diff options
Diffstat (limited to 'darwin_x86_64.cmake')
| -rw-r--r-- | darwin_x86_64.cmake | 545 |
1 files changed, 416 insertions, 129 deletions
diff --git a/darwin_x86_64.cmake b/darwin_x86_64.cmake index 4670cd58be..2777aa47ff 100644 --- a/darwin_x86_64.cmake +++ b/darwin_x86_64.cmake @@ -1,4 +1,4 @@ -# Generated on 04/21/21 for target: Darwin +# Generated on 06/23/21 for target: Darwin # This is an autogenerated file by calling: # ./import-webrtc.py --target webrtc_api_video_codecs_builtin_video_decoder_factory --target webrtc_api_video_codecs_builtin_video_encoder_factory --target webrtc_api_libjingle_peerconnection_api --target webrtc_pc_peerconnection --target webrtc_api_create_peerconnection_factory --target webrtc_api_audio_codecs_builtin_audio_decoder_factory --target webrtc_api_audio_codecs_builtin_audio_encoder_factory --target webrtc_common_audio_common_audio_unittests --target webrtc_common_video_common_video_unittests --target webrtc_media_rtc_media_unittests --target webrtc_modules_audio_coding_audio_decoder_unittests --target webrtc_pc_peerconnection_unittests --target webrtc_pc_rtc_pc_unittests --root /usr/local/google/home/jansene/src/webrtc_g3/ --platform Darwin BUILD . @@ -6,11 +6,11 @@ # Re-running this script will require you to merge in the latest upstream-master for webrtc # Expecting jsoncpp at 9059f5cad030ba11d37818847443a53918c327b1 -# Expecting libaom at 6c93db7ff63430d2e5dcdfc42e84e3a0514da608 -# Expecting libsrtp2 at 7990ca64c616b150a9cb4714601c4a3b0c84fe91 +# Expecting libaom at 12287adee94fc3b1f5349d3f4bd85cea4e57f62b +# Expecting libsrtp2 at 5b7c744eb8310250ccc534f3f86a2015b3887a0a # Expecting libvpx at 61edec1efbea1c02d71857e2aff9426d9cd2df4e -# Expecting libyuv at 64994843e652443df2d5201c6ae3fb725097360f -# Expecting usrsctp at 70d42ae95a1de83bd317c8cc9503f894671d1392 +# Expecting libyuv at 49ebc996aa8c4bdf89c1b5ea461eb677234c61cc +# Expecting usrsctp at 22ba62ffe79c3881581ab430368bf3764d9533eb @@ -294,12 +294,12 @@ target_include_directories(webrtc_api_call_api INTERFACE ${WEBRTC_ROOT} ${CMAKE_ # api:callfactory_api add_library(webrtc_api_callfactory_api INTERFACE) -target_link_libraries(webrtc_api_callfactory_api INTERFACE webrtc_rtc_base_system_rtc_export ) +target_link_libraries(webrtc_api_callfactory_api INTERFACE webrtc_call_rtp_interfaces webrtc_rtc_base_system_rtc_export ) target_include_directories(webrtc_api_callfactory_api INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:callfactory_api.headers add_library(webrtc_api_callfactory_api.headers INTERFACE) -target_link_libraries(webrtc_api_callfactory_api.headers INTERFACE webrtc_rtc_base_system_rtc_export.headers ) +target_link_libraries(webrtc_api_callfactory_api.headers INTERFACE webrtc_call_rtp_interfaces.headers webrtc_rtc_base_system_rtc_export.headers ) target_include_directories(webrtc_api_callfactory_api.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:create_frame_generator @@ -328,7 +328,7 @@ add_library(webrtc_api_create_time_controller ${WEBRTC_ROOT}/api/test/create_tim target_include_directories(webrtc_api_create_time_controller PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_api_create_time_controller PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_api_create_time_controller PRIVATE -fno-exceptions) -target_link_libraries(webrtc_api_create_time_controller PUBLIC webrtc_api_callfactory_api webrtc_api_time_controller webrtc_call_call webrtc_call_call_interfaces webrtc_test_time_controller_time_controller ) +target_link_libraries(webrtc_api_create_time_controller PUBLIC webrtc_api_callfactory_api webrtc_api_time_controller webrtc_call_call webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_test_time_controller_time_controller ) # api/crypto:frame_decryptor_interface add_library(webrtc_api_crypto_frame_decryptor_interface INTERFACE) @@ -429,15 +429,15 @@ add_library(webrtc_api_libjingle_logging_api.headers INTERFACE) target_include_directories(webrtc_api_libjingle_logging_api.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:libjingle_peerconnection_api -add_library(webrtc_api_libjingle_peerconnection_api ${WEBRTC_ROOT}/api/candidate.cc ${WEBRTC_ROOT}/api/data_channel_interface.cc ${WEBRTC_ROOT}/api/dtls_transport_interface.cc ${WEBRTC_ROOT}/api/jsep.cc ${WEBRTC_ROOT}/api/jsep_ice_candidate.cc ${WEBRTC_ROOT}/api/peer_connection_interface.cc ${WEBRTC_ROOT}/api/proxy.cc ${WEBRTC_ROOT}/api/rtp_receiver_interface.cc ${WEBRTC_ROOT}/api/rtp_sender_interface.cc ${WEBRTC_ROOT}/api/rtp_transceiver_interface.cc ${WEBRTC_ROOT}/api/sctp_transport_interface.cc ${WEBRTC_ROOT}/api/stats_types.cc) +add_library(webrtc_api_libjingle_peerconnection_api ${WEBRTC_ROOT}/api/candidate.cc ${WEBRTC_ROOT}/api/data_channel_interface.cc ${WEBRTC_ROOT}/api/dtls_transport_interface.cc ${WEBRTC_ROOT}/api/jsep.cc ${WEBRTC_ROOT}/api/jsep_ice_candidate.cc ${WEBRTC_ROOT}/api/peer_connection_interface.cc ${WEBRTC_ROOT}/api/rtp_receiver_interface.cc ${WEBRTC_ROOT}/api/rtp_sender_interface.cc ${WEBRTC_ROOT}/api/rtp_transceiver_interface.cc ${WEBRTC_ROOT}/api/sctp_transport_interface.cc ${WEBRTC_ROOT}/api/stats_types.cc) target_include_directories(webrtc_api_libjingle_peerconnection_api PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_api_libjingle_peerconnection_api PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_api_libjingle_peerconnection_api PRIVATE -fno-exceptions) -target_link_libraries(webrtc_api_libjingle_peerconnection_api PUBLIC absl::algorithm absl::algorithm_container absl::config absl::core_headers absl::algorithm_container absl::memory absl::type_traits absl::strings absl::optional absl::variant webrtc_api_array_view webrtc_api_async_dns_resolver webrtc_api_audio_options_api webrtc_api_callfactory_api webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_libjingle_logging_api webrtc_api_media_stream_interface webrtc_api_network_state_predictor_api webrtc_api_packet_socket_factory webrtc_api_priority webrtc_api_rtc_error webrtc_api_rtc_stats_api webrtc_api_rtp_packet_info webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_adaptation_resource_adaptation_api webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_enums webrtc_api_transport_network_control webrtc_api_transport_sctp_transport_factory_interface webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_data_rate webrtc_api_units_timestamp webrtc_api_video_encoded_image webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_media_rtc_media_base webrtc_media_rtc_media_base.headers webrtc_media_rtc_media_config webrtc_modules_audio_processing_audio_processing_statistics webrtc_p2p_rtc_p2p.headers webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_system_rtc_export ) +target_link_libraries(webrtc_api_libjingle_peerconnection_api PUBLIC absl::algorithm absl::algorithm_container absl::config absl::core_headers absl::algorithm_container absl::memory absl::type_traits absl::strings absl::optional absl::variant webrtc_api_array_view webrtc_api_async_dns_resolver webrtc_api_audio_options_api webrtc_api_callfactory_api webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_libjingle_logging_api webrtc_api_media_stream_interface webrtc_api_network_state_predictor_api webrtc_api_packet_socket_factory webrtc_api_priority webrtc_api_rtc_error webrtc_api_rtc_stats_api webrtc_api_rtp_packet_info webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_adaptation_resource_adaptation_api webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_enums webrtc_api_transport_network_control webrtc_api_transport_sctp_transport_factory_interface webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_data_rate webrtc_api_units_timestamp webrtc_api_video_encoded_image webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_call_rtp_interfaces webrtc_media_rtc_media_base webrtc_media_rtc_media_base.headers webrtc_media_rtc_media_config webrtc_modules_audio_processing_audio_processing_statistics webrtc_p2p_rtc_p2p.headers webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_system_rtc_export ) # api:libjingle_peerconnection_api.headers add_library(webrtc_api_libjingle_peerconnection_api.headers INTERFACE) -target_link_libraries(webrtc_api_libjingle_peerconnection_api.headers INTERFACE webrtc_api_array_view.headers webrtc_api_async_dns_resolver.headers webrtc_api_audio_options_api.headers webrtc_api_callfactory_api.headers webrtc_api_fec_controller_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_libjingle_logging_api.headers webrtc_api_media_stream_interface.headers webrtc_api_network_state_predictor_api.headers webrtc_api_packet_socket_factory.headers webrtc_api_priority.headers webrtc_api_rtc_error.headers webrtc_api_rtc_stats_api.headers webrtc_api_rtp_packet_info.headers webrtc_api_rtp_parameters.headers webrtc_api_rtp_transceiver_direction.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_adaptation_resource_adaptation_api.headers webrtc_api_audio_audio_mixer_api.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_neteq_neteq_api.headers webrtc_api_rtc_event_log_rtc_event_log.headers webrtc_api_task_queue_task_queue.headers webrtc_api_transport_bitrate_settings.headers webrtc_api_transport_enums.headers webrtc_api_transport_network_control.headers webrtc_api_transport_sctp_transport_factory_interface.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_api_transport_rtp_rtp_source.headers webrtc_api_units_data_rate.headers webrtc_api_units_timestamp.headers webrtc_api_video_encoded_image.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_media_rtc_media_base.headers webrtc_media_rtc_media_config.headers webrtc_modules_audio_processing_audio_processing_statistics.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_ip_address.headers webrtc_rtc_base_network_constants.headers webrtc_rtc_base_rtc_base.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_socket_address.headers webrtc_rtc_base_threading.headers webrtc_rtc_base_system_rtc_export.headers ) +target_link_libraries(webrtc_api_libjingle_peerconnection_api.headers INTERFACE webrtc_api_array_view.headers webrtc_api_async_dns_resolver.headers webrtc_api_audio_options_api.headers webrtc_api_callfactory_api.headers webrtc_api_fec_controller_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_libjingle_logging_api.headers webrtc_api_media_stream_interface.headers webrtc_api_network_state_predictor_api.headers webrtc_api_packet_socket_factory.headers webrtc_api_priority.headers webrtc_api_rtc_error.headers webrtc_api_rtc_stats_api.headers webrtc_api_rtp_packet_info.headers webrtc_api_rtp_parameters.headers webrtc_api_rtp_transceiver_direction.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_adaptation_resource_adaptation_api.headers webrtc_api_audio_audio_mixer_api.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_neteq_neteq_api.headers webrtc_api_rtc_event_log_rtc_event_log.headers webrtc_api_task_queue_task_queue.headers webrtc_api_transport_bitrate_settings.headers webrtc_api_transport_enums.headers webrtc_api_transport_network_control.headers webrtc_api_transport_sctp_transport_factory_interface.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_api_transport_rtp_rtp_source.headers webrtc_api_units_data_rate.headers webrtc_api_units_timestamp.headers webrtc_api_video_encoded_image.headers webrtc_api_video_video_bitrate_allocator_factory.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_call_rtp_interfaces.headers webrtc_media_rtc_media_base.headers webrtc_media_rtc_media_config.headers webrtc_modules_audio_processing_audio_processing_statistics.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_ip_address.headers webrtc_rtc_base_network_constants.headers webrtc_rtc_base_rtc_base.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_socket_address.headers webrtc_rtc_base_threading.headers webrtc_rtc_base_system_rtc_export.headers ) target_include_directories(webrtc_api_libjingle_peerconnection_api.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:media_stream_interface @@ -573,12 +573,12 @@ target_include_directories(webrtc_api_priority.headers INTERFACE ${WEBRTC_ROOT} # api:refcountedbase add_library(webrtc_api_refcountedbase INTERFACE) -target_link_libraries(webrtc_api_refcountedbase INTERFACE webrtc_rtc_base_rtc_base_approved ) +target_link_libraries(webrtc_api_refcountedbase INTERFACE webrtc_rtc_base_macromagic webrtc_rtc_base_refcount ) target_include_directories(webrtc_api_refcountedbase INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:refcountedbase.headers add_library(webrtc_api_refcountedbase.headers INTERFACE) -target_link_libraries(webrtc_api_refcountedbase.headers INTERFACE webrtc_rtc_base_rtc_base_approved.headers ) +target_link_libraries(webrtc_api_refcountedbase.headers INTERFACE webrtc_rtc_base_macromagic.headers webrtc_rtc_base_refcount.headers ) target_include_directories(webrtc_api_refcountedbase.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:rtc_error @@ -621,12 +621,12 @@ target_link_libraries(webrtc_api_rtc_event_log_rtc_event_log_factory PUBLIC webr # api:rtc_stats_api add_library(webrtc_api_rtc_stats_api INTERFACE) -target_link_libraries(webrtc_api_rtc_stats_api INTERFACE webrtc_api_scoped_refptr webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export ) +target_link_libraries(webrtc_api_rtc_stats_api INTERFACE webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export ) target_include_directories(webrtc_api_rtc_stats_api INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:rtc_stats_api.headers add_library(webrtc_api_rtc_stats_api.headers INTERFACE) -target_link_libraries(webrtc_api_rtc_stats_api.headers INTERFACE webrtc_api_scoped_refptr.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_system_rtc_export.headers ) +target_link_libraries(webrtc_api_rtc_stats_api.headers INTERFACE webrtc_api_refcountedbase.headers webrtc_api_scoped_refptr.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_system_rtc_export.headers ) target_include_directories(webrtc_api_rtc_stats_api.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:rtp_headers @@ -646,11 +646,11 @@ add_library(webrtc_api_rtp_packet_info ${WEBRTC_ROOT}/api/rtp_packet_info.cc) target_include_directories(webrtc_api_rtp_packet_info PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_api_rtp_packet_info PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_api_rtp_packet_info PRIVATE -fno-exceptions) -target_link_libraries(webrtc_api_rtp_packet_info PUBLIC absl::optional webrtc_api_array_view webrtc_api_refcountedbase webrtc_api_rtp_headers webrtc_api_scoped_refptr webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export ) +target_link_libraries(webrtc_api_rtp_packet_info PUBLIC absl::optional webrtc_api_array_view webrtc_api_refcountedbase webrtc_api_rtp_headers webrtc_api_scoped_refptr webrtc_api_units_timestamp webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export ) # api:rtp_packet_info.headers add_library(webrtc_api_rtp_packet_info.headers INTERFACE) -target_link_libraries(webrtc_api_rtp_packet_info.headers INTERFACE webrtc_api_array_view.headers webrtc_api_refcountedbase.headers webrtc_api_rtp_headers.headers webrtc_api_scoped_refptr.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_system_rtc_export.headers ) +target_link_libraries(webrtc_api_rtp_packet_info.headers INTERFACE webrtc_api_array_view.headers webrtc_api_refcountedbase.headers webrtc_api_rtp_headers.headers webrtc_api_scoped_refptr.headers webrtc_api_units_timestamp.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_system_rtc_export.headers ) target_include_directories(webrtc_api_rtp_packet_info.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api:rtp_parameters @@ -767,6 +767,11 @@ add_library(webrtc_api_transport_datagram_transport_interface INTERFACE) target_link_libraries(webrtc_api_transport_datagram_transport_interface INTERFACE absl::optional webrtc_api_array_view webrtc_api_rtc_error webrtc_rtc_base_rtc_base_approved ) target_include_directories(webrtc_api_transport_datagram_transport_interface INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +# api/transport:datagram_transport_interface.headers +add_library(webrtc_api_transport_datagram_transport_interface.headers INTERFACE) +target_link_libraries(webrtc_api_transport_datagram_transport_interface.headers INTERFACE webrtc_api_array_view.headers webrtc_api_rtc_error.headers webrtc_rtc_base_rtc_base_approved.headers ) +target_include_directories(webrtc_api_transport_datagram_transport_interface.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + # api/transport:enums add_library(webrtc_api_transport_enums INTERFACE) target_include_directories(webrtc_api_transport_enums INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) @@ -957,15 +962,15 @@ target_compile_options(webrtc_api_video_codecs_rtc_software_fallback_wrappers PR target_link_libraries(webrtc_api_video_codecs_rtc_software_fallback_wrappers PUBLIC absl::core_headers absl::strings absl::optional webrtc_api_fec_controller_api webrtc_api_video_encoded_image webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_media_rtc_media_base webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) # api/video_codecs:video_codecs_api -add_library(webrtc_api_video_codecs_video_codecs_api ${WEBRTC_ROOT}/api/video_codecs/h264_profile_level_id.cc ${WEBRTC_ROOT}/api/video_codecs/sdp_video_format.cc ${WEBRTC_ROOT}/api/video_codecs/spatial_layer.cc ${WEBRTC_ROOT}/api/video_codecs/video_codec.cc ${WEBRTC_ROOT}/api/video_codecs/video_decoder.cc ${WEBRTC_ROOT}/api/video_codecs/video_decoder_factory.cc ${WEBRTC_ROOT}/api/video_codecs/video_encoder.cc ${WEBRTC_ROOT}/api/video_codecs/video_encoder_config.cc ${WEBRTC_ROOT}/api/video_codecs/vp8_frame_config.cc ${WEBRTC_ROOT}/api/video_codecs/vp8_temporal_layers.cc ${WEBRTC_ROOT}/api/video_codecs/vp9_profile.cc) +add_library(webrtc_api_video_codecs_video_codecs_api ${WEBRTC_ROOT}/api/video_codecs/h264_profile_level_id.cc ${WEBRTC_ROOT}/api/video_codecs/sdp_video_format.cc ${WEBRTC_ROOT}/api/video_codecs/spatial_layer.cc ${WEBRTC_ROOT}/api/video_codecs/video_codec.cc ${WEBRTC_ROOT}/api/video_codecs/video_decoder.cc ${WEBRTC_ROOT}/api/video_codecs/video_encoder.cc ${WEBRTC_ROOT}/api/video_codecs/video_encoder_config.cc ${WEBRTC_ROOT}/api/video_codecs/vp8_frame_config.cc ${WEBRTC_ROOT}/api/video_codecs/vp8_temporal_layers.cc ${WEBRTC_ROOT}/api/video_codecs/vp9_profile.cc) target_include_directories(webrtc_api_video_codecs_video_codecs_api PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_api_video_codecs_video_codecs_api PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_api_video_codecs_video_codecs_api PRIVATE -fno-exceptions) -target_link_libraries(webrtc_api_video_codecs_video_codecs_api PUBLIC absl::algorithm_container absl::algorithm_container absl::strings absl::optional webrtc_api_fec_controller_api webrtc_api_scoped_refptr webrtc_api_units_data_rate webrtc_api_video_encoded_image webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_modules_video_coding_codec_globals_headers webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export ) +target_link_libraries(webrtc_api_video_codecs_video_codecs_api PUBLIC absl::algorithm_container absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_fec_controller_api webrtc_api_scoped_refptr webrtc_api_units_data_rate webrtc_api_video_encoded_image webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_modules_video_coding_codec_globals_headers webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_rtc_export ) # api/video_codecs:video_codecs_api.headers add_library(webrtc_api_video_codecs_video_codecs_api.headers INTERFACE) -target_link_libraries(webrtc_api_video_codecs_video_codecs_api.headers INTERFACE webrtc_api_fec_controller_api.headers webrtc_api_scoped_refptr.headers webrtc_api_units_data_rate.headers webrtc_api_video_encoded_image.headers webrtc_api_video_video_bitrate_allocation.headers webrtc_api_video_video_codec_constants.