| # 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" ] |
| } |
| |
| rtc_source_set("rtc_media_config") { |
| visibility = [ "*" ] |
| sources = [ "base/media_config.h" ] |
| } |
| |
| rtc_library("rtc_sdp_video_format_utils") { |
| visibility = [ "*" ] |
| sources = [ |
| "base/sdp_video_format_utils.cc", |
| "base/sdp_video_format_utils.h", |
| ] |
| |
| deps = [ |
| "../api/video_codecs:video_codecs_api", |
| "../rtc_base:checks", |
| "../rtc_base:stringutils", |
| ] |
| absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] |
| } |
| |
| rtc_library("rtc_media_base") { |
| visibility = [ "*" ] |
| defines = [] |
| libs = [] |
| deps = [ |
| ":rtc_media_config", |
| "../api:array_view", |
| "../api:audio_options_api", |
| "../api:frame_transformer_interface", |
| "../api:media_stream_interface", |
| "../api:rtc_error", |
| "../api:rtp_parameters", |
| "../api:scoped_refptr", |
| "../api:sequence_checker", |
| "../api/audio:audio_frame_processor", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/crypto:frame_decryptor_interface", |
| "../api/crypto:frame_encryptor_interface", |
| "../api/crypto:options", |
| "../api/transport:datagram_transport_interface", |
| "../api/transport:stun_types", |
| "../api/transport:webrtc_key_value_config", |
| "../api/transport/rtp:rtp_source", |
| "../api/units:time_delta", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_bitrate_allocator_factory", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../api/video_codecs:video_codecs_api", |
| "../call:call_interfaces", |
| "../call:video_stream_api", |
| "../common_video", |
| "../modules/async_audio_processing", |
| "../modules/audio_processing:audio_processing_statistics", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:sanitizer", |
| "../rtc_base:socket", |
| "../rtc_base:stringutils", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/system:file_wrapper", |
| "../rtc_base/system:no_unique_address", |
| "../rtc_base/system:rtc_export", |
| "../rtc_base/task_utils:pending_task_safety_flag", |
| "../rtc_base/task_utils:to_queued_task", |
| "../rtc_base/third_party/sigslot", |
| "../system_wrappers:field_trial", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| sources = [ |
| "base/adapted_video_track_source.cc", |
| "base/adapted_video_track_source.h", |
| "base/audio_source.h", |
| "base/codec.cc", |
| "base/codec.h", |
| "base/delayable.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/rid_description.cc", |
| "base/rid_description.h", |
| "base/rtp_utils.cc", |
| "base/rtp_utils.h", |
| "base/stream_params.cc", |
| "base/stream_params.h", |
| "base/turn_utils.cc", |
| "base/turn_utils.h", |
| "base/video_adapter.cc", |
| "base/video_adapter.h", |
| "base/video_broadcaster.cc", |
| "base/video_broadcaster.h", |
| "base/video_common.cc", |
| "base/video_common.h", |
| "base/video_source_base.cc", |
| "base/video_source_base.h", |
| ] |
| } |
| |
| rtc_library("rtc_simulcast_encoder_adapter") { |
| visibility = [ "*" ] |
| defines = [] |
| libs = [] |
| sources = [ |
| "engine/simulcast_encoder_adapter.cc", |
| "engine/simulcast_encoder_adapter.h", |
| ] |
| deps = [ |
| ":rtc_media_base", |
| "../api:fec_controller_api", |
| "../api:scoped_refptr", |
| "../api:sequence_checker", |
| "../api/video:video_codec_constants", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../api/video_codecs:rtc_software_fallback_wrappers", |
| "../api/video_codecs:video_codecs_api", |
| "../call:video_stream_api", |
| "../common_video", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:video_coding_utility", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base/experiments:encoder_info_settings", |
| "../rtc_base/experiments:rate_control_settings", |
| "../rtc_base/system:no_unique_address", |
| "../rtc_base/system:rtc_export", |
| "../system_wrappers", |
| "../system_wrappers:field_trial", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| rtc_library("rtc_encoder_simulcast_proxy") { |
| visibility = [ "*" ] |
| defines = [] |
| libs = [] |
| sources = [ |
| "engine/encoder_simulcast_proxy.cc", |
| "engine/encoder_simulcast_proxy.h", |
| ] |
| deps = [ |
| ":rtc_simulcast_encoder_adapter", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../api/video_codecs:video_codecs_api", |
| "../modules/video_coding:video_codec_interface", |
| "../rtc_base/system:rtc_export", |
| ] |
| } |
| |
| rtc_library("rtc_internal_video_codecs") { |
| visibility = [ "*" ] |
| allow_poison = [ "software_video_codecs" ] |
| defines = [] |
| libs = [] |
