/sigslot", |
| "../system_wrappers", |
| ] |
| } |
| |
| rtc_source_set("rtc_media") { |
| visibility = [ "*" ] |
| allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove. |
| deps = [ |
| ":rtc_audio_video", |
| ":rtc_data", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_source_set("rtc_media_tests_utils") { |
| testonly = true |
| |
| defines = [] |
| include_dirs = [] |
| deps = [ |
| ":rtc_audio_video", |
| ":rtc_simulcast_encoder_adapter", |
| "../api:libjingle_peerconnection_api", |
| "../api/video:encoded_image", |
| "../api/video:video_frame_i420", |
| "../call:video_stream_api", |
| "../common_video:common_video", |
| "../modules/audio_processing:api", |
| "../modules/audio_processing:audio_processing", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:video_coding_utility", |
| "../p2p:rtc_p2p", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:stringutils", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| ] |
| sources = [ |
| "base/fake_frame_source.cc", |
| "base/fake_frame_source.h", |
| "base/fake_media_engine.cc", |
| "base/fake_media_engine.h", |
| "base/fake_network_interface.h", |
| "base/fake_rtp.cc", |
| "base/fake_rtp.h", |
| "base/fake_video_capturer.cc", |
| "base/fake_video_capturer.h", |
| "base/fake_video_renderer.cc", |
| "base/fake_video_renderer.h", |
| "base/fakeframesource.h", |
| "base/fakemediaengine.h", |
| "base/fakenetworkinterface.h", |
| "base/fakertp.h", |
| "base/fakevideocapturer.h", |
| "base/fakevideorenderer.h", |
| "base/test_utils.cc", |
| "base/test_utils.h", |
| "base/testutils.h", |
| "engine/fake_webrtc_call.cc", |
| "engine/fake_webrtc_call.h", |
| "engine/fake_webrtc_video_engine.cc", |
| "engine/fake_webrtc_video_engine.h", |
| "engine/fakewebrtccall.h", |
| "engine/fakewebrtcvideoengine.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" ] |
| } |
| |
| if (rtc_use_h264) { |
| defines += [ "WEBRTC_USE_H264" ] |
| } |
| |
| deps += [ |
| ":rtc_internal_video_codecs", |
| ":rtc_media", |
| ":rtc_media_base", |
| "..:webrtc_common", |
| "../api:call_api", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video_codecs:video_codecs_api", |
| "../call:call_interfaces", |
| "../call:mock_rtp_interfaces", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue_for_test", |
| "../rtc_base/third_party/sigslot", |
| "../test:test_support", |
| "//testing/gtest", |
| ] |
| } |
| |
| 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 = [ |
| ":rtc_audio_video", |
| ":rtc_constants", |
| ":rtc_data", |
| "../:webrtc_common", |
| "../api/test/video:function_video_factory", |
| "../api/units:time_delta", |
| "../api/video:video_frame_i420", |
| "../modules/audio_processing:api", |
| "../modules/audio_processing:mocks", |
| "../modules/rtp_rtcp", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:webrtc_vp8", |
| "../pc:rtc_pc", |
| "../pc:rtc_pc_base", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base_tests_utils", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:stringutils", |
| "../rtc_base/third_party/sigslot:sigslot", |
| "../test:field_trial", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| ] |
| sources = [ |
| "base/codec_unittest.cc", |
| "base/rtp_data_engine_unittest.cc", |
| "base/rtp_utils_unittest.cc", |
| "base/stream_params_unittest.cc", |
| "base/turn_utils_unittest.cc", |
| "base/video_adapter_unittest.cc", |
| "base/video_broadcaster_unittest.cc", |
| "base/video_capturer_unittest.cc", |
| "base/video_common_unittest.cc", |
| "engine/apm_helpers_unittest.cc", |
| "engine/encoder_simulcast_proxy_unittest.cc", |
| "engine/internal_decoder_factory_unittest.cc", |
| "engine/multiplex_codec_factory_unittest.cc", |
| "engine/null_webrtc_video_engine_unittest.cc", |
| "engine/payload_type_mapper_unittest.cc", |
| "engine/simulcast_encoder_adapter_unittest.cc", |
| "engine/simulcast_unittest.cc", |
| "engine/webrtc_media_engine_unittest.cc", |
| "engine/webrtc_video_engine_unittest.cc", |
| ] |
| |
| # TODO(kthelgason): Reenable this test on iOS. |
| # See bugs.webrtc.org/5569 |
| if (!is_ios) { |
| sources += [ "engine/webrtc_voice_engine_unittest.cc" ] |
| } |
| |
| if (rtc_enable_sctp) { |
| sources += [ "sctp/sctp_transport_unittest.cc" ] |
| } |
| |
| 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 (!build_with_chromium && is_clang) { |
| suppressed_configs += [ |
| # 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_encoder_simulcast_proxy", |
| ":rtc_internal_video_codecs", |
| ":rtc_media", |
| ":rtc_media_base", |
| ":rtc_media_tests_utils", |
| ":rtc_simulcast_encoder_adapter", |
| ":rtc_vp9_profile", |
| "../api:create_simulcast_test_fixture_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:mock_video_bitrate_allocator", |
| "../api:mock_video_bitrate_allocator_factory", |
| "../api:mock_video_codec_factory", |
| "../api:simulcast_test_fixture_api", |
| "../api/audio_codecs:builtin_audio_decoder_factory", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../api/video:builtin_video_bitrate_allocator_factory", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video_codecs:builtin_video_decoder_factory", |
| "../api/video_codecs:builtin_video_encoder_factory", |
| "../api/video_codecs:video_codecs_api", |
| "../audio", |
| "../call:call_interfaces", |
| "../common_video:common_video", |
| "../logging:rtc_event_log_api", |
| "../logging:rtc_event_log_impl_base", |
| "../modules/audio_device:mock_audio_device", |
| "../modules/audio_processing:audio_processing", |
| "../modules/video_coding:simulcast_test_fixture_impl", |
| "../p2p:p2p_test_utils", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_tests_main", |
| "../test:audio_codec_mocks", |
| "../test:test_support", |
| "../test:video_test_common", |
| ] |
| } |
| } |