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_modules_video_coding_codec_globals_headers.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_system_rtc_export.headers ) +target_link_libraries(webrtc_api_video_codecs_video_codecs_api.headers INTERFACE webrtc_api_array_view.headers webrtc_api_fec_controller_api.headers webrtc_api_scoped_refptr.headers webrtc_api_units_data_rate.headers webrtc_api_video_encoded_image.headers webrtc_api_video_video_bitrate_allocation.headers webrtc_api_video_video_codec_constants.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_modules_video_coding_codec_globals_headers.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_system_rtc_export.headers ) target_include_directories(webrtc_api_video_codecs_video_codecs_api.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # api/video_codecs:vp8_temporal_layers_factory @@ -1144,7 +1149,7 @@ add_library(webrtc_audio_audio ${WEBRTC_ROOT}/audio/audio_level.cc ${WEBRTC_ROOT target_include_directories(webrtc_audio_audio PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_audio_audio PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_audio_audio PRIVATE -fno-exceptions) -target_link_libraries(webrtc_audio_audio PUBLIC absl::memory absl::optional webrtc_api_array_view webrtc_api_call_api webrtc_api_frame_transformer_interface webrtc_api_function_view webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_transport_api webrtc_api_audio_aec3_factory webrtc_api_audio_audio_frame_api webrtc_api_audio_audio_frame_processor webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_rtp_rtp_source webrtc_audio_utility_audio_frame_operations webrtc_call_audio_sender_interface webrtc_call_bitrate_allocator webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_common_audio_common_audio webrtc_common_audio_common_audio_c webrtc_logging_rtc_event_audio webrtc_logging_rtc_stream_config webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_coding_audio_coding webrtc_modules_audio_coding_audio_coding_module_typedefs webrtc_modules_audio_coding_audio_encoder_cng webrtc_modules_audio_coding_audio_network_adaptor_config webrtc_modules_audio_coding_red webrtc_modules_audio_device_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_frame_proxies webrtc_modules_audio_processing_rms_level webrtc_modules_pacing_pacing webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_rtc_base_rtc_base webrtc_rtc_base_audio_format_to_string webrtc_rtc_base_checks webrtc_rtc_base_rate_limiter webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_threading webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_task_utils_to_queued_task webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_audio_audio PUBLIC absl::memory absl::optional webrtc_api_array_view webrtc_api_call_api webrtc_api_frame_transformer_interface webrtc_api_function_view webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_transport_api webrtc_api_audio_aec3_factory webrtc_api_audio_audio_frame_api webrtc_api_audio_audio_frame_processor webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_rtp_rtp_source webrtc_audio_utility_audio_frame_operations webrtc_call_audio_sender_interface webrtc_call_bitrate_allocator webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_common_audio_common_audio webrtc_common_audio_common_audio_c webrtc_logging_rtc_event_audio webrtc_logging_rtc_stream_config webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_coding_audio_coding webrtc_modules_audio_coding_audio_coding_module_typedefs webrtc_modules_audio_coding_audio_encoder_cng webrtc_modules_audio_coding_audio_network_adaptor_config webrtc_modules_audio_coding_red webrtc_modules_audio_device_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_frame_proxies webrtc_modules_audio_processing_rms_level webrtc_modules_pacing_pacing webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_rtc_base_rtc_base webrtc_rtc_base_audio_format_to_string webrtc_rtc_base_checks webrtc_rtc_base_rate_limiter webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_threading webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) # audio/utility:audio_frame_operations add_library(webrtc_audio_utility_audio_frame_operations ${WEBRTC_ROOT}/audio/utility/audio_frame_operations.cc ${WEBRTC_ROOT}/audio/utility/channel_mixer.cc ${WEBRTC_ROOT}/audio/utility/channel_mixing_matrix.cc) @@ -1201,18 +1206,18 @@ add_library(webrtc_call_call ${WEBRTC_ROOT}/call/call.cc ${WEBRTC_ROOT}/call/cal target_include_directories(webrtc_call_call PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_call_call PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_call_call PRIVATE -fno-exceptions) -target_link_libraries(webrtc_call_call PUBLIC absl::optional webrtc_api_array_view webrtc_api_callfactory_api webrtc_api_fec_controller_api webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_sequence_checker webrtc_api_simulated_network_api webrtc_api_transport_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_transport_network_control webrtc_api_units_time_delta webrtc_api_video_codecs_video_codecs_api webrtc_audio_audio webrtc_call_bitrate_allocator webrtc_call_call_interfaces webrtc_call_fake_network webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_call_rtp_sender webrtc_call_simulated_network webrtc_call_version webrtc_call_video_stream_api webrtc_call_adaptation_resource_adaptation webrtc_logging_rtc_event_audio webrtc_logging_rtc_event_rtp_rtcp webrtc_logging_rtc_event_video webrtc_logging_rtc_stream_config webrtc_modules_congestion_controller_congestion_controller webrtc_modules_pacing_pacing webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_modules_video_coding_video_coding webrtc_rtc_base_checks webrtc_rtc_base_rate_limiter webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_network_sent_packet webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics webrtc_video_video ) +target_link_libraries(webrtc_call_call PUBLIC absl::bind_front absl::optional webrtc_api_array_view webrtc_api_callfactory_api webrtc_api_fec_controller_api webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_sequence_checker webrtc_api_simulated_network_api webrtc_api_transport_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_transport_network_control webrtc_api_units_time_delta webrtc_api_video_codecs_video_codecs_api webrtc_audio_audio webrtc_call_bitrate_allocator webrtc_call_call_interfaces webrtc_call_fake_network webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_call_rtp_sender webrtc_call_simulated_network webrtc_call_version webrtc_call_video_stream_api webrtc_call_adaptation_resource_adaptation webrtc_logging_rtc_event_audio webrtc_logging_rtc_event_rtp_rtcp webrtc_logging_rtc_event_video webrtc_logging_rtc_stream_config webrtc_modules_congestion_controller_congestion_controller webrtc_modules_pacing_pacing webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_modules_video_coding_video_coding webrtc_rtc_base_checks webrtc_rtc_base_rate_limiter webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_network_sent_packet webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics webrtc_video_video ) # call:call_interfaces add_library(webrtc_call_call_interfaces ${WEBRTC_ROOT}/call/audio_receive_stream.cc ${WEBRTC_ROOT}/call/audio_send_stream.cc ${WEBRTC_ROOT}/call/audio_state.cc ${WEBRTC_ROOT}/call/call_config.cc ${WEBRTC_ROOT}/call/flexfec_receive_stream.cc ${WEBRTC_ROOT}/call/syncable.cc) target_include_directories(webrtc_call_call_interfaces PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_call_call_interfaces PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_call_call_interfaces PRIVATE -fno-exceptions) -target_link_libraries(webrtc_call_call_interfaces PUBLIC absl::optional webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_network_state_predictor_api webrtc_api_rtc_error webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_transport_api webrtc_api_adaptation_resource_adaptation_api webrtc_api_audio_audio_frame_processor webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_network_control webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_call_audio_sender_interface webrtc_call_rtp_interfaces webrtc_call_video_stream_api webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_device_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_rtc_base_rtc_base webrtc_rtc_base_audio_format_to_string webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_network_sent_packet ) +target_link_libraries(webrtc_call_call_interfaces PUBLIC absl::bind_front absl::optional webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_network_state_predictor_api webrtc_api_rtc_error webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_transport_api webrtc_api_adaptation_resource_adaptation_api webrtc_api_audio_audio_frame_processor webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_network_control webrtc_api_transport_webrtc_key_value_config webrtc_call_audio_sender_interface webrtc_call_receive_stream_interface webrtc_call_rtp_interfaces webrtc_call_video_stream_api webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_device_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_rtc_base_rtc_base webrtc_rtc_base_audio_format_to_string webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_network_sent_packet ) # call:call_interfaces.headers add_library(webrtc_call_call_interfaces.headers INTERFACE) -target_link_libraries(webrtc_call_call_interfaces.headers INTERFACE webrtc_api_fec_controller_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_network_state_predictor_api.headers webrtc_api_rtc_error.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_transport_api.headers webrtc_api_adaptation_resource_adaptation_api.headers webrtc_api_audio_audio_frame_processor.headers webrtc_api_audio_audio_mixer_api.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_neteq_neteq_api.headers webrtc_api_task_queue_task_queue.headers webrtc_api_transport_bitrate_settings.headers webrtc_api_transport_network_control.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_api_transport_rtp_rtp_source.headers webrtc_call_audio_sender_interface.headers webrtc_call_rtp_interfaces.headers webrtc_call_video_stream_api.headers webrtc_modules_async_audio_processing_async_audio_processing.headers webrtc_modules_audio_device_audio_device.headers webrtc_modules_audio_processing_api.headers webrtc_modules_audio_processing_audio_processing.headers webrtc_modules_audio_processing_audio_processing_statistics.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_modules_utility_utility.headers webrtc_rtc_base_audio_format_to_string.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_network_sent_packet.headers ) +target_link_libraries(webrtc_call_call_interfaces.headers INTERFACE webrtc_api_fec_controller_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_network_state_predictor_api.headers webrtc_api_rtc_error.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_transport_api.headers webrtc_api_adaptation_resource_adaptation_api.headers webrtc_api_audio_audio_frame_processor.headers webrtc_api_audio_audio_mixer_api.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_neteq_neteq_api.headers webrtc_api_task_queue_task_queue.headers webrtc_api_transport_bitrate_settings.headers webrtc_api_transport_network_control.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_call_audio_sender_interface.headers webrtc_call_receive_stream_interface.headers webrtc_call_rtp_interfaces.headers webrtc_call_video_stream_api.headers webrtc_modules_async_audio_processing_async_audio_processing.headers webrtc_modules_audio_device_audio_device.headers webrtc_modules_audio_processing_api.headers webrtc_modules_audio_processing_audio_processing.headers webrtc_modules_audio_processing_audio_processing_statistics.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_modules_utility_utility.headers webrtc_rtc_base_audio_format_to_string.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_network_sent_packet.headers ) target_include_directories(webrtc_call_call_interfaces.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # call:fake_network @@ -1227,16 +1232,26 @@ add_library(webrtc_call_mock_rtp_interfaces INTERFACE) target_link_libraries(webrtc_call_mock_rtp_interfaces INTERFACE webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_transport_bitrate_settings webrtc_call_rtp_interfaces webrtc_modules_pacing_pacing webrtc_rtc_base_rtc_base webrtc_rtc_base_rate_limiter webrtc_rtc_base_network_sent_packet webrtc_test_test_support ) target_include_directories(webrtc_call_mock_rtp_interfaces INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +# call:receive_stream_interface +add_library(webrtc_call_receive_stream_interface INTERFACE) +target_link_libraries(webrtc_call_receive_stream_interface INTERFACE webrtc_api_frame_transformer_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_crypto_frame_decryptor_interface webrtc_api_transport_rtp_rtp_source ) +target_include_directories(webrtc_call_receive_stream_interface INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# call:receive_stream_interface.headers +add_library(webrtc_call_receive_stream_interface.headers INTERFACE) +target_link_libraries(webrtc_call_receive_stream_interface.headers INTERFACE webrtc_api_frame_transformer_interface.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_transport_rtp_rtp_source.headers ) +target_include_directories(webrtc_call_receive_stream_interface.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + # call:rtp_interfaces add_library(webrtc_call_rtp_interfaces ${WEBRTC_ROOT}/call/rtp_config.cc) target_include_directories(webrtc_call_rtp_interfaces PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_call_rtp_interfaces PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_call_rtp_interfaces PRIVATE -fno-exceptions) -target_link_libraries(webrtc_call_rtp_interfaces PUBLIC absl::algorithm_container absl::optional webrtc_api_array_view webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_transport_bitrate_settings webrtc_api_units_timestamp webrtc_common_video_frame_counts webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) +target_link_libraries(webrtc_call_rtp_interfaces PUBLIC absl::algorithm_container absl::optional webrtc_api_array_view webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_network_state_predictor_api webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_transport_bitrate_settings webrtc_api_transport_network_control webrtc_api_transport_webrtc_key_value_config webrtc_api_units_timestamp webrtc_common_video_frame_counts webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue ) # call:rtp_interfaces.headers add_library(webrtc_call_rtp_interfaces.headers INTERFACE) -target_link_libraries(webrtc_call_rtp_interfaces.headers INTERFACE webrtc_api_array_view.headers webrtc_api_fec_controller_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_crypto_options.headers webrtc_api_rtc_event_log_rtc_event_log.headers webrtc_api_transport_bitrate_settings.headers webrtc_api_units_timestamp.headers webrtc_common_video_frame_counts.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers ) +target_link_libraries(webrtc_call_rtp_interfaces.headers INTERFACE webrtc_api_array_view.headers webrtc_api_fec_controller_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_network_state_predictor_api.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_crypto_options.headers webrtc_api_rtc_event_log_rtc_event_log.headers webrtc_api_transport_bitrate_settings.headers webrtc_api_transport_network_control.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_api_units_timestamp.headers webrtc_common_video_frame_counts.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_modules_utility_utility.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_task_queue.headers ) target_include_directories(webrtc_call_rtp_interfaces.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # call:rtp_receiver @@ -1276,11 +1291,11 @@ add_library(webrtc_call_video_stream_api ${WEBRTC_ROOT}/call/video_receive_strea target_include_directories(webrtc_call_video_stream_api PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_call_video_stream_api PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_call_video_stream_api PRIVATE -fno-exceptions) -target_link_libraries(webrtc_call_video_stream_api PUBLIC absl::optional webrtc_api_frame_transformer_interface webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_transport_api webrtc_api_adaptation_resource_adaptation_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_transport_rtp_rtp_source webrtc_api_video_recordable_encoded_frame webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_video_stream_encoder webrtc_api_video_codecs_video_codecs_api webrtc_call_rtp_interfaces webrtc_common_video_common_video webrtc_common_video_frame_counts webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) +target_link_libraries(webrtc_call_video_stream_api PUBLIC absl::optional webrtc_api_frame_transformer_interface webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_transport_api webrtc_api_adaptation_resource_adaptation_api webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_video_recordable_encoded_frame webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_video_stream_encoder webrtc_api_video_codecs_video_codecs_api webrtc_call_receive_stream_interface webrtc_call_rtp_interfaces webrtc_common_video_common_video webrtc_common_video_frame_counts webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) # call:video_stream_api.headers add_library(webrtc_call_video_stream_api.headers INTERFACE) -target_link_libraries(webrtc_call_video_stream_api.headers INTERFACE webrtc_api_frame_transformer_interface.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_transport_api.headers webrtc_api_adaptation_resource_adaptation_api.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_transport_rtp_rtp_source.headers webrtc_api_video_recordable_encoded_frame.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_api_video_video_stream_encoder.headers webrtc_api_video_codecs_video_codecs_api.headers webrtc_call_rtp_interfaces.headers webrtc_common_video_common_video.headers webrtc_common_video_frame_counts.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers ) +target_link_libraries(webrtc_call_video_stream_api.headers INTERFACE webrtc_api_frame_transformer_interface.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_transport_api.headers webrtc_api_adaptation_resource_adaptation_api.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_video_recordable_encoded_frame.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_api_video_video_stream_encoder.headers webrtc_api_video_codecs_video_codecs_api.headers webrtc_call_receive_stream_interface.headers webrtc_call_rtp_interfaces.headers webrtc_common_video_common_video.headers webrtc_common_video_frame_counts.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base_approved.headers ) target_include_directories(webrtc_call_video_stream_api.