| deps = [ |
| ":rtc_encoder_simulcast_proxy", |
| ":rtc_media_base", |
| ":rtc_simulcast_encoder_adapter", |
| "../api/video:encoded_image", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../api/video_codecs:rtc_software_fallback_wrappers", |
| "../api/video_codecs:video_codecs_api", |
| "../call:call_interfaces", |
| "../call:video_stream_api", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:webrtc_h264", |
| "../modules/video_coding:webrtc_multiplex", |
| "../modules/video_coding:webrtc_vp8", |
| "../modules/video_coding:webrtc_vp9", |
| "../modules/video_coding/codecs/av1:libaom_av1_decoder", |
| "../modules/video_coding/codecs/av1:libaom_av1_encoder_if_supported", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base/system:rtc_export", |
| "../system_wrappers:field_trial", |
| "../test:fake_video_codecs", |
| ] |
| if (rtc_include_dav1d_in_internal_decoder_factory) { |
| deps += [ "../modules/video_coding/codecs/av1:dav1d_decoder" ] |
| } |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| 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/multiplex_codec_factory.cc", |
| "engine/multiplex_codec_factory.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", |
| ] |
| } |
| |
| rtc_library("rtc_audio_video") { |
| visibility = [ "*" ] |
| allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove. |
| defines = [] |
| libs = [] |
| deps = [ |
| ":rtc_media_base", |
| "../api:call_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:media_stream_interface", |
| "../api:rtp_parameters", |
| "../api:scoped_refptr", |
| "../api:sequence_checker", |
| "../api:transport_api", |
| "../api/audio:audio_frame_processor", |
| "../api/audio:audio_mixer_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/task_queue", |
| "../api/transport:bitrate_settings", |
| "../api/transport:field_trial_based_config", |
| "../api/transport:webrtc_key_value_config", |
| "../api/transport/rtp:rtp_source", |
| "../api/units:data_rate", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_bitrate_allocator_factory", |
| "../api/video:video_codec_constants", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../api/video_codecs:rtc_software_fallback_wrappers", |
| "../api/video_codecs:video_codecs_api", |
| "../call", |
| "../call:call_interfaces", |
| "../call:video_stream_api", |
| "../common_video", |
| "../modules/async_audio_processing:async_audio_processing", |
| "../modules/audio_device", |
| "../modules/audio_device:audio_device_impl", |
| "../modules/audio_mixer:audio_mixer_impl", |
| "../modules/audio_processing:api", |
| "../modules/audio_processing/aec_dump", |
| "../modules/audio_processing/agc:gain_control_interface", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/video_coding", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:video_coding_utility", |
| "../rtc_base", |
| "../rtc_base:audio_format_to_string", |
| "../rtc_base:checks", |
| "../rtc_base:ignore_wundef", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:stringutils", |
| "../rtc_base:threading", |
| "../rtc_base/experiments:field_trial_parser", |
| "../rtc_base/experiments:min_video_bitrate_experiment", |
| "../rtc_base/experiments:normalize_simulcast_size_experiment", |
| "../rtc_base/experiments:rate_control_settings", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/system:rtc_export", |
| "../rtc_base/task_utils:pending_task_safety_flag", |
| "../rtc_base/task_utils:to_queued_task", |
| "../rtc_base/third_party/base64", |
| "../system_wrappers", |
| "../system_wrappers:metrics", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| sources = [ |
| "engine/adm_helpers.cc", |
| "engine/adm_helpers.h", |
| "engine/null_webrtc_video_engine.h", |
| "engine/payload_type_mapper.cc", |
| "engine/payload_type_mapper.h", |
| "engine/simulcast.cc", |
| "engine/simulcast.h", |
| "engine/unhandled_packets_buffer.cc", |
| "engine/unhandled_packets_buffer.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", |
| ] |
| |
| public_configs = [] |
| if (!build_with_chromium) { |
| public_configs += [ ":rtc_media_defines_config" ] |
| deps += [ "../modules/video_capture:video_capture_internal_impl" ] |
| } |
| if (rtc_enable_protobuf) { |
| deps += [ |
| "../modules/audio_coding:ana_config_proto", |
| "../modules/audio_processing/aec_dump:aec_dump_impl", |
| ] |
| } else { |
| deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ] |
| } |
| } |
| |
| # Heavy but optional helper for unittests and webrtc users who prefer to use |