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # common_audio @@ -1439,11 +1454,11 @@ add_library(webrtc_logging_ice_log ${WEBRTC_ROOT}/logging/rtc_event_log/events/r target_include_directories(webrtc_logging_ice_log PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_ice_log PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_ice_log PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_ice_log PUBLIC absl::memory webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_event_log_rtc_event_log webrtc_rtc_base_rtc_base_approved ) +target_link_libraries(webrtc_logging_ice_log PUBLIC absl::memory webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_timestamp webrtc_rtc_base_rtc_base_approved ) # logging:ice_log.headers add_library(webrtc_logging_ice_log.headers INTERFACE) -target_link_libraries(webrtc_logging_ice_log.headers INTERFACE webrtc_api_libjingle_logging_api.headers webrtc_api_libjingle_peerconnection_api.headers webrtc_api_rtc_event_log_rtc_event_log.headers webrtc_rtc_base_rtc_base_approved.headers ) +target_link_libraries(webrtc_logging_ice_log.headers INTERFACE webrtc_api_libjingle_logging_api.headers webrtc_api_libjingle_peerconnection_api.headers webrtc_api_rtc_event_log_rtc_event_log.headers webrtc_api_units_timestamp.headers webrtc_rtc_base_rtc_base_approved.headers ) target_include_directories(webrtc_logging_ice_log.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # logging:rtc_event_audio @@ -1451,41 +1466,41 @@ add_library(webrtc_logging_rtc_event_audio ${WEBRTC_ROOT}/logging/rtc_event_log/ target_include_directories(webrtc_logging_rtc_event_audio PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_audio PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_audio PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_audio PUBLIC absl::memory webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_logging_rtc_stream_config webrtc_modules_audio_coding_audio_network_adaptor_config webrtc_rtc_base_checks ) +target_link_libraries(webrtc_logging_rtc_event_audio PUBLIC absl::memory webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_timestamp webrtc_logging_rtc_stream_config webrtc_modules_audio_coding_audio_network_adaptor_config webrtc_rtc_base_checks ) # logging:rtc_event_bwe add_library(webrtc_logging_rtc_event_bwe ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_probe_result_success.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_route_change.cc) target_include_directories(webrtc_logging_rtc_event_bwe PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_bwe PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_bwe PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_bwe PUBLIC absl::memory absl::optional webrtc_api_network_state_predictor_api webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_data_rate ) +target_link_libraries(webrtc_logging_rtc_event_bwe PUBLIC absl::memory absl::optional webrtc_api_network_state_predictor_api webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_data_rate webrtc_api_units_timestamp ) # logging:rtc_event_frame_events add_library(webrtc_logging_rtc_event_frame_events ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_frame_decoded.cc) target_include_directories(webrtc_logging_rtc_event_frame_events PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_frame_events PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_frame_events PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_frame_events PUBLIC absl::memory absl::optional webrtc_api_rtc_event_log_rtc_event_log webrtc_api_video_video_frame webrtc_rtc_base_timeutils ) +target_link_libraries(webrtc_logging_rtc_event_frame_events PUBLIC absl::memory absl::optional webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_timestamp webrtc_api_video_video_frame webrtc_rtc_base_timeutils ) # logging:rtc_event_generic_packet_events add_library(webrtc_logging_rtc_event_generic_packet_events ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_generic_ack_received.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_generic_packet_received.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_generic_packet_sent.cc) target_include_directories(webrtc_logging_rtc_event_generic_packet_events PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_generic_packet_events PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_generic_packet_events PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_generic_packet_events PUBLIC absl::memory absl::optional webrtc_api_rtc_event_log_rtc_event_log webrtc_rtc_base_timeutils ) +target_link_libraries(webrtc_logging_rtc_event_generic_packet_events PUBLIC absl::memory absl::optional webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_timestamp webrtc_rtc_base_timeutils ) # //third_party/webrtc/files/stable/webrtc/logging:rtc_event_log2_proto -add_library(webrtc_logging_rtc_event_log2_proto_bridge) +add_library(webrtc_logging_rtc_event_log2_proto) protobuf_generate_with_plugin( - TARGET webrtc_logging_rtc_event_log2_proto_bridge + TARGET webrtc_logging_rtc_event_log2_proto PROTOS ${WEBRTC_ROOT}/logging/rtc_event_log/rtc_event_log2.proto HEADERFILEEXTENSION .pb.h APPEND_PATH PROTOPATH -I${WEBRTC_ROOT}/logging/rtc_event_log PROTOC_OUT_DIR ${CMAKE_CURRENT_BINARY_DIR}/logging/rtc_event_log) -target_include_directories(webrtc_logging_rtc_event_log2_proto_bridge PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/logging/rtc_event_log) -add_library(webrtc_logging_rtc_event_log2_proto_lib ALIAS webrtc_logging_rtc_event_log2_proto_bridge) -target_link_libraries(webrtc_logging_rtc_event_log2_proto_bridge PUBLIC libprotobuf) +target_include_directories(webrtc_logging_rtc_event_log2_proto PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/logging/rtc_event_log) +add_library(webrtc_logging_rtc_event_log2_proto_lib ALIAS webrtc_logging_rtc_event_log2_proto) +target_link_libraries(webrtc_logging_rtc_event_log2_proto PUBLIC libprotobuf) # logging:rtc_event_log_api add_library(webrtc_logging_rtc_event_log_api INTERFACE) @@ -1504,27 +1519,27 @@ add_library(webrtc_logging_rtc_event_log_impl_encoder ${WEBRTC_ROOT}/logging/rtc target_include_directories(webrtc_logging_rtc_event_log_impl_encoder PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_log_impl_encoder PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_log_impl_encoder PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_log_impl_encoder PUBLIC absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_network_state_predictor_api webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_transport_network_control webrtc_logging_ice_log webrtc_logging_rtc_event_audio webrtc_logging_rtc_event_bwe webrtc_logging_rtc_event_frame_events webrtc_logging_rtc_event_generic_packet_events webrtc_logging_rtc_event_log2_proto_bridge webrtc_logging_rtc_event_log_api webrtc_logging_rtc_event_log_proto_bridge webrtc_logging_rtc_event_pacing webrtc_logging_rtc_event_rtp_rtcp webrtc_logging_rtc_event_video webrtc_logging_rtc_stream_config webrtc_modules_audio_coding_audio_network_adaptor webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_rtc_base_approved ) +target_link_libraries(webrtc_logging_rtc_event_log_impl_encoder PUBLIC absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_network_state_predictor_api webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_transport_network_control webrtc_logging_ice_log webrtc_logging_rtc_event_audio webrtc_logging_rtc_event_bwe webrtc_logging_rtc_event_frame_events webrtc_logging_rtc_event_generic_packet_events webrtc_logging_rtc_event_log2_proto webrtc_logging_rtc_event_log_api webrtc_logging_rtc_event_log_proto webrtc_logging_rtc_event_pacing webrtc_logging_rtc_event_rtp_rtcp webrtc_logging_rtc_event_video webrtc_logging_rtc_stream_config webrtc_modules_audio_coding_audio_network_adaptor webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_rtc_base_approved ) # //third_party/webrtc/files/stable/webrtc/logging:rtc_event_log_proto -add_library(webrtc_logging_rtc_event_log_proto_bridge) +add_library(webrtc_logging_rtc_event_log_proto) protobuf_generate_with_plugin( - TARGET webrtc_logging_rtc_event_log_proto_bridge + TARGET webrtc_logging_rtc_event_log_proto PROTOS ${WEBRTC_ROOT}/logging/rtc_event_log/rtc_event_log.proto HEADERFILEEXTENSION .pb.h APPEND_PATH PROTOPATH -I${WEBRTC_ROOT}/logging/rtc_event_log PROTOC_OUT_DIR ${CMAKE_CURRENT_BINARY_DIR}/logging/rtc_event_log) -target_include_directories(webrtc_logging_rtc_event_log_proto_bridge PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/logging/rtc_event_log) -add_library(webrtc_logging_rtc_event_log_proto_lib ALIAS webrtc_logging_rtc_event_log_proto_bridge) -target_link_libraries(webrtc_logging_rtc_event_log_proto_bridge PUBLIC libprotobuf) +target_include_directories(webrtc_logging_rtc_event_log_proto PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/logging/rtc_event_log) +add_library(webrtc_logging_rtc_event_log_proto_lib ALIAS webrtc_logging_rtc_event_log_proto) +target_link_libraries(webrtc_logging_rtc_event_log_proto PUBLIC libprotobuf) # logging:rtc_event_pacing add_library(webrtc_logging_rtc_event_pacing ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_alr_state.cc) target_include_directories(webrtc_logging_rtc_event_pacing PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_pacing PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_pacing PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_pacing PUBLIC absl::memory webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log ) +target_link_libraries(webrtc_logging_rtc_event_pacing PUBLIC absl::memory webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_timestamp ) # logging:rtc_event_rtp_rtcp add_library(webrtc_logging_rtc_event_rtp_rtcp ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc ${WEBRTC_ROOT}/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc) @@ -1538,7 +1553,7 @@ add_library(webrtc_logging_rtc_event_video ${WEBRTC_ROOT}/logging/rtc_event_log/ target_include_directories(webrtc_logging_rtc_event_video PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_logging_rtc_event_video PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_logging_rtc_event_video PRIVATE -fno-exceptions) -target_link_libraries(webrtc_logging_rtc_event_video PUBLIC absl::memory webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_logging_rtc_stream_config webrtc_rtc_base_checks ) +target_link_libraries(webrtc_logging_rtc_event_video PUBLIC absl::memory webrtc_api_scoped_refptr webrtc_api_rtc_event_log_rtc_event_log webrtc_api_units_timestamp webrtc_logging_rtc_stream_config webrtc_rtc_base_checks ) # logging:rtc_stream_config add_library(webrtc_logging_rtc_stream_config ${WEBRTC_ROOT}/logging/rtc_event_log/rtc_stream_config.cc) @@ -1552,18 +1567,25 @@ add_library(webrtc_media_rtc_audio_video ${WEBRTC_ROOT}/media/engine/adm_helpers target_include_directories(webrtc_media_rtc_audio_video PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_media_rtc_audio_video PRIVATE WEBRTC_MAC WEBRTC_POSIX HAVE_WEBRTC_VIDEO WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_media_rtc_audio_video PRIVATE -fno-exceptions) -target_link_libraries(webrtc_media_rtc_audio_video PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_call_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_transport_api webrtc_api_audio_audio_frame_processor webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_data_rate webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_rtc_software_fallback_wrappers webrtc_api_video_codecs_video_codecs_api webrtc_call_call webrtc_call_call_interfaces webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_coding_ana_config_proto_bridge webrtc_modules_audio_device_audio_device webrtc_modules_audio_device_audio_device_impl webrtc_modules_audio_mixer_audio_mixer_impl webrtc_modules_audio_processing_api webrtc_modules_audio_processing_aec_dump_aec_dump webrtc_modules_audio_processing_aec_dump_aec_dump_impl webrtc_modules_audio_processing_agc_gain_control_interface webrtc_modules_video_capture_video_capture_internal_impl webrtc_modules_video_coding_video_coding webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_rtc_base_rtc_base webrtc_rtc_base_audio_format_to_string webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_experiments_min_video_bitrate_experiment webrtc_rtc_base_experiments_normalize_simulcast_size_experiment webrtc_rtc_base_experiments_rate_control_settings webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_media_rtc_audio_video PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_call_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_transport_api webrtc_api_audio_audio_frame_processor webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_data_rate webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_rtc_software_fallback_wrappers webrtc_api_video_codecs_video_codecs_api webrtc_call_call webrtc_call_call_interfaces webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_coding_ana_config_proto webrtc_modules_audio_device_audio_device webrtc_modules_audio_device_audio_device_impl webrtc_modules_audio_mixer_audio_mixer_impl webrtc_modules_audio_processing_api webrtc_modules_audio_processing_aec_dump_aec_dump webrtc_modules_audio_processing_aec_dump_aec_dump_impl webrtc_modules_audio_processing_agc_gain_control_interface webrtc_modules_video_capture_video_capture_internal_impl webrtc_modules_video_coding_video_coding webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_rtc_base_rtc_base webrtc_rtc_base_audio_format_to_string webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_experiments_min_video_bitrate_experiment webrtc_rtc_base_experiments_normalize_simulcast_size_experiment webrtc_rtc_base_experiments_rate_control_settings webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_metrics ) + +# media:rtc_data_dcsctp_transport +add_library(webrtc_media_rtc_data_dcsctp_transport ${WEBRTC_ROOT}/media/sctp/dcsctp_transport.cc) +target_include_directories(webrtc_media_rtc_data_dcsctp_transport PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_media_rtc_data_dcsctp_transport PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_media_rtc_data_dcsctp_transport PRIVATE -fno-exceptions) +target_link_libraries(webrtc_media_rtc_data_dcsctp_transport PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_net_dcsctp_public_factory webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_timer_task_queue_timeout webrtc_p2p_rtc_p2p webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_system_wrappers ) # media:rtc_data_sctp_transport_factory add_library(webrtc_media_rtc_data_sctp_transport_factory ${WEBRTC_ROOT}/media/sctp/sctp_transport_factory.cc) target_include_directories(webrtc_media_rtc_data_sctp_transport_factory PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) -target_compile_definitions(webrtc_media_rtc_data_sctp_transport_factory PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_HAVE_USRSCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_definitions(webrtc_media_rtc_data_sctp_transport_factory PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_DCSCTP WEBRTC_HAVE_SCTP WEBRTC_HAVE_USRSCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_media_rtc_data_sctp_transport_factory PRIVATE -fno-exceptions) -target_link_libraries(webrtc_media_rtc_data_sctp_transport_factory PUBLIC webrtc_api_transport_sctp_transport_factory_interface webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_data_usrsctp_transport webrtc_rtc_base_threading webrtc_rtc_base_system_unused ) +target_link_libraries(webrtc_media_rtc_data_sctp_transport_factory PUBLIC webrtc_api_transport_sctp_transport_factory_interface webrtc_media_rtc_data_dcsctp_transport webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_data_usrsctp_transport webrtc_rtc_base_threading webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_system_unused webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial ) # media:rtc_data_sctp_transport_internal add_library(webrtc_media_rtc_data_sctp_transport_internal INTERFACE) -target_link_libraries(webrtc_media_rtc_data_sctp_transport_internal INTERFACE webrtc_media_rtc_media_base webrtc_p2p_rtc_p2p webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading webrtc_rtc_base_third_party_sigslot_sigslot ) +target_link_libraries(webrtc_media_rtc_data_sctp_transport_internal INTERFACE webrtc_api_transport_datagram_transport_interface webrtc_media_rtc_media_base webrtc_p2p_rtc_p2p webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading webrtc_rtc_base_third_party_sigslot_sigslot ) target_include_directories(webrtc_media_rtc_data_sctp_transport_internal INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # media:rtc_data_usrsctp_transport @@ -1604,11 +1626,11 @@ add_library(webrtc_media_rtc_media_base ${WEBRTC_ROOT}/media/base/adapted_video_ target_include_directories(webrtc_media_rtc_media_base PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_media_rtc_media_base PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_media_rtc_media_base PRIVATE -fno-exceptions) -target_link_libraries(webrtc_media_rtc_media_base PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_frame_transformer_interface webrtc_api_media_stream_interface webrtc_api_rtc_error webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_audio_audio_frame_processor webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_transport_stun_types webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_media_config webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_sanitizer webrtc_rtc_base_socket webrtc_rtc_base_stringutils webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_rtc_export webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial ) +target_link_libraries(webrtc_media_rtc_media_base PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_frame_transformer_interface webrtc_api_media_stream_interface webrtc_api_rtc_error webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_audio_audio_frame_processor webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_transport_datagram_transport_interface webrtc_api_transport_stun_types webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_media_config webrtc_modules_async_audio_processing_async_audio_processing webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_sanitizer webrtc_rtc_base_socket webrtc_rtc_base_stringutils webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial ) # media:rtc_media_base.headers add_library(webrtc_media_rtc_media_base.headers INTERFACE) -target_link_libraries(webrtc_media_rtc_media_base.headers INTERFACE webrtc_api_array_view.headers webrtc_api_audio_options_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_media_stream_interface.headers webrtc_api_rtc_error.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_audio_audio_frame_processor.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_transport_stun_types.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_api_transport_rtp_rtp_source.headers webrtc_api_video_video_bitrate_allocation.headers webrtc_api_video_video_bitrate_allocator_factory.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_api_video_codecs_video_codecs_api.headers webrtc_call_call_interfaces.headers webrtc_call_video_stream_api.headers webrtc_common_video_common_video.headers webrtc_media_rtc_media_config.headers webrtc_modules_async_audio_processing_async_audio_processing.headers webrtc_modules_audio_processing_audio_processing_statistics.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_sanitizer.headers webrtc_rtc_base_socket.headers webrtc_rtc_base_stringutils.headers webrtc_rtc_base_synchronization_mutex.headers webrtc_rtc_base_system_file_wrapper.headers webrtc_rtc_base_system_rtc_export.headers webrtc_rtc_base_third_party_sigslot_sigslot.headers webrtc_system_wrappers_field_trial.headers ) +target_link_libraries(webrtc_media_rtc_media_base.headers INTERFACE webrtc_api_array_view.headers webrtc_api_audio_options_api.headers webrtc_api_frame_transformer_interface.headers webrtc_api_media_stream_interface.headers webrtc_api_rtc_error.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_audio_audio_frame_processor.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_crypto_frame_decryptor_interface.headers webrtc_api_crypto_frame_encryptor_interface.headers webrtc_api_crypto_options.headers webrtc_api_transport_datagram_transport_interface.headers webrtc_api_transport_stun_types.headers webrtc_api_transport_webrtc_key_value_config.headers webrtc_api_transport_rtp_rtp_source.headers webrtc_api_video_video_bitrate_allocation.headers webrtc_api_video_video_bitrate_allocator_factory.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_rtp_headers.headers webrtc_api_video_codecs_video_codecs_api.headers webrtc_call_call_interfaces.headers webrtc_call_video_stream_api.headers webrtc_common_video_common_video.headers webrtc_media_rtc_media_config.headers webrtc_modules_async_audio_processing_async_audio_processing.headers webrtc_modules_audio_processing_audio_processing_statistics.headers webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_rtc_base.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_sanitizer.headers webrtc_rtc_base_socket.headers webrtc_rtc_base_stringutils.headers webrtc_rtc_base_synchronization_mutex.headers webrtc_rtc_base_system_file_wrapper.headers webrtc_rtc_base_system_no_unique_address.headers webrtc_rtc_base_system_rtc_export.headers webrtc_rtc_base_task_utils_pending_task_safety_flag.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_rtc_base_third_party_sigslot_sigslot.headers webrtc_system_wrappers_field_trial.headers ) target_include_directories(webrtc_media_rtc_media_base.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # media:rtc_media_config @@ -1638,7 +1660,7 @@ target_link_libraries(webrtc_media_rtc_media_tests_utils PUBLIC gmock gtest absl target_include_directories(webrtc_media_rtc_media_unittests PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_media_rtc_media_unittests PRIVATE WEBRTC_MAC WEBRTC_POSIX HAVE_WEBRTC_VIDEO WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_OPUS_SUPPORT_120MS_PTIME=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_media_rtc_media_unittests PRIVATE -fno-exceptions) -target_link_libraries(webrtc_media_rtc_media_unittests PUBLIC absl::algorithm_container absl::memory absl::strings absl::optional usrsctp webrtc_api_create_simulcast_test_fixture_api webrtc_api_libjingle_peerconnection_api webrtc_api_mock_video_bitrate_allocator webrtc_api_mock_video_bitrate_allocator_factory webrtc_api_mock_video_codec_factory webrtc_api_mock_video_encoder webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_simulcast_test_fixture_api webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_test_video_function_video_factory webrtc_api_transport_field_trial_based_config webrtc_api_units_time_delta webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_api_video_codecs_video_codecs_api webrtc_audio_audio webrtc_call_call_interfaces webrtc_common_video_common_video webrtc_media_rtc_audio_video webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_data_usrsctp_transport webrtc_media_rtc_encoder_simulcast_proxy webrtc_media_rtc_internal_video_codecs webrtc_media_rtc_media webrtc_media_rtc_media_base webrtc_media_rtc_media_engine_defaults webrtc_media_rtc_media_tests_utils webrtc_media_rtc_sdp_video_format_utils webrtc_media_rtc_simulcast_encoder_adapter webrtc_modules_audio_device_mock_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_mocks webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_video_coding_simulcast_test_fixture_impl webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_webrtc_h264 webrtc_modules_video_coding_webrtc_vp8 webrtc_modules_video_coding_codecs_av1_libaom_av1_decoder webrtc_p2p_p2p_test_utils webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_event webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_experiments_min_video_bitrate_experiment webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_test_audio_codec_mocks webrtc_test_fake_video_codecs webrtc_test_field_trial webrtc_test_rtp_test_utils webrtc_test_test_main webrtc_test_test_support webrtc_test_video_test_common ) +target_link_libraries(webrtc_media_rtc_media_unittests PUBLIC absl::algorithm_container absl::memory absl::strings absl::optional usrsctp webrtc_api_create_simulcast_test_fixture_api webrtc_api_libjingle_peerconnection_api webrtc_api_mock_video_bitrate_allocator webrtc_api_mock_video_bitrate_allocator_factory webrtc_api_mock_video_codec_factory webrtc_api_mock_video_encoder webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_simulcast_test_fixture_api webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_test_video_function_video_factory webrtc_api_transport_field_trial_based_config webrtc_api_units_time_delta webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_api_video_codecs_video_codecs_api webrtc_audio_audio webrtc_call_call_interfaces webrtc_common_video_common_video webrtc_media_rtc_audio_video webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_data_usrsctp_transport webrtc_media_rtc_encoder_simulcast_proxy webrtc_media_rtc_internal_video_codecs webrtc_media_rtc_media webrtc_media_rtc_media_base webrtc_media_rtc_media_engine_defaults webrtc_media_rtc_media_tests_utils webrtc_media_rtc_sdp_video_format_utils webrtc_media_rtc_simulcast_encoder_adapter webrtc_modules_audio_device_mock_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_mocks webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_video_coding_simulcast_test_fixture_impl webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_webrtc_h264 webrtc_modules_video_coding_webrtc_vp8 webrtc_modules_video_coding_codecs_av1_libaom_av1_decoder webrtc_p2p_p2p_test_utils webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_event webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_experiments_min_video_bitrate_experiment webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial webrtc_test_audio_codec_mocks webrtc_test_fake_video_codecs webrtc_test_field_trial webrtc_test_rtp_test_utils webrtc_test_test_main webrtc_test_test_support webrtc_test_video_test_common ) # media:rtc_sdp_video_format_utils add_library(webrtc_media_rtc_sdp_video_format_utils ${WEBRTC_ROOT}/media/base/sdp_video_format_utils.cc) @@ -1667,31 +1689,31 @@ target_link_libraries(webrtc_modules_async_audio_processing_async_audio_processi target_include_directories(webrtc_modules_async_audio_processing_async_audio_processing.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # //third_party/webrtc/files/stable/webrtc/modules/audio_coding:ana_config_proto -add_library(webrtc_modules_audio_coding_ana_config_proto_bridge) +add_library(webrtc_modules_audio_coding_ana_config_proto) protobuf_generate_with_plugin( - TARGET webrtc_modules_audio_coding_ana_config_proto_bridge + TARGET webrtc_modules_audio_coding_ana_config_proto PROTOS ${WEBRTC_ROOT}/modules/audio_coding/audio_network_adaptor/config.proto HEADERFILEEXTENSION .pb.h APPEND_PATH PROTOPATH -I${WEBRTC_ROOT}/modules/audio_coding/audio_network_adaptor PROTOC_OUT_DIR ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_coding/audio_network_adaptor) -target_include_directories(webrtc_modules_audio_coding_ana_config_proto_bridge PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_coding/audio_network_adaptor) -add_library(webrtc_modules_audio_coding_ana_config_proto_lib ALIAS webrtc_modules_audio_coding_ana_config_proto_bridge) -target_link_libraries(webrtc_modules_audio_coding_ana_config_proto_bridge PUBLIC libprotobuf) +target_include_directories(webrtc_modules_audio_coding_ana_config_proto PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_coding/audio_network_adaptor) +add_library(webrtc_modules_audio_coding_ana_config_proto_lib ALIAS webrtc_modules_audio_coding_ana_config_proto) +target_link_libraries(webrtc_modules_audio_coding_ana_config_proto PUBLIC libprotobuf) # //third_party/webrtc/files/stable/webrtc/modules/audio_coding:ana_debug_dump_proto -add_library(webrtc_modules_audio_coding_ana_debug_dump_proto_bridge) +add_library(webrtc_modules_audio_coding_ana_debug_dump_proto) protobuf_generate_with_plugin( - TARGET webrtc_modules_audio_coding_ana_debug_dump_proto_bridge + TARGET webrtc_modules_audio_coding_ana_debug_dump_proto PROTOS ${WEBRTC_ROOT}/modules/audio_coding/audio_network_adaptor/debug_dump.proto HEADERFILEEXTENSION .pb.h APPEND_PATH PROTOPATH -I${WEBRTC_ROOT}/modules/audio_coding/audio_network_adaptor PROTOC_OUT_DIR ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_coding/audio_network_adaptor) -target_include_directories(webrtc_modules_audio_coding_ana_debug_dump_proto_bridge PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_coding/audio_network_adaptor) -add_library(webrtc_modules_audio_coding_ana_debug_dump_proto_lib ALIAS webrtc_modules_audio_coding_ana_debug_dump_proto_bridge) -target_link_libraries(webrtc_modules_audio_coding_ana_debug_dump_proto_bridge PUBLIC libprotobuf) -target_link_libraries(webrtc_modules_audio_coding_ana_debug_dump_proto_bridge PRIVATE webrtc_modules_audio_coding_ana_config_proto_lib) +target_include_directories(webrtc_modules_audio_coding_ana_debug_dump_proto PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_coding/audio_network_adaptor) +add_library(webrtc_modules_audio_coding_ana_debug_dump_proto_lib ALIAS webrtc_modules_audio_coding_ana_debug_dump_proto) +target_link_libraries(webrtc_modules_audio_coding_ana_debug_dump_proto PUBLIC libprotobuf) +target_link_libraries(webrtc_modules_audio_coding_ana_debug_dump_proto PRIVATE webrtc_modules_audio_coding_ana_config_proto_lib) # modules/audio_coding add_library(webrtc_modules_audio_coding_audio_coding ${WEBRTC_ROOT}/modules/audio_coding/acm2/acm_receiver.cc ${WEBRTC_ROOT}/modules/audio_coding/acm2/acm_remixing.cc ${WEBRTC_ROOT}/modules/audio_coding/acm2/acm_resampler.cc ${WEBRTC_ROOT}/modules/audio_coding/acm2/audio_coding_module.cc ${WEBRTC_ROOT}/modules/audio_coding/acm2/call_statistics.cc) @@ -1730,7 +1752,7 @@ add_library(webrtc_modules_audio_coding_audio_network_adaptor ${WEBRTC_ROOT}/mod target_include_directories(webrtc_modules_audio_coding_audio_network_adaptor PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_coding_audio_network_adaptor PRIVATE WEBRTC_MAC WEBRTC_POSIX GOOGLE_PROTOBUF_NO_RTTI GOOGLE_PROTOBUF_NO_STATIC_INITIALIZER WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_audio_coding_audio_network_adaptor PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_coding_audio_network_adaptor PUBLIC absl::algorithm_container absl::optional libprotobuf webrtc_api_audio_codecs_audio_codecs_api webrtc_api_rtc_event_log_rtc_event_log webrtc_common_audio_common_audio webrtc_logging_rtc_event_audio webrtc_modules_audio_coding_ana_config_proto_bridge webrtc_modules_audio_coding_ana_debug_dump_proto_bridge webrtc_modules_audio_coding_audio_network_adaptor_config webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_file_wrapper webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial ) +target_link_libraries(webrtc_modules_audio_coding_audio_network_adaptor PUBLIC absl::algorithm_container absl::optional webrtc_api_audio_codecs_audio_codecs_api webrtc_api_rtc_event_log_rtc_event_log webrtc_common_audio_common_audio webrtc_logging_rtc_event_audio webrtc_modules_audio_coding_ana_config_proto webrtc_modules_audio_coding_ana_debug_dump_proto webrtc_modules_audio_coding_audio_network_adaptor_config webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_protobuf_utils webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_system_file_wrapper webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial ) # modules/audio_coding:audio_network_adaptor_config add_library(webrtc_modules_audio_coding_audio_network_adaptor_config ${WEBRTC_ROOT}/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc) @@ -1881,7 +1903,7 @@ add_library(webrtc_modules_audio_coding_neteq_tools_minimal ${WEBRTC_ROOT}/modul target_include_directories(webrtc_modules_audio_coding_neteq_tools_minimal PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_coding_neteq_tools_minimal PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_CODEC_OPUS WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_OPUS_SUPPORT_120MS_PTIME=1 WEBRTC_CODEC_ISAC) target_compile_options(webrtc_modules_audio_coding_neteq_tools_minimal PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_coding_neteq_tools_minimal PUBLIC absl::optional webrtc_api_neteq_simulator_api webrtc_api_rtp_headers webrtc_api_audio_audio_frame_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_neteq_custom_neteq_factory webrtc_api_neteq_default_neteq_controller_factory webrtc_api_neteq_neteq_api webrtc_modules_audio_coding_default_neteq_factory webrtc_modules_audio_coding_neteq webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_system_wrappers_system_wrappers ) +target_link_libraries(webrtc_modules_audio_coding_neteq_tools_minimal PUBLIC absl::optional webrtc_api_array_view webrtc_api_neteq_simulator_api webrtc_api_rtp_headers webrtc_api_audio_audio_frame_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_neteq_custom_neteq_factory webrtc_api_neteq_default_neteq_controller_factory webrtc_api_neteq_neteq_api webrtc_modules_audio_coding_default_neteq_factory webrtc_modules_audio_coding_neteq webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_system_wrappers_system_wrappers ) # modules/audio_coding:pcm16b add_library(webrtc_modules_audio_coding_pcm16b ${WEBRTC_ROOT}/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc ${WEBRTC_ROOT}/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc ${WEBRTC_ROOT}/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc) @@ -1901,7 +1923,7 @@ add_library(webrtc_modules_audio_coding_red ${WEBRTC_ROOT}/modules/audio_coding/ target_include_directories(webrtc_modules_audio_coding_red PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_coding_red PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_audio_coding_red PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_coding_red PUBLIC absl::optional webrtc_api_array_view webrtc_api_audio_codecs_audio_codecs_api webrtc_api_units_time_delta webrtc_common_audio_common_audio webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) +target_link_libraries(webrtc_modules_audio_coding_red PUBLIC absl::optional webrtc_api_array_view webrtc_api_audio_codecs_audio_codecs_api webrtc_api_units_time_delta webrtc_common_audio_common_audio webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_system_wrappers_field_trial ) # modules/audio_coding:webrtc_cng add_library(webrtc_modules_audio_coding_webrtc_cng ${WEBRTC_ROOT}/modules/audio_coding/codecs/cng/webrtc_cng.cc) @@ -1922,7 +1944,7 @@ add_library(webrtc_modules_audio_coding_webrtc_opus ${WEBRTC_ROOT}/modules/audio target_include_directories(webrtc_modules_audio_coding_webrtc_opus PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_coding_webrtc_opus PRIVATE WEBRTC_MAC WEBRTC_POSIX GOOGLE_PROTOBUF_NO_RTTI GOOGLE_PROTOBUF_NO_STATIC_INITIALIZER WEBRTC_ABSL_MUTEX WEBRTC_CODEC_OPUS WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_OPUS_SUPPORT_120MS_PTIME=1 WEBRTC_CODEC_ISAC) target_compile_options(webrtc_modules_audio_coding_webrtc_opus PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_coding_webrtc_opus PUBLIC absl::strings absl::optional libprotobuf webrtc_api_array_view webrtc_api_audio_codecs_audio_codecs_api webrtc_api_audio_codecs_opus_audio_encoder_opus_config webrtc_common_audio_common_audio webrtc_modules_audio_coding_audio_coding_opus_common webrtc_modules_audio_coding_audio_network_adaptor webrtc_modules_audio_coding_webrtc_opus_wrapper webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_safe_minmax webrtc_system_wrappers_field_trial ) +target_link_libraries(webrtc_modules_audio_coding_webrtc_opus PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_codecs_audio_codecs_api webrtc_api_audio_codecs_opus_audio_encoder_opus_config webrtc_common_audio_common_audio webrtc_modules_audio_coding_audio_coding_opus_common webrtc_modules_audio_coding_audio_network_adaptor webrtc_modules_audio_coding_webrtc_opus_wrapper webrtc_rtc_base_checks webrtc_rtc_base_protobuf_utils webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_safe_minmax webrtc_system_wrappers_field_trial ) # modules/audio_coding:webrtc_opus_wrapper add_library(webrtc_modules_audio_coding_webrtc_opus_wrapper ${WEBRTC_ROOT}/modules/audio_coding/codecs/opus/opus_interface.cc) @@ -2024,7 +2046,7 @@ add_library(webrtc_modules_audio_mixer_audio_mixer_impl ${WEBRTC_ROOT}/modules/a target_include_directories(webrtc_modules_audio_mixer_audio_mixer_impl PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_mixer_audio_mixer_impl PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_audio_mixer_audio_mixer_impl PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_mixer_audio_mixer_impl PUBLIC webrtc_api_array_view webrtc_api_scoped_refptr webrtc_api_audio_audio_frame_api webrtc_api_audio_audio_mixer_api webrtc_audio_utility_audio_frame_operations webrtc_common_audio_common_audio webrtc_modules_audio_mixer_audio_frame_manipulator webrtc_modules_audio_processing_api webrtc_modules_audio_processing_apm_logging webrtc_modules_audio_processing_audio_frame_view webrtc_modules_audio_processing_agc2_fixed_digital webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_synchronization_mutex webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_modules_audio_mixer_audio_mixer_impl PUBLIC webrtc_api_array_view webrtc_api_rtp_packet_info webrtc_api_scoped_refptr webrtc_api_audio_audio_frame_api webrtc_api_audio_audio_mixer_api webrtc_audio_utility_audio_frame_operations webrtc_common_audio_common_audio webrtc_modules_audio_mixer_audio_frame_manipulator webrtc_modules_audio_processing_api webrtc_modules_audio_processing_apm_logging webrtc_modules_audio_processing_audio_frame_view webrtc_modules_audio_processing_agc2_fixed_digital webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_safe_conversions webrtc_rtc_base_synchronization_mutex webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_metrics ) # modules/audio_processing/aec3:adaptive_fir_filter add_library(webrtc_modules_audio_processing_aec3_adaptive_fir_filter INTERFACE) @@ -2143,11 +2165,11 @@ add_library(webrtc_modules_audio_processing_aec_dump_aec_dump_impl ${WEBRTC_ROOT target_include_directories(webrtc_modules_audio_processing_aec_dump_aec_dump_impl PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_processing_aec_dump_aec_dump_impl PRIVATE WEBRTC_MAC WEBRTC_POSIX GOOGLE_PROTOBUF_NO_RTTI GOOGLE_PROTOBUF_NO_STATIC_INITIALIZER WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_audio_processing_aec_dump_aec_dump_impl PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_processing_aec_dump_aec_dump_impl PUBLIC libprotobuf webrtc_api_audio_audio_frame_api webrtc_api_task_queue_task_queue webrtc_modules_audio_processing_aec_dump_interface webrtc_modules_audio_processing_audioproc_debug_proto_bridge webrtc_modules_audio_processing_aec_dump_aec_dump webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_system_file_wrapper webrtc_system_wrappers_system_wrappers ) +target_link_libraries(webrtc_modules_audio_processing_aec_dump_aec_dump_impl PUBLIC webrtc_api_audio_audio_frame_api webrtc_api_task_queue_task_queue webrtc_modules_audio_processing_aec_dump_interface webrtc_modules_audio_processing_audioproc_debug_proto webrtc_modules_audio_processing_aec_dump_aec_dump webrtc_rtc_base_checks webrtc_rtc_base_ignore_wundef webrtc_rtc_base_protobuf_utils webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_system_file_wrapper webrtc_system_wrappers_system_wrappers ) # modules/audio_processing/aec_dump:aec_dump_impl.headers add_library(webrtc_modules_audio_processing_aec_dump_aec_dump_impl.headers INTERFACE) -target_link_libraries(webrtc_modules_audio_processing_aec_dump_aec_dump_impl.headers INTERFACE webrtc_api_audio_audio_frame_api.headers webrtc_api_task_queue_task_queue.headers webrtc_modules_audio_processing_aec_dump_interface.headers webrtc_modules_audio_processing_aec_dump_aec_dump.