| # defaults factories or do not worry about extra dependencies and binary size. |
| rtc_library("rtc_media_engine_defaults") { |
| visibility = [ "*" ] |
| allow_poison = [ |
| "audio_codecs", |
| "default_task_queue", |
| "software_video_codecs", |
| ] |
| sources = [ |
| "engine/webrtc_media_engine_defaults.cc", |
| "engine/webrtc_media_engine_defaults.h", |
| ] |
| deps = [ |
| ":rtc_audio_video", |
| "../api/audio_codecs:builtin_audio_decoder_factory", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../api/task_queue:default_task_queue_factory", |
| "../api/video:builtin_video_bitrate_allocator_factory", |
| "../api/video_codecs:builtin_video_decoder_factory", |
| "../api/video_codecs:builtin_video_encoder_factory", |
| "../modules/audio_processing:api", |
| "../rtc_base:checks", |
| "../rtc_base/system:rtc_export", |
| ] |
| } |
| |
| rtc_source_set("rtc_data_sctp_transport_internal") { |
| sources = [ "sctp/sctp_transport_internal.h" ] |
| deps = [ |
| "../api/transport:datagram_transport_interface", |
| "../media:rtc_media_base", |
| "../p2p:rtc_p2p", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:threading", |
| "../rtc_base/third_party/sigslot", |
| ] |
| } |
| |
| if (rtc_build_dcsctp) { |
| rtc_library("rtc_data_dcsctp_transport") { |
| sources = [ |
| "sctp/dcsctp_transport.cc", |
| "sctp/dcsctp_transport.h", |
| ] |
| deps = [ |
| ":rtc_data_sctp_transport_internal", |
| "../api:array_view", |
| "../media:rtc_media_base", |
| "../net/dcsctp/public:factory", |
| "../net/dcsctp/public:socket", |
| "../net/dcsctp/public:types", |
| "../net/dcsctp/public:utils", |
| "../net/dcsctp/timer:task_queue_timeout", |
| "../p2p:rtc_p2p", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:socket", |
| "../rtc_base:threading", |
| "../rtc_base/task_utils:pending_task_safety_flag", |
| "../rtc_base/task_utils:to_queued_task", |
| "../rtc_base/third_party/sigslot:sigslot", |
| "../system_wrappers", |
| ] |
| absl_deps += [ |
| "//third_party/abseil-cpp/absl/strings:strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| } |
| |
| if (rtc_build_usrsctp) { |
| rtc_library("rtc_data_usrsctp_transport") { |
| defines = [ |
| # "SCTP_DEBUG" # Uncomment for SCTP debugging. |
| ] |
| sources = [ |
| "sctp/usrsctp_transport.cc", |
| "sctp/usrsctp_transport.h", |
| ] |
| deps = [ |
| ":rtc_data_sctp_transport_internal", |
| "../media:rtc_media_base", |
| "../p2p:rtc_p2p", |
| "../rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:threading", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/task_utils:pending_task_safety_flag", |
| "../rtc_base/task_utils:to_queued_task", |
| "../rtc_base/third_party/sigslot:sigslot", |
| "//third_party/usrsctp", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| } |
| |
| rtc_library("rtc_data_sctp_transport_factory") { |
| defines = [] |
| sources = [ |
| "sctp/sctp_transport_factory.cc", |
| "sctp/sctp_transport_factory.h", |
| ] |
| deps = [ |
| ":rtc_data_sctp_transport_internal", |
| "../api/transport:sctp_transport_factory_interface", |
| "../rtc_base:threading", |
| "../rtc_base/experiments:field_trial_parser", |
| "../rtc_base/system:unused", |
| ] |
| |
| if (rtc_enable_sctp) { |
| assert(rtc_build_dcsctp || rtc_build_usrsctp, |
| "An SCTP backend is required to enable SCTP") |
| } |
| |
| if (rtc_build_dcsctp) { |
| defines += [ "WEBRTC_HAVE_DCSCTP" ] |
| deps += [ |
| ":rtc_data_dcsctp_transport", |
| "../system_wrappers", |
| "../system_wrappers:field_trial", |
| ] |
| } |
| |
| if (rtc_build_usrsctp) { |
| defines += [ "WEBRTC_HAVE_USRSCTP" ] |
| deps += [ ":rtc_data_usrsctp_transport" ] |
| } |
| } |
| |
| rtc_source_set("rtc_media") { |
| visibility = [ "*" ] |
| allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove. |
| deps = [ ":rtc_audio_video" ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_library("rtc_media_tests_utils") { |
| testonly = true |
| |
| defines = [] |
| deps = [ |
| ":rtc_audio_video", |
| ":rtc_internal_video_codecs", |
| ":rtc_media", |
| ":rtc_media_base", |
| ":rtc_simulcast_encoder_adapter", |
| "../api:call_api", |
| "../api:fec_controller_api", |
| "../api:scoped_refptr", |
| "../api/transport:field_trial_based_config", |
| "../api/video:encoded_image", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../api/video_codecs:video_codecs_api", |
| "../call:call_interfaces", |
| "../call:mock_rtp_interfaces", |
| "../call:video_stream_api", |
| "../common_video", |
| "../modules/audio_processing", |
| "../modules/audio_processing:api", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:video_coding_utility", |