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_ignore_wundef.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_system_file_wrapper.headers webrtc_system_wrappers_system_wrappers.headers ) +target_link_libraries(webrtc_modules_audio_processing_aec_dump_aec_dump_impl.headers INTERFACE webrtc_api_audio_audio_frame_api.headers webrtc_api_task_queue_task_queue.headers webrtc_modules_audio_processing_aec_dump_interface.headers webrtc_modules_audio_processing_aec_dump_aec_dump.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_ignore_wundef.headers webrtc_rtc_base_protobuf_utils.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_system_file_wrapper.headers webrtc_system_wrappers_system_wrappers.headers ) target_include_directories(webrtc_modules_audio_processing_aec_dump_aec_dump_impl.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # modules/audio_processing:aec_dump_interface @@ -2405,13 +2427,49 @@ add_library(webrtc_modules_audio_processing_agc_agc ${WEBRTC_ROOT}/modules/audio target_include_directories(webrtc_modules_audio_processing_agc_agc PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_audio_processing_agc_agc PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_audio_processing_agc_agc PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_audio_processing_agc_agc PUBLIC absl::optional webrtc_common_audio_common_audio webrtc_common_audio_common_audio_c webrtc_modules_audio_processing_apm_logging webrtc_modules_audio_processing_audio_buffer webrtc_modules_audio_processing_agc_gain_control_interface webrtc_modules_audio_processing_agc_gain_map webrtc_modules_audio_processing_agc_level_estimation webrtc_modules_audio_processing_vad_vad webrtc_rtc_base_checks webrtc_rtc_base_gtest_prod webrtc_rtc_base_logging webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_safe_minmax webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_modules_audio_processing_agc_agc PUBLIC absl::optional webrtc_common_audio_common_audio webrtc_common_audio_common_audio_c webrtc_modules_audio_processing_apm_logging webrtc_modules_audio_processing_audio_buffer webrtc_modules_audio_processing_audio_frame_view webrtc_modules_audio_processing_agc_clipping_predictor webrtc_modules_audio_processing_agc_clipping_predictor_evaluator webrtc_modules_audio_processing_agc_gain_control_interface webrtc_modules_audio_processing_agc_gain_map webrtc_modules_audio_processing_agc_level_estimation webrtc_modules_audio_processing_vad_vad webrtc_rtc_base_checks webrtc_rtc_base_gtest_prod webrtc_rtc_base_logging webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_safe_minmax webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) # modules/audio_processing/agc:agc.headers add_library(webrtc_modules_audio_processing_agc_agc.headers INTERFACE) -target_link_libraries(webrtc_modules_audio_processing_agc_agc.headers INTERFACE webrtc_common_audio_common_audio.headers webrtc_common_audio_common_audio_c.headers webrtc_modules_audio_processing_apm_logging.headers webrtc_modules_audio_processing_audio_buffer.headers webrtc_modules_audio_processing_agc_gain_control_interface.headers webrtc_modules_audio_processing_agc_gain_map.headers webrtc_modules_audio_processing_agc_level_estimation.headers webrtc_modules_audio_processing_vad_vad.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_gtest_prod.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_safe_minmax.headers webrtc_system_wrappers_field_trial.headers webrtc_system_wrappers_metrics.headers ) +target_link_libraries(webrtc_modules_audio_processing_agc_agc.headers INTERFACE webrtc_common_audio_common_audio.headers webrtc_common_audio_common_audio_c.headers webrtc_modules_audio_processing_apm_logging.headers webrtc_modules_audio_processing_audio_buffer.headers webrtc_modules_audio_processing_audio_frame_view.headers webrtc_modules_audio_processing_agc_clipping_predictor.headers webrtc_modules_audio_processing_agc_clipping_predictor_evaluator.headers webrtc_modules_audio_processing_agc_gain_control_interface.headers webrtc_modules_audio_processing_agc_gain_map.headers webrtc_modules_audio_processing_agc_level_estimation.headers webrtc_modules_audio_processing_vad_vad.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_gtest_prod.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_safe_minmax.headers webrtc_system_wrappers_field_trial.headers webrtc_system_wrappers_metrics.headers ) target_include_directories(webrtc_modules_audio_processing_agc_agc.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +# modules/audio_processing/agc:clipping_predictor +add_library(webrtc_modules_audio_processing_agc_clipping_predictor ${WEBRTC_ROOT}/modules/audio_processing/agc/clipping_predictor.cc) +target_include_directories(webrtc_modules_audio_processing_agc_clipping_predictor PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_modules_audio_processing_agc_clipping_predictor PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_modules_audio_processing_agc_clipping_predictor PRIVATE -fno-exceptions) +target_link_libraries(webrtc_modules_audio_processing_agc_clipping_predictor PUBLIC absl::optional webrtc_common_audio_common_audio webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_frame_view webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer webrtc_modules_audio_processing_agc_gain_map webrtc_rtc_base_checks webrtc_rtc_base_logging webrtc_rtc_base_safe_minmax ) + +# modules/audio_processing/agc:clipping_predictor.headers +add_library(webrtc_modules_audio_processing_agc_clipping_predictor.headers INTERFACE) +target_link_libraries(webrtc_modules_audio_processing_agc_clipping_predictor.headers INTERFACE webrtc_common_audio_common_audio.headers webrtc_modules_audio_processing_api.headers webrtc_modules_audio_processing_audio_frame_view.headers webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer.headers webrtc_modules_audio_processing_agc_gain_map.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_safe_minmax.headers ) +target_include_directories(webrtc_modules_audio_processing_agc_clipping_predictor.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# modules/audio_processing/agc:clipping_predictor_evaluator +add_library(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator ${WEBRTC_ROOT}/modules/audio_processing/agc/clipping_predictor_evaluator.cc) +target_include_directories(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator PRIVATE -fno-exceptions) +target_link_libraries(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator PUBLIC absl::optional webrtc_rtc_base_checks webrtc_rtc_base_logging ) + +# modules/audio_processing/agc:clipping_predictor_evaluator.headers +add_library(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator.headers INTERFACE) +target_link_libraries(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator.headers INTERFACE webrtc_rtc_base_checks.headers webrtc_rtc_base_logging.headers ) +target_include_directories(webrtc_modules_audio_processing_agc_clipping_predictor_evaluator.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# modules/audio_processing/agc:clipping_predictor_level_buffer +add_library(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer ${WEBRTC_ROOT}/modules/audio_processing/agc/clipping_predictor_level_buffer.cc) +target_include_directories(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer PRIVATE -fno-exceptions) +target_link_libraries(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer PUBLIC absl::optional webrtc_rtc_base_checks webrtc_rtc_base_logging webrtc_rtc_base_rtc_base_approved ) + +# modules/audio_processing/agc:clipping_predictor_level_buffer.headers +add_library(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer.headers INTERFACE) +target_link_libraries(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer.headers INTERFACE webrtc_rtc_base_checks.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_rtc_base_approved.headers ) +target_include_directories(webrtc_modules_audio_processing_agc_clipping_predictor_level_buffer.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + # modules/audio_processing/agc:gain_control_interface add_library(webrtc_modules_audio_processing_agc_gain_control_interface INTERFACE) target_include_directories(webrtc_modules_audio_processing_agc_gain_control_interface INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) @@ -2535,17 +2593,17 @@ target_link_libraries(webrtc_modules_audio_processing_audio_processing_statistic target_include_directories(webrtc_modules_audio_processing_audio_processing_statistics.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # //third_party/webrtc/files/stable/webrtc/modules/audio_processing:audioproc_debug_proto -add_library(webrtc_modules_audio_processing_audioproc_debug_proto_bridge) +add_library(webrtc_modules_audio_processing_audioproc_debug_proto) protobuf_generate_with_plugin( - TARGET webrtc_modules_audio_processing_audioproc_debug_proto_bridge + TARGET webrtc_modules_audio_processing_audioproc_debug_proto PROTOS ${WEBRTC_ROOT}/modules/audio_processing/debug.proto HEADERFILEEXTENSION .pb.h APPEND_PATH PROTOPATH -I${WEBRTC_ROOT}/modules/audio_processing PROTOC_OUT_DIR ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_processing) -target_include_directories(webrtc_modules_audio_processing_audioproc_debug_proto_bridge PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_processing) -add_library(webrtc_modules_audio_processing_audioproc_debug_proto_lib ALIAS webrtc_modules_audio_processing_audioproc_debug_proto_bridge) -target_link_libraries(webrtc_modules_audio_processing_audioproc_debug_proto_bridge PUBLIC libprotobuf) +target_include_directories(webrtc_modules_audio_processing_audioproc_debug_proto PUBLIC ${CMAKE_CURRENT_BINARY_DIR}/modules/audio_processing) +add_library(webrtc_modules_audio_processing_audioproc_debug_proto_lib ALIAS webrtc_modules_audio_processing_audioproc_debug_proto) +target_link_libraries(webrtc_modules_audio_processing_audioproc_debug_proto PUBLIC libprotobuf) # modules/audio_processing:audioproc_test_utils add_library(webrtc_modules_audio_processing_audioproc_test_utils ${WEBRTC_ROOT}/modules/audio_processing/test/audio_buffer_tools.cc ${WEBRTC_ROOT}/modules/audio_processing/test/audio_processing_builder_for_testing.cc ${WEBRTC_ROOT}/modules/audio_processing/test/bitexactness_tools.cc ${WEBRTC_ROOT}/modules/audio_processing/test/performance_timer.cc ${WEBRTC_ROOT}/modules/audio_processing/test/simulator_buffers.cc ${WEBRTC_ROOT}/modules/audio_processing/test/test_utils.cc) @@ -2712,11 +2770,11 @@ target_link_libraries(webrtc_modules_audio_processing_voice_detection.headers IN target_include_directories(webrtc_modules_audio_processing_voice_detection.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # modules/congestion_controller -add_library(webrtc_modules_congestion_controller_congestion_controller ${WEBRTC_ROOT}/modules/congestion_controller/receive_side_congestion_controller.cc) +add_library(webrtc_modules_congestion_controller_congestion_controller ${WEBRTC_ROOT}/modules/congestion_controller/receive_side_congestion_controller.cc ${WEBRTC_ROOT}/modules/congestion_controller/remb_throttler.cc) target_include_directories(webrtc_modules_congestion_controller_congestion_controller PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_congestion_controller_congestion_controller PRIVATE WEBRTC_MAC WEBRTC_POSIX BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0 WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_congestion_controller_congestion_controller PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_congestion_controller_congestion_controller PUBLIC webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_modules_module_api webrtc_modules_pacing_pacing webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_rtc_base webrtc_rtc_base_synchronization_mutex ) +target_link_libraries(webrtc_modules_congestion_controller_congestion_controller PUBLIC webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_api_units_data_rate webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_modules_module_api webrtc_modules_pacing_pacing webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_rtc_base webrtc_rtc_base_synchronization_mutex ) # modules/congestion_controller/goog_cc:alr_detector add_library(webrtc_modules_congestion_controller_goog_cc_alr_detector ${WEBRTC_ROOT}/modules/congestion_controller/goog_cc/alr_detector.cc) @@ -2835,11 +2893,11 @@ target_compile_options(webrtc_modules_pacing_pacing PRIVATE -fno-exceptions) target_link_libraries(webrtc_modules_pacing_pacing PUBLIC absl::memory absl::strings absl::optional webrtc_api_function_view webrtc_api_sequence_checker webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_api_transport_webrtc_key_value_config webrtc_api_units_data_rate webrtc_api_units_data_size webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_logging_rtc_event_bwe webrtc_logging_rtc_event_pacing webrtc_modules_module_api webrtc_modules_pacing_interval_budget webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_to_queued_task webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_metrics ) # modules/remote_bitrate_estimator -add_library(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/aimd_rate_control.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/bwe_defines.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/inter_arrival.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/overuse_detector.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/overuse_estimator.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/remote_estimator_proxy.cc) +add_library(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/aimd_rate_control.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/bwe_defines.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/inter_arrival.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/overuse_detector.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/overuse_estimator.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/packet_arrival_map.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc ${WEBRTC_ROOT}/modules/remote_bitrate_estimator/remote_estimator_proxy.cc) target_include_directories(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator PRIVATE WEBRTC_MAC WEBRTC_POSIX BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0 WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator PUBLIC absl::strings absl::optional webrtc_api_network_state_predictor_api webrtc_api_rtp_headers webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_api_transport_webrtc_key_value_config webrtc_api_units_data_rate webrtc_api_units_timestamp webrtc_modules_module_api webrtc_modules_module_api_public webrtc_modules_congestion_controller_goog_cc_link_capacity_estimator webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_safe_minmax webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_synchronization_mutex webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator PUBLIC absl::strings absl::optional webrtc_api_network_state_predictor_api webrtc_api_rtp_headers webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_api_transport_webrtc_key_value_config webrtc_api_units_data_rate webrtc_api_units_data_size webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_modules_module_api webrtc_modules_module_api_public webrtc_modules_congestion_controller_goog_cc_link_capacity_estimator webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_safe_minmax webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_synchronization_mutex webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) # modules/rtp_rtcp:mock_rtp_rtcp add_library(webrtc_modules_rtp_rtcp_mock_rtp_rtcp INTERFACE) @@ -2847,7 +2905,7 @@ target_link_libraries(webrtc_modules_rtp_rtcp_mock_rtp_rtcp INTERFACE absl::opti target_include_directories(webrtc_modules_rtp_rtcp_mock_rtp_rtcp INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # modules/rtp_rtcp -add_library(webrtc_modules_rtp_rtcp_rtp_rtcp ${WEBRTC_ROOT}/modules/rtp_rtcp/source/absolute_capture_time_receiver.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/absolute_capture_time_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/active_decode_targets_helper.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/dtmf_queue.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/fec_private_tables_bursty.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/fec_private_tables_random.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/flexfec_receiver.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/flexfec_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/forward_error_correction.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/forward_error_correction_internal.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/packet_loss_stats.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/packet_sequencer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/receive_statistics_impl.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtcp_nack_stats.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtcp_receiver.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtcp_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_descriptor_authentication.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_h264.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_video_generic.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_vp8.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_vp9.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_header_extension_size.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_packet_history.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_packetizer_av1.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_audio.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_egress.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_video.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sequence_number_map.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_utility.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/source_tracker.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/time_util.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/tmmbr_help.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/ulpfec_generator.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc) +add_library(webrtc_modules_rtp_rtcp_rtp_rtcp ${WEBRTC_ROOT}/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/absolute_capture_time_receiver.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/absolute_capture_time_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/active_decode_targets_helper.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/capture_clock_offset_updater.