| "../p2p:rtc_p2p", |
| "../rtc_base", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:stringutils", |
| "../rtc_base:threading", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/third_party/sigslot", |
| "../test:test_support", |
| "//testing/gtest", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//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_renderer.cc", |
| "base/fake_video_renderer.h", |
| "base/test_utils.cc", |
| "base/test_utils.h", |
| "engine/fake_webrtc_call.cc", |
| "engine/fake_webrtc_call.h", |
| "engine/fake_webrtc_video_engine.cc", |
| "engine/fake_webrtc_video_engine.h", |
| ] |
| } |
| |
| if (!build_with_chromium) { |
| rtc_media_unittests_resources = [ |
| "../resources/media/captured-320x240-2s-48.frames", |
| "../resources/media/faces.1280x720_P420.yuv", |
| "../resources/media/faces_I400.jpg", |
| "../resources/media/faces_I411.jpg", |
| "../resources/media/faces_I420.jpg", |
| "../resources/media/faces_I422.jpg", |
| "../resources/media/faces_I444.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_encoder_simulcast_proxy", |
| ":rtc_internal_video_codecs", |
| ":rtc_media", |
| ":rtc_media_base", |
| ":rtc_media_engine_defaults", |
| ":rtc_media_tests_utils", |
| ":rtc_sdp_video_format_utils", |
| ":rtc_simulcast_encoder_adapter", |
| "../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:mock_video_encoder", |
| "../api:rtp_parameters", |
| "../api:scoped_refptr", |
| "../api:simulcast_test_fixture_api", |
| "../api/audio_codecs:builtin_audio_decoder_factory", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../api/rtc_event_log", |
| "../api/task_queue", |
| "../api/task_queue:default_task_queue_factory", |
| "../api/test/video:function_video_factory", |
| "../api/transport:field_trial_based_config", |
| "../api/units:time_delta", |
| "../api/units:timestamp", |
| "../api/video:builtin_video_bitrate_allocator_factory", |
| "../api/video:video_bitrate_allocation", |
| "../api/video:video_codec_constants", |
| "../api/video:video_frame", |
| "../api/video:video_rtp_headers", |
| "../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", |
| "../modules/audio_device:mock_audio_device", |
| "../modules/audio_processing", |
| "../modules/audio_processing:api", |
| "../modules/audio_processing:mocks", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/video_coding:simulcast_test_fixture_impl", |
| "../modules/video_coding:video_codec_interface", |
| "../modules/video_coding:webrtc_h264", |
| "../modules/video_coding:webrtc_vp8", |
| "../modules/video_coding/codecs/av1:libaom_av1_decoder", |
| "../modules/video_coding/codecs/av1:libaom_av1_encoder_if_supported", |
| "../p2p:p2p_test_utils", |
| "../rtc_base", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_tests_utils", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:stringutils", |
| "../rtc_base:threading", |
| "../rtc_base/experiments:min_video_bitrate_experiment", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/third_party/sigslot", |
| "../system_wrappers:field_trial", |
| "../test:audio_codec_mocks", |
| "../test:fake_video_codecs", |
| "../test:field_trial", |
| "../test:rtp_test_utils", |
| "../test:test_main", |
| "../test:test_support", |
| "../test:video_test_common", |
| "../test/time_controller", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| sources = [ |
| "base/codec_unittest.cc", |
| "base/media_engine_unittest.cc", |
| "base/rtp_utils_unittest.cc", |
| "base/sdp_video_format_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_common_unittest.cc", |
| "engine/encoder_simulcast_proxy_unittest.cc", |
| "engine/internal_decoder_factory_unittest.cc", |
| "engine/internal_encoder_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/unhandled_packets_buffer_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_build_usrsctp) { |
| sources += [ |
| "sctp/usrsctp_transport_reliability_unittest.cc", |
| "sctp/usrsctp_transport_unittest.cc", |
| ] |
| deps += [ |
| ":rtc_data_sctp_transport_internal", |
| ":rtc_data_usrsctp_transport", |
| "../rtc_base:rtc_event", |
| "../rtc_base/task_utils:pending_task_safety_flag", |
| "../rtc_base/task_utils:to_queued_task", |
| "//third_party/usrsctp", |
| ] |
| } |
| |
| if (rtc_opus_support_120ms_ptime) { |
| defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ] |
| } else { |
| defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] |
| } |
| |
| 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" ] |
| } |
| } |
| } |
| } |