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/dtmf_queue.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/fec_private_tables_bursty.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/fec_private_tables_random.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/flexfec_receiver.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/flexfec_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/forward_error_correction.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/forward_error_correction_internal.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/packet_loss_stats.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/packet_sequencer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/receive_statistics_impl.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtcp_nack_stats.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtcp_receiver.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtcp_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_descriptor_authentication.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_h264.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_video_generic.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_vp8.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_format_vp9.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_header_extension_size.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_packet_history.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_packetizer_av1.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_audio.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_egress.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_video.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_sequence_number_map.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/rtp_utility.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/source_tracker.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/time_util.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/tmmbr_help.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/ulpfec_generator.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc ${WEBRTC_ROOT}/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc) target_include_directories(webrtc_modules_rtp_rtcp_rtp_rtcp PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_rtp_rtcp_rtp_rtcp PRIVATE WEBRTC_MAC WEBRTC_POSIX BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0 WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_rtp_rtcp_rtp_rtcp PRIVATE -fno-exceptions) @@ -2858,11 +2916,11 @@ add_library(webrtc_modules_rtp_rtcp_rtp_rtcp_format ${WEBRTC_ROOT}/modules/rtp_r target_include_directories(webrtc_modules_rtp_rtcp_rtp_rtcp_format PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_rtp_rtcp_rtp_rtcp_format PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_rtp_rtcp_rtp_rtcp_format PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_rtp_rtcp_rtp_rtcp_format PUBLIC absl::algorithm_container absl::strings absl::optional absl::variant webrtc_api_array_view webrtc_api_function_view webrtc_api_refcountedbase webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_audio_codecs_audio_codecs_api webrtc_api_transport_network_control webrtc_api_transport_rtp_dependency_descriptor webrtc_api_units_time_delta webrtc_api_video_video_frame webrtc_api_video_video_layers_allocation webrtc_api_video_video_rtp_headers webrtc_common_video_common_video webrtc_modules_module_api_public webrtc_modules_video_coding_codec_globals_headers webrtc_rtc_base_checks webrtc_rtc_base_divide_round webrtc_rtc_base_rtc_base_approved webrtc_system_wrappers_system_wrappers ) +target_link_libraries(webrtc_modules_rtp_rtcp_rtp_rtcp_format PUBLIC absl::algorithm_container absl::strings absl::optional absl::variant webrtc_api_array_view webrtc_api_function_view webrtc_api_refcountedbase webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_audio_codecs_audio_codecs_api webrtc_api_transport_network_control webrtc_api_transport_rtp_dependency_descriptor webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_api_video_video_frame webrtc_api_video_video_layers_allocation webrtc_api_video_video_rtp_headers webrtc_common_video_common_video webrtc_modules_module_api_public webrtc_modules_video_coding_codec_globals_headers webrtc_rtc_base_checks webrtc_rtc_base_divide_round webrtc_rtc_base_rtc_base_approved webrtc_system_wrappers_system_wrappers ) # modules/rtp_rtcp:rtp_rtcp_format.headers add_library(webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers INTERFACE) -target_link_libraries(webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers INTERFACE webrtc_api_array_view.headers webrtc_api_function_view.headers webrtc_api_refcountedbase.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_transport_network_control.headers webrtc_api_transport_rtp_dependency_descriptor.headers webrtc_api_units_time_delta.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_layers_allocation.headers webrtc_api_video_video_rtp_headers.headers webrtc_common_video_common_video.headers webrtc_modules_module_api_public.headers webrtc_modules_video_coding_codec_globals_headers.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_divide_round.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_system_wrappers_system_wrappers.headers ) +target_link_libraries(webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers INTERFACE webrtc_api_array_view.headers webrtc_api_function_view.headers webrtc_api_refcountedbase.headers webrtc_api_rtp_headers.headers webrtc_api_rtp_parameters.headers webrtc_api_scoped_refptr.headers webrtc_api_audio_codecs_audio_codecs_api.headers webrtc_api_transport_network_control.headers webrtc_api_transport_rtp_dependency_descriptor.headers webrtc_api_units_time_delta.headers webrtc_api_units_timestamp.headers webrtc_api_video_video_frame.headers webrtc_api_video_video_layers_allocation.headers webrtc_api_video_video_rtp_headers.headers webrtc_common_video_common_video.headers webrtc_modules_module_api_public.headers webrtc_modules_video_coding_codec_globals_headers.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_divide_round.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_system_wrappers_system_wrappers.headers ) target_include_directories(webrtc_modules_rtp_rtcp_rtp_rtcp_format.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # modules/rtp_rtcp:rtp_video_header @@ -3093,7 +3151,7 @@ add_library(webrtc_modules_video_coding_webrtc_vp9 ${WEBRTC_ROOT}/modules/video_ target_include_directories(webrtc_modules_video_coding_webrtc_vp9 PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_modules_video_coding_webrtc_vp9 PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_modules_video_coding_webrtc_vp9 PRIVATE -fno-exceptions) -target_link_libraries(webrtc_modules_video_coding_webrtc_vp9 PUBLIC absl::algorithm_container absl::memory absl::strings webrtc-yuv libvpx webrtc_api_fec_controller_api webrtc_api_scoped_refptr webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_video_video_frame webrtc_api_video_video_frame_i010 webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_modules_video_coding_webrtc_libvpx_interface webrtc_modules_video_coding_webrtc_vp9_helpers webrtc_modules_video_coding_svc_scalability_structures webrtc_modules_video_coding_svc_scalable_video_controller webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_experiments_encoder_info_settings webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_experiments_rate_control_settings webrtc_rtc_base_synchronization_mutex webrtc_system_wrappers_field_trial ) +target_link_libraries(webrtc_modules_video_coding_webrtc_vp9 PUBLIC absl::algorithm_container absl::memory absl::strings webrtc-yuv libvpx webrtc_api_fec_controller_api webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_video_video_frame webrtc_api_video_video_frame_i010 webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_modules_video_coding_webrtc_libvpx_interface webrtc_modules_video_coding_webrtc_vp9_helpers webrtc_modules_video_coding_svc_scalability_structures webrtc_modules_video_coding_svc_scalable_video_controller webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_experiments_encoder_info_settings webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_experiments_rate_control_settings webrtc_rtc_base_synchronization_mutex webrtc_system_wrappers_field_trial ) # modules/video_coding:webrtc_vp9_helpers add_library(webrtc_modules_video_coding_webrtc_vp9_helpers ${WEBRTC_ROOT}/modules/video_coding/codecs/vp9/svc_config.cc) @@ -3120,6 +3178,215 @@ target_compile_definitions(webrtc_modules_video_processing_video_processing_sse2 target_compile_options(webrtc_modules_video_processing_video_processing_sse2 PRIVATE -fno-exceptions -msse2) target_link_libraries(webrtc_modules_video_processing_video_processing_sse2 PUBLIC webrtc_modules_video_processing_denoiser_filter webrtc_rtc_base_rtc_base_approved webrtc_system_wrappers_system_wrappers ) +# net/dcsctp/common:internal_types +add_library(webrtc_net_dcsctp_common_internal_types INTERFACE) +target_link_libraries(webrtc_net_dcsctp_common_internal_types INTERFACE webrtc_net_dcsctp_public_strong_alias webrtc_net_dcsctp_public_types ) +target_include_directories(webrtc_net_dcsctp_common_internal_types INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/common:math +add_library(webrtc_net_dcsctp_common_math INTERFACE) +target_include_directories(webrtc_net_dcsctp_common_math INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/common:pair_hash +add_library(webrtc_net_dcsctp_common_pair_hash INTERFACE) +target_include_directories(webrtc_net_dcsctp_common_pair_hash INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/common:sequence_numbers +add_library(webrtc_net_dcsctp_common_sequence_numbers INTERFACE) +target_link_libraries(webrtc_net_dcsctp_common_sequence_numbers INTERFACE webrtc_net_dcsctp_common_internal_types ) +target_include_directories(webrtc_net_dcsctp_common_sequence_numbers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/common:str_join +add_library(webrtc_net_dcsctp_common_str_join INTERFACE) +target_link_libraries(webrtc_net_dcsctp_common_str_join INTERFACE absl::strings webrtc_rtc_base_stringutils ) +target_include_directories(webrtc_net_dcsctp_common_str_join INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/packet:bounded_io +add_library(webrtc_net_dcsctp_packet_bounded_io INTERFACE) +target_link_libraries(webrtc_net_dcsctp_packet_bounded_io INTERFACE webrtc_api_array_view webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) +target_include_directories(webrtc_net_dcsctp_packet_bounded_io INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/packet:chunk +add_library(webrtc_net_dcsctp_packet_chunk ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/abort_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/cookie_ack_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/cookie_echo_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/data_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/error_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/forward_tsn_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/heartbeat_ack_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/heartbeat_request_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/idata_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/iforward_tsn_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/init_ack_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/init_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/reconfig_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/sack_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/shutdown_ack_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/shutdown_chunk.cc ${WEBRTC_ROOT}/net/dcsctp/packet/chunk/shutdown_complete_chunk.cc) +target_include_directories(webrtc_net_dcsctp_packet_chunk PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_chunk PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_chunk PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_chunk PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_math webrtc_net_dcsctp_common_str_join webrtc_net_dcsctp_packet_bounded_io webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_packet_error_cause webrtc_net_dcsctp_packet_parameter webrtc_net_dcsctp_packet_tlv_trait webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/packet:chunk_validators +add_library(webrtc_net_dcsctp_packet_chunk_validators ${WEBRTC_ROOT}/net/dcsctp/packet/chunk_validators.cc) +target_include_directories(webrtc_net_dcsctp_packet_chunk_validators PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_chunk_validators PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_chunk_validators PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_chunk_validators PUBLIC webrtc_net_dcsctp_packet_chunk webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/packet:crc32c +add_library(webrtc_net_dcsctp_packet_crc32c ${WEBRTC_ROOT}/net/dcsctp/packet/crc32c.cc) +target_include_directories(webrtc_net_dcsctp_packet_crc32c PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_crc32c PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_crc32c PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_crc32c PUBLIC crc32c webrtc_api_array_view webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/packet:data +add_library(webrtc_net_dcsctp_packet_data INTERFACE) +target_link_libraries(webrtc_net_dcsctp_packet_data INTERFACE webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) +target_include_directories(webrtc_net_dcsctp_packet_data INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/packet:error_cause +add_library(webrtc_net_dcsctp_packet_error_cause ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/cookie_received_while_shutting_down_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/error_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/invalid_mandatory_parameter_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/invalid_stream_identifier_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/missing_mandatory_parameter_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/no_user_data_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/out_of_resource_error_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/protocol_violation_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/restart_of_an_association_with_new_address_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/stale_cookie_error_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/unrecognized_chunk_type_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/unrecognized_parameter_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/unresolvable_address_cause.cc ${WEBRTC_ROOT}/net/dcsctp/packet/error_cause/user_initiated_abort_cause.cc) +target_include_directories(webrtc_net_dcsctp_packet_error_cause PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_error_cause PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_error_cause PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_error_cause PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_common_math webrtc_net_dcsctp_common_str_join webrtc_net_dcsctp_packet_bounded_io webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_packet_parameter webrtc_net_dcsctp_packet_tlv_trait webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/packet:parameter +add_library(webrtc_net_dcsctp_packet_parameter ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/forward_tsn_supported_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/heartbeat_info_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/reconfiguration_response_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/state_cookie_parameter.cc ${WEBRTC_ROOT}/net/dcsctp/packet/parameter/supported_extensions_parameter.cc) +target_include_directories(webrtc_net_dcsctp_packet_parameter PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_parameter PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_parameter PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_parameter PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_common_math webrtc_net_dcsctp_common_str_join webrtc_net_dcsctp_packet_bounded_io webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_packet_tlv_trait webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/packet:sctp_packet +add_library(webrtc_net_dcsctp_packet_sctp_packet ${WEBRTC_ROOT}/net/dcsctp/packet/sctp_packet.cc) +target_include_directories(webrtc_net_dcsctp_packet_sctp_packet PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_sctp_packet PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_sctp_packet PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_sctp_packet PUBLIC absl::memory absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_common_math webrtc_net_dcsctp_packet_bounded_io webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_crc32c webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/packet:tlv_trait +add_library(webrtc_net_dcsctp_packet_tlv_trait ${WEBRTC_ROOT}/net/dcsctp/packet/tlv_trait.cc) +target_include_directories(webrtc_net_dcsctp_packet_tlv_trait PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_packet_tlv_trait PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_packet_tlv_trait PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_packet_tlv_trait PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_packet_bounded_io webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/public:factory +add_library(webrtc_net_dcsctp_public_factory ${WEBRTC_ROOT}/net/dcsctp/public/dcsctp_socket_factory.cc) +target_include_directories(webrtc_net_dcsctp_public_factory PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_public_factory PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_public_factory PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_public_factory PUBLIC absl::strings webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_socket_dcsctp_socket ) + +# net/dcsctp/public:socket +add_library(webrtc_net_dcsctp_public_socket INTERFACE) +target_link_libraries(webrtc_net_dcsctp_public_socket INTERFACE absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) +target_include_directories(webrtc_net_dcsctp_public_socket INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/public:strong_alias +add_library(webrtc_net_dcsctp_public_strong_alias INTERFACE) +target_include_directories(webrtc_net_dcsctp_public_strong_alias INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/public:types +add_library(webrtc_net_dcsctp_public_types INTERFACE) +target_link_libraries(webrtc_net_dcsctp_public_types INTERFACE webrtc_api_array_view webrtc_net_dcsctp_public_strong_alias ) +target_include_directories(webrtc_net_dcsctp_public_types INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/rx:data_tracker +add_library(webrtc_net_dcsctp_rx_data_tracker ${WEBRTC_ROOT}/net/dcsctp/rx/data_tracker.cc) +target_include_directories(webrtc_net_dcsctp_rx_data_tracker PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_rx_data_tracker PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_rx_data_tracker PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_rx_data_tracker PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_sequence_numbers webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_timer_timer webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/rx:reassembly_queue +add_library(webrtc_net_dcsctp_rx_reassembly_queue ${WEBRTC_ROOT}/net/dcsctp/rx/reassembly_queue.cc) +target_include_directories(webrtc_net_dcsctp_rx_reassembly_queue PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_rx_reassembly_queue PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_rx_reassembly_queue PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_rx_reassembly_queue PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_common_sequence_numbers webrtc_net_dcsctp_common_str_join webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_packet_parameter webrtc_net_dcsctp_public_types webrtc_net_dcsctp_rx_reassembly_streams webrtc_net_dcsctp_rx_traditional_reassembly_streams webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/rx:reassembly_streams +add_library(webrtc_net_dcsctp_rx_reassembly_streams INTERFACE) +target_link_libraries(webrtc_net_dcsctp_rx_reassembly_streams INTERFACE absl::strings webrtc_api_array_view webrtc_net_dcsctp_common_sequence_numbers webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_public_types ) +target_include_directories(webrtc_net_dcsctp_rx_reassembly_streams INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/rx:traditional_reassembly_streams +add_library(webrtc_net_dcsctp_rx_traditional_reassembly_streams ${WEBRTC_ROOT}/net/dcsctp/rx/traditional_reassembly_streams.cc) +target_include_directories(webrtc_net_dcsctp_rx_traditional_reassembly_streams PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_rx_traditional_reassembly_streams PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_rx_traditional_reassembly_streams PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_rx_traditional_reassembly_streams PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_sequence_numbers webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_public_types webrtc_net_dcsctp_rx_reassembly_streams webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/socket:context +add_library(webrtc_net_dcsctp_socket_context INTERFACE) +target_link_libraries(webrtc_net_dcsctp_socket_context INTERFACE absl::strings webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_packet_sctp_packet webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types ) +target_include_directories(webrtc_net_dcsctp_socket_context INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# net/dcsctp/socket:dcsctp_socket +add_library(webrtc_net_dcsctp_socket_dcsctp_socket ${WEBRTC_ROOT}/net/dcsctp/socket/dcsctp_socket.cc ${WEBRTC_ROOT}/net/dcsctp/socket/state_cookie.cc) +target_include_directories(webrtc_net_dcsctp_socket_dcsctp_socket PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_socket_dcsctp_socket PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_socket_dcsctp_socket PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_socket_dcsctp_socket PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_packet_bounded_io webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_chunk_validators webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_packet_error_cause webrtc_net_dcsctp_packet_parameter webrtc_net_dcsctp_packet_sctp_packet webrtc_net_dcsctp_packet_tlv_trait webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_rx_data_tracker webrtc_net_dcsctp_rx_reassembly_queue webrtc_net_dcsctp_socket_context webrtc_net_dcsctp_socket_heartbeat_handler webrtc_net_dcsctp_socket_stream_reset_handler webrtc_net_dcsctp_socket_transmission_control_block webrtc_net_dcsctp_timer_timer webrtc_net_dcsctp_tx_retransmission_error_counter webrtc_net_dcsctp_tx_retransmission_queue webrtc_net_dcsctp_tx_retransmission_timeout webrtc_net_dcsctp_tx_rr_send_queue webrtc_net_dcsctp_tx_send_queue webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/socket:heartbeat_handler +add_library(webrtc_net_dcsctp_socket_heartbeat_handler ${WEBRTC_ROOT}/net/dcsctp/socket/heartbeat_handler.cc) +target_include_directories(webrtc_net_dcsctp_socket_heartbeat_handler PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_socket_heartbeat_handler PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_socket_heartbeat_handler PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_socket_heartbeat_handler PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_packet_bounded_io webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_parameter webrtc_net_dcsctp_packet_sctp_packet webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_socket_context webrtc_net_dcsctp_timer_timer webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/socket:stream_reset_handler +add_library(webrtc_net_dcsctp_socket_stream_reset_handler ${WEBRTC_ROOT}/net/dcsctp/socket/stream_reset_handler.cc) +target_include_directories(webrtc_net_dcsctp_socket_stream_reset_handler PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_socket_stream_reset_handler PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_socket_stream_reset_handler PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_socket_stream_reset_handler PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_common_str_join webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_parameter webrtc_net_dcsctp_packet_sctp_packet webrtc_net_dcsctp_packet_tlv_trait webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_rx_data_tracker webrtc_net_dcsctp_rx_reassembly_queue webrtc_net_dcsctp_socket_context webrtc_net_dcsctp_timer_timer webrtc_net_dcsctp_tx_retransmission_queue webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/socket:transmission_control_block +add_library(webrtc_net_dcsctp_socket_transmission_control_block ${WEBRTC_ROOT}/net/dcsctp/socket/transmission_control_block.cc) +target_include_directories(webrtc_net_dcsctp_socket_transmission_control_block PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_socket_transmission_control_block PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_socket_transmission_control_block PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_socket_transmission_control_block PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_sequence_numbers webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_sctp_packet webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_rx_data_tracker webrtc_net_dcsctp_rx_reassembly_queue webrtc_net_dcsctp_socket_context webrtc_net_dcsctp_socket_heartbeat_handler webrtc_net_dcsctp_socket_stream_reset_handler webrtc_net_dcsctp_timer_timer webrtc_net_dcsctp_tx_retransmission_error_counter webrtc_net_dcsctp_tx_retransmission_queue webrtc_net_dcsctp_tx_retransmission_timeout webrtc_net_dcsctp_tx_send_queue webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/timer:task_queue_timeout +add_library(webrtc_net_dcsctp_timer_task_queue_timeout ${WEBRTC_ROOT}/net/dcsctp/timer/task_queue_timeout.cc) +target_include_directories(webrtc_net_dcsctp_timer_task_queue_timeout PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_timer_task_queue_timeout PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_timer_task_queue_timeout PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_timer_task_queue_timeout PUBLIC webrtc_api_array_view webrtc_api_task_queue_task_queue webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_strong_alias webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task ) + +# net/dcsctp/timer +add_library(webrtc_net_dcsctp_timer_timer ${WEBRTC_ROOT}/net/dcsctp/timer/timer.cc) +target_include_directories(webrtc_net_dcsctp_timer_timer PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_timer_timer PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_timer_timer PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_timer_timer PUBLIC absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_strong_alias webrtc_net_dcsctp_public_types webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/tx:retransmission_error_counter +add_library(webrtc_net_dcsctp_tx_retransmission_error_counter ${WEBRTC_ROOT}/net/dcsctp/tx/retransmission_error_counter.cc) +target_include_directories(webrtc_net_dcsctp_tx_retransmission_error_counter PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_tx_retransmission_error_counter PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_tx_retransmission_error_counter PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_tx_retransmission_error_counter PUBLIC absl::strings webrtc_net_dcsctp_public_types webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/tx:retransmission_queue +add_library(webrtc_net_dcsctp_tx_retransmission_queue ${WEBRTC_ROOT}/net/dcsctp/tx/retransmission_queue.cc) +target_include_directories(webrtc_net_dcsctp_tx_retransmission_queue PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_tx_retransmission_queue PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_tx_retransmission_queue PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_tx_retransmission_queue PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_math webrtc_net_dcsctp_common_pair_hash webrtc_net_dcsctp_common_sequence_numbers webrtc_net_dcsctp_common_str_join webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_public_types webrtc_net_dcsctp_timer_timer webrtc_net_dcsctp_tx_retransmission_timeout webrtc_net_dcsctp_tx_send_queue webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/tx:retransmission_timeout +add_library(webrtc_net_dcsctp_tx_retransmission_timeout ${WEBRTC_ROOT}/net/dcsctp/tx/retransmission_timeout.cc) +target_include_directories(webrtc_net_dcsctp_tx_retransmission_timeout PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_tx_retransmission_timeout PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_tx_retransmission_timeout PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_tx_retransmission_timeout PUBLIC webrtc_net_dcsctp_public_types webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/tx:rr_send_queue +add_library(webrtc_net_dcsctp_tx_rr_send_queue ${WEBRTC_ROOT}/net/dcsctp/tx/rr_send_queue.cc) +target_include_directories(webrtc_net_dcsctp_tx_rr_send_queue PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_net_dcsctp_tx_rr_send_queue PRIVATE WEBRTC_MAC WEBRTC_POSIX DLOG_ALWAYS_ON WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_net_dcsctp_tx_rr_send_queue PRIVATE -fno-exceptions) +target_link_libraries(webrtc_net_dcsctp_tx_rr_send_queue PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_pair_hash webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_public_socket webrtc_net_dcsctp_public_types webrtc_net_dcsctp_tx_send_queue webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved ) + +# net/dcsctp/tx:send_queue +add_library(webrtc_net_dcsctp_tx_send_queue INTERFACE) +target_link_libraries(webrtc_net_dcsctp_tx_send_queue INTERFACE absl::optional webrtc_api_array_view webrtc_net_dcsctp_common_internal_types webrtc_net_dcsctp_packet_chunk webrtc_net_dcsctp_packet_data webrtc_net_dcsctp_public_types ) +target_include_directories(webrtc_net_dcsctp_tx_send_queue INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + # p2p:fake_ice_transport add_library(webrtc_p2p_fake_ice_transport INTERFACE) target_link_libraries(webrtc_p2p_fake_ice_transport INTERFACE absl::algorithm_container absl::optional webrtc_api_libjingle_peerconnection_api webrtc_p2p_rtc_p2p webrtc_rtc_base_rtc_base webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task ) @@ -3161,7 +3428,7 @@ add_library(webrtc_pc_audio_rtp_receiver ${WEBRTC_ROOT}/pc/audio_rtp_receiver.cc target_include_directories(webrtc_pc_audio_rtp_receiver PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_audio_rtp_receiver PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_audio_rtp_receiver PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_audio_rtp_receiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_crypto_frame_decryptor_interface webrtc_api_transport_rtp_rtp_source webrtc_media_rtc_media_base webrtc_pc_audio_track webrtc_pc_jitter_buffer_delay webrtc_pc_jitter_buffer_delay_interface webrtc_pc_jitter_buffer_delay_proxy webrtc_pc_media_stream webrtc_pc_remote_audio_source webrtc_pc_rtp_receiver webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_refcount webrtc_rtc_base_threading ) +target_link_libraries(webrtc_pc_audio_rtp_receiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_crypto_frame_decryptor_interface webrtc_api_transport_rtp_rtp_source webrtc_media_rtc_media_base webrtc_pc_audio_track webrtc_pc_jitter_buffer_delay webrtc_pc_media_stream webrtc_pc_remote_audio_source webrtc_pc_rtc_pc_base webrtc_pc_rtp_receiver webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_refcount webrtc_rtc_base_threading webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task ) # pc:audio_track add_library(webrtc_pc_audio_track ${WEBRTC_ROOT}/pc/audio_track.cc) @@ -3175,38 +3442,28 @@ add_library(webrtc_pc_connection_context ${WEBRTC_ROOT}/pc/connection_context.cc target_include_directories(webrtc_pc_connection_context PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_connection_context PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_connection_context PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_connection_context PUBLIC webrtc_api_callfactory_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_neteq_neteq_api webrtc_api_transport_field_trial_based_config webrtc_api_transport_sctp_transport_factory_interface webrtc_api_transport_webrtc_key_value_config webrtc_media_rtc_data_sctp_transport_factory webrtc_media_rtc_media_base webrtc_p2p_rtc_p2p webrtc_pc_rtc_pc_base webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_threading webrtc_rtc_base_task_utils_to_queued_task ) +target_link_libraries(webrtc_pc_connection_context PUBLIC webrtc_api_callfactory_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_neteq_neteq_api webrtc_api_transport_field_trial_based_config webrtc_api_transport_sctp_transport_factory_interface webrtc_api_transport_webrtc_key_value_config webrtc_media_rtc_data_sctp_transport_factory webrtc_media_rtc_media_base webrtc_p2p_rtc_p2p webrtc_pc_rtc_pc_base webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_threading webrtc_rtc_base_task_utils_to_queued_task ) # pc:dtmf_sender add_library(webrtc_pc_dtmf_sender ${WEBRTC_ROOT}/pc/dtmf_sender.cc) target_include_directories(webrtc_pc_dtmf_sender PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_dtmf_sender PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_dtmf_sender PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_dtmf_sender PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_libjingle_peerconnection_api webrtc_api_scoped_refptr webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_threading webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot ) +target_link_libraries(webrtc_pc_dtmf_sender PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_libjingle_peerconnection_api webrtc_api_scoped_refptr webrtc_pc_proxy webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_threading webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot ) # pc:integration_test_helpers add_library(webrtc_pc_integration_test_helpers ${WEBRTC_ROOT}/pc/test/integration_test_helpers.cc) target_include_directories(webrtc_pc_integration_test_helpers PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_integration_test_helpers PRIVATE WEBRTC_MAC WEBRTC_POSIX HAVE_WEBRTC_VIDEO WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_pc_integration_test_helpers PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_integration_test_helpers PUBLIC absl::algorithm_container absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_callfactory_api webrtc_api_create_peerconnection_factory webrtc_api_fake_frame_decryptor webrtc_api_fake_frame_encryptor webrtc_api_function_view webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_mock_rtp webrtc_api_packet_socket_factory webrtc_api_rtc_error webrtc_api_rtc_stats_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_audio_audio_mixer_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_time_delta webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_adaptation_resource_adaptation_test_utilities webrtc_logging_fake_rtc_event_log webrtc_media_rtc_audio_video webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_media_rtc_media_engine_defaults webrtc_media_rtc_media_tests_utils webrtc_modules_audio_device_audio_device_api webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_audio_processing_audioproc_test_utils webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_fake_ice_transport webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_server_utils webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_audio_rtp_receiver webrtc_pc_audio_track webrtc_pc_dtmf_sender webrtc_pc_jitter_buffer_delay webrtc_pc_jitter_buffer_delay_interface webrtc_pc_media_stream webrtc_pc_pc_test_utils webrtc_pc_peerconnection webrtc_pc_remote_audio_source webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_rtp_transceiver webrtc_pc_usage_pattern webrtc_pc_video_rtp_receiver webrtc_pc_video_rtp_track_source webrtc_pc_video_track webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_ip_address webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_json webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_timeutils webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_metrics webrtc_test_field_trial webrtc_test_fileutils webrtc_test_rtp_test_utils webrtc_test_test_support webrtc_test_pc_sctp_fake_sctp_transport ) +target_link_libraries(webrtc_pc_integration_test_helpers PUBLIC absl::algorithm_container absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_callfactory_api webrtc_api_create_peerconnection_factory webrtc_api_fake_frame_decryptor webrtc_api_fake_frame_encryptor webrtc_api_function_view webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_mock_rtp webrtc_api_packet_socket_factory webrtc_api_rtc_error webrtc_api_rtc_stats_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_audio_audio_mixer_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_time_delta webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_adaptation_resource_adaptation_test_utilities webrtc_logging_fake_rtc_event_log webrtc_media_rtc_audio_video webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_media_rtc_media_engine_defaults webrtc_media_rtc_media_tests_utils webrtc_modules_audio_device_audio_device_api webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_audio_processing_audioproc_test_utils webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_fake_ice_transport webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_server_utils webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_audio_rtp_receiver webrtc_pc_audio_track webrtc_pc_dtmf_sender webrtc_pc_jitter_buffer_delay webrtc_pc_media_stream webrtc_pc_pc_test_utils webrtc_pc_peerconnection webrtc_pc_remote_audio_source webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_rtp_transceiver webrtc_pc_session_description webrtc_pc_usage_pattern webrtc_pc_video_rtp_receiver webrtc_pc_video_rtp_track_source webrtc_pc_video_track webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_ip_address webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_json webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_timeutils webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_metrics webrtc_test_field_trial webrtc_test_fileutils webrtc_test_rtp_test_utils webrtc_test_test_support webrtc_test_pc_sctp_fake_sctp_transport ) # pc:jitter_buffer_delay add_library(webrtc_pc_jitter_buffer_delay ${WEBRTC_ROOT}/pc/jitter_buffer_delay.cc) target_include_directories(webrtc_pc_jitter_buffer_delay PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_jitter_buffer_delay PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_jitter_buffer_delay PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_jitter_buffer_delay PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_sequence_checker webrtc_media_rtc_media_base webrtc_pc_jitter_buffer_delay_interface webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_refcount webrtc_rtc_base_safe_minmax webrtc_rtc_base_threading ) - -# pc:jitter_buffer_delay_interface -add_library(webrtc_pc_jitter_buffer_delay_interface INTERFACE) -target_link_libraries(webrtc_pc_jitter_buffer_delay_interface INTERFACE absl::algorithm_container absl::strings absl::optional webrtc_media_rtc_media_base webrtc_rtc_base_refcount ) -target_include_directories(webrtc_pc_jitter_buffer_delay_interface INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) - -# pc:jitter_buffer_delay_proxy -add_library(webrtc_pc_jitter_buffer_delay_proxy INTERFACE) -target_link_libraries(webrtc_pc_jitter_buffer_delay_proxy INTERFACE webrtc_api_libjingle_peerconnection_api webrtc_media_rtc_media_base webrtc_pc_jitter_buffer_delay_interface ) -target_include_directories(webrtc_pc_jitter_buffer_delay_proxy INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_link_libraries(webrtc_pc_jitter_buffer_delay PUBLIC absl::optional webrtc_api_sequence_checker webrtc_rtc_base_checks webrtc_rtc_base_safe_conversions webrtc_rtc_base_safe_minmax webrtc_rtc_base_system_no_unique_address ) # pc:libjingle_peerconnection add_library(webrtc_pc_libjingle_peerconnection INTERFACE) @@ -3231,7 +3488,7 @@ add_library(webrtc_pc_pc_test_utils ${WEBRTC_ROOT}/pc/test/fake_audio_capture_mo target_include_directories(webrtc_pc_pc_test_utils PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_pc_test_utils PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_pc_pc_test_utils PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_pc_test_utils PUBLIC absl::optional webrtc_api_audio_options_api webrtc_api_create_frame_generator webrtc_api_create_peerconnection_factory webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtc_error webrtc_api_rtc_stats_api webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_media_rtc_media webrtc_media_rtc_media_base webrtc_media_rtc_media_tests_utils webrtc_modules_audio_device_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_jitter_buffer_delay webrtc_pc_jitter_buffer_delay_interface webrtc_pc_libjingle_peerconnection webrtc_pc_peerconnection webrtc_pc_rtc_pc_base webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_task_queue_for_test webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_test_test_support webrtc_test_video_test_common ) +target_link_libraries(webrtc_pc_pc_test_utils PUBLIC absl::optional webrtc_api_audio_options_api webrtc_api_create_frame_generator webrtc_api_create_peerconnection_factory webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtc_error webrtc_api_rtc_stats_api webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_media_rtc_media webrtc_media_rtc_media_base webrtc_media_rtc_media_tests_utils webrtc_modules_audio_device_audio_device webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_jitter_buffer_delay webrtc_pc_libjingle_peerconnection webrtc_pc_peerconnection webrtc_pc_rtc_pc_base webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_task_queue_for_test webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_test_test_support webrtc_test_video_test_common ) # pc:peer_connection_message_handler add_library(webrtc_pc_peer_connection_message_handler ${WEBRTC_ROOT}/pc/peer_connection_message_handler.cc) @@ -3245,14 +3502,21 @@ add_library(webrtc_pc_peerconnection ${WEBRTC_ROOT}/pc/data_channel_controller.c target_include_directories(webrtc_pc_peerconnection PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_peerconnection PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_peerconnection PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_peerconnection PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_async_dns_resolver webrtc_api_audio_options_api webrtc_api_call_api webrtc_api_callfactory_api webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_ice_transport_factory webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_network_state_predictor_api webrtc_api_packet_socket_factory webrtc_api_priority webrtc_api_rtc_error webrtc_api_rtc_event_log_output_file webrtc_api_rtc_stats_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_adaptation_resource_adaptation_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_datagram_transport_interface webrtc_api_transport_enums webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_api_transport_sctp_transport_factory_interface webrtc_api_transport_webrtc_key_value_config webrtc_api_units_data_rate webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_common_video_common_video webrtc_logging_ice_log webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_rtc_p2p webrtc_pc_audio_rtp_receiver webrtc_pc_audio_track webrtc_pc_connection_context webrtc_pc_dtmf_sender webrtc_pc_jitter_buffer_delay webrtc_pc_jitter_buffer_delay_interface webrtc_pc_jitter_buffer_delay_proxy webrtc_pc_media_protocol_names webrtc_pc_media_stream webrtc_pc_peer_connection_message_handler webrtc_pc_remote_audio_source webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_rtp_transceiver webrtc_pc_rtp_transmission_manager webrtc_pc_sdp_state_provider webrtc_pc_stats_collector_interface webrtc_pc_transceiver_list webrtc_pc_usage_pattern webrtc_pc_video_rtp_receiver webrtc_pc_video_track webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_callback_list webrtc_rtc_base_checks webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_operations_chain webrtc_rtc_base_safe_minmax webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_weak_ptr webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_network_sent_packet webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_stats_stats webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_pc_peerconnection PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_async_dns_resolver webrtc_api_audio_options_api webrtc_api_call_api webrtc_api_callfactory_api webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_ice_transport_factory webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_network_state_predictor_api webrtc_api_packet_socket_factory webrtc_api_priority webrtc_api_rtc_error webrtc_api_rtc_event_log_output_file webrtc_api_rtc_stats_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_adaptation_resource_adaptation_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_options webrtc_api_neteq_neteq_api webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_bitrate_settings webrtc_api_transport_datagram_transport_interface webrtc_api_transport_enums webrtc_api_transport_field_trial_based_config webrtc_api_transport_network_control webrtc_api_transport_sctp_transport_factory_interface webrtc_api_transport_webrtc_key_value_config webrtc_api_units_data_rate webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_call_rtp_sender webrtc_common_video_common_video webrtc_logging_ice_log webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_rtc_p2p webrtc_pc_audio_rtp_receiver webrtc_pc_audio_track webrtc_pc_connection_context webrtc_pc_dtmf_sender webrtc_pc_jitter_buffer_delay webrtc_pc_media_protocol_names webrtc_pc_media_stream webrtc_pc_peer_connection_message_handler webrtc_pc_proxy webrtc_pc_remote_audio_source webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_rtp_transceiver webrtc_pc_rtp_transmission_manager webrtc_pc_sdp_state_provider webrtc_pc_session_description webrtc_pc_simulcast_description webrtc_pc_stats_collector_interface webrtc_pc_transceiver_list webrtc_pc_usage_pattern webrtc_pc_video_rtp_receiver webrtc_pc_video_track webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_callback_list webrtc_rtc_base_checks webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_operations_chain webrtc_rtc_base_safe_minmax webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_weak_ptr webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_network_sent_packet webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_stats_stats webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) # pc:peerconnection_unittests android_add_executable(TARGET webrtc_pc_peerconnection_unittests NODISTRIBUTE SRC ${WEBRTC_ROOT}/pc/data_channel_integrationtest.cc ${WEBRTC_ROOT}/pc/data_channel_unittest.cc ${WEBRTC_ROOT}/pc/dtmf_sender_unittest.cc ${WEBRTC_ROOT}/pc/ice_server_parsing_unittest.cc ${WEBRTC_ROOT}/pc/jitter_buffer_delay_unittest.cc ${WEBRTC_ROOT}/pc/jsep_session_description_unittest.cc ${WEBRTC_ROOT}/pc/local_audio_source_unittest.cc ${WEBRTC_ROOT}/pc/media_stream_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_adaptation_integrationtest.cc ${WEBRTC_ROOT}/pc/peer_connection_bundle_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_crypto_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_data_channel_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_end_to_end_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_factory_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_header_extension_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_histogram_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_ice_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_integrationtest.cc ${WEBRTC_ROOT}/pc/peer_connection_interface_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_jsep_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_media_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_rtp_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_signaling_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_simulcast_unittest.cc ${WEBRTC_ROOT}/pc/peer_connection_wrapper.cc ${WEBRTC_ROOT}/pc/peer_connection_wrapper.h ${WEBRTC_ROOT}/pc/proxy_unittest.cc ${WEBRTC_ROOT}/pc/rtc_stats_collector_unittest.cc ${WEBRTC_ROOT}/pc/rtc_stats_integrationtest.cc ${WEBRTC_ROOT}/pc/rtc_stats_traversal_unittest.cc ${WEBRTC_ROOT}/pc/rtp_media_utils_unittest.cc ${WEBRTC_ROOT}/pc/rtp_parameters_conversion_unittest.cc ${WEBRTC_ROOT}/pc/rtp_sender_receiver_unittest.cc ${WEBRTC_ROOT}/pc/rtp_transceiver_unittest.cc ${WEBRTC_ROOT}/pc/sctp_utils_unittest.cc ${WEBRTC_ROOT}/pc/sdp_serializer_unittest.cc ${WEBRTC_ROOT}/pc/stats_collector_unittest.cc ${WEBRTC_ROOT}/pc/test/fake_audio_capture_module_unittest.cc ${WEBRTC_ROOT}/pc/test/test_sdp_strings.h ${WEBRTC_ROOT}/pc/track_media_info_map_unittest.cc ${WEBRTC_ROOT}/pc/video_rtp_track_source_unittest.cc ${WEBRTC_ROOT}/pc/video_track_unittest.cc ${WEBRTC_ROOT}/pc/webrtc_sdp_unittest.cc) target_include_directories(webrtc_pc_peerconnection_unittests PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_peerconnection_unittests PRIVATE WEBRTC_MAC WEBRTC_POSIX HAVE_WEBRTC_VIDEO WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_pc_peerconnection_unittests PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_peerconnection_unittests PUBLIC absl::algorithm_container absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_callfactory_api webrtc_api_create_peerconnection_factory webrtc_api_fake_frame_decryptor webrtc_api_fake_frame_encryptor webrtc_api_function_view webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_mock_rtp webrtc_api_packet_socket_factory webrtc_api_rtc_error webrtc_api_rtc_event_log_output_file webrtc_api_rtc_stats_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_audio_codecs_opus_audio_decoder_factory webrtc_api_audio_codecs_opus_audio_encoder_factory webrtc_api_audio_codecs_L16_audio_decoder_L16 webrtc_api_audio_codecs_L16_audio_encoder_L16 webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_time_delta webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_adaptation_resource_adaptation_test_utilities webrtc_logging_fake_rtc_event_log webrtc_media_rtc_audio_video webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_media_rtc_media_engine_defaults webrtc_media_rtc_media_tests_utils webrtc_modules_audio_device_audio_device_api webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_audio_processing_audioproc_test_utils webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_p2p_fake_ice_transport webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_server_utils webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_audio_rtp_receiver webrtc_pc_audio_track webrtc_pc_dtmf_sender webrtc_pc_integration_test_helpers webrtc_pc_jitter_buffer_delay webrtc_pc_jitter_buffer_delay_interface webrtc_pc_libjingle_peerconnection webrtc_pc_media_stream webrtc_pc_pc_test_utils webrtc_pc_peerconnection webrtc_pc_remote_audio_source webrtc_pc_rtc_pc webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_rtp_transceiver webrtc_pc_usage_pattern webrtc_pc_video_rtp_receiver webrtc_pc_video_rtp_track_source webrtc_pc_video_track webrtc_pc_video_track_source webrtc_pc_scenario_tests_pc_scenario_tests webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_ip_address webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_json webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_conversions webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_metrics webrtc_test_audio_codec_mocks webrtc_test_field_trial webrtc_test_fileutils webrtc_test_rtp_test_utils webrtc_test_test_main webrtc_test_test_support webrtc_test_pc_sctp_fake_sctp_transport ) +target_link_libraries(webrtc_pc_peerconnection_unittests PUBLIC absl::algorithm_container absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_callfactory_api webrtc_api_create_peerconnection_factory webrtc_api_fake_frame_decryptor webrtc_api_fake_frame_encryptor webrtc_api_function_view webrtc_api_libjingle_logging_api webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_mock_rtp webrtc_api_packet_socket_factory webrtc_api_rtc_error webrtc_api_rtc_event_log_output_file webrtc_api_rtc_stats_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_audio_audio_mixer_api webrtc_api_audio_codecs_audio_codecs_api webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_audio_codecs_opus_audio_decoder_factory webrtc_api_audio_codecs_opus_audio_encoder_factory webrtc_api_audio_codecs_L16_audio_decoder_L16 webrtc_api_audio_codecs_L16_audio_encoder_L16 webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_frame_encryptor_interface webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_task_queue_task_queue webrtc_api_task_queue_default_task_queue_factory webrtc_api_transport_field_trial_based_config webrtc_api_transport_webrtc_key_value_config webrtc_api_transport_rtp_rtp_source webrtc_api_units_time_delta webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_adaptation_resource_adaptation_test_utilities webrtc_logging_fake_rtc_event_log webrtc_media_rtc_audio_video webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_media_rtc_media_engine_defaults webrtc_media_rtc_media_tests_utils webrtc_modules_audio_device_audio_device_api webrtc_modules_audio_processing_audio_processing webrtc_modules_audio_processing_api webrtc_modules_audio_processing_audio_processing_statistics webrtc_modules_audio_processing_audioproc_test_utils webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_utility_utility webrtc_p2p_fake_ice_transport webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_server_utils webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_audio_rtp_receiver webrtc_pc_audio_track webrtc_pc_dtmf_sender webrtc_pc_integration_test_helpers webrtc_pc_jitter_buffer_delay webrtc_pc_libjingle_peerconnection webrtc_pc_media_stream webrtc_pc_pc_test_utils webrtc_pc_peerconnection webrtc_pc_proxy webrtc_pc_remote_audio_source webrtc_pc_rtc_pc webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_rtp_transceiver webrtc_pc_session_description webrtc_pc_usage_pattern webrtc_pc_video_rtp_receiver webrtc_pc_video_rtp_track_source webrtc_pc_video_track webrtc_pc_video_track_source webrtc_pc_scenario_tests_pc_scenario_tests webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_ip_address webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_json webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_conversions webrtc_rtc_base_socket_address webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics webrtc_test_audio_codec_mocks webrtc_test_field_trial webrtc_test_fileutils webrtc_test_rtp_test_utils webrtc_test_test_common webrtc_test_test_main webrtc_test_test_support webrtc_test_pc_sctp_fake_sctp_transport ) + +# pc:proxy +add_library(webrtc_pc_proxy ${WEBRTC_ROOT}/pc/proxy.cc) +target_include_directories(webrtc_pc_proxy PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_pc_proxy PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_pc_proxy PRIVATE -fno-exceptions) +target_link_libraries(webrtc_pc_proxy PUBLIC webrtc_api_scoped_refptr webrtc_api_task_queue_task_queue webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading webrtc_rtc_base_system_rtc_export ) # pc:remote_audio_source add_library(webrtc_pc_remote_audio_source ${WEBRTC_ROOT}/pc/remote_audio_source.cc) @@ -3267,32 +3531,32 @@ target_link_libraries(webrtc_pc_rtc_pc INTERFACE libsrtp webrtc_media_rtc_audio_ target_include_directories(webrtc_pc_rtc_pc INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # pc:rtc_pc_base -add_library(webrtc_pc_rtc_pc_base ${WEBRTC_ROOT}/pc/channel.cc ${WEBRTC_ROOT}/pc/channel_manager.cc ${WEBRTC_ROOT}/pc/dtls_srtp_transport.cc ${WEBRTC_ROOT}/pc/dtls_transport.cc ${WEBRTC_ROOT}/pc/external_hmac.cc ${WEBRTC_ROOT}/pc/ice_transport.cc ${WEBRTC_ROOT}/pc/jsep_transport.cc ${WEBRTC_ROOT}/pc/jsep_transport_controller.cc ${WEBRTC_ROOT}/pc/media_session.cc ${WEBRTC_ROOT}/pc/rtcp_mux_filter.cc ${WEBRTC_ROOT}/pc/rtp_media_utils.cc ${WEBRTC_ROOT}/pc/rtp_transport.cc ${WEBRTC_ROOT}/pc/sctp_data_channel_transport.cc ${WEBRTC_ROOT}/pc/sctp_transport.cc ${WEBRTC_ROOT}/pc/sctp_utils.cc ${WEBRTC_ROOT}/pc/session_description.cc ${WEBRTC_ROOT}/pc/simulcast_description.cc ${WEBRTC_ROOT}/pc/srtp_filter.cc ${WEBRTC_ROOT}/pc/srtp_session.cc ${WEBRTC_ROOT}/pc/srtp_transport.cc ${WEBRTC_ROOT}/pc/transport_stats.cc) +add_library(webrtc_pc_rtc_pc_base ${WEBRTC_ROOT}/pc/channel.cc ${WEBRTC_ROOT}/pc/channel_manager.cc ${WEBRTC_ROOT}/pc/dtls_srtp_transport.cc ${WEBRTC_ROOT}/pc/dtls_transport.cc ${WEBRTC_ROOT}/pc/external_hmac.cc ${WEBRTC_ROOT}/pc/ice_transport.cc ${WEBRTC_ROOT}/pc/jsep_transport.cc ${WEBRTC_ROOT}/pc/jsep_transport_collection.cc ${WEBRTC_ROOT}/pc/jsep_transport_controller.cc ${WEBRTC_ROOT}/pc/media_session.cc ${WEBRTC_ROOT}/pc/rtcp_mux_filter.cc ${WEBRTC_ROOT}/pc/rtp_media_utils.cc ${WEBRTC_ROOT}/pc/rtp_transport.cc ${WEBRTC_ROOT}/pc/sctp_data_channel_transport.cc ${WEBRTC_ROOT}/pc/sctp_transport.cc ${WEBRTC_ROOT}/pc/sctp_utils.cc ${WEBRTC_ROOT}/pc/srtp_filter.cc ${WEBRTC_ROOT}/pc/srtp_session.cc ${WEBRTC_ROOT}/pc/srtp_transport.cc ${WEBRTC_ROOT}/pc/transport_stats.cc ${WEBRTC_ROOT}/pc/video_track_source_proxy.cc) target_include_directories(webrtc_pc_rtc_pc_base PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_rtc_pc_base PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_rtc_pc_base PRIVATE -fno-exceptions -Ithird_party/libsrtp/include) -target_link_libraries(webrtc_pc_rtc_pc_base PUBLIC absl::algorithm_container absl::core_headers absl::memory absl::strings absl::optional libsrtp webrtc_api_array_view webrtc_api_async_dns_resolver webrtc_api_audio_options_api webrtc_api_call_api webrtc_api_function_view webrtc_api_ice_transport_factory webrtc_api_libjingle_peerconnection_api webrtc_api_packet_socket_factory webrtc_api_priority webrtc_api_rtc_error webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_datagram_transport_interface webrtc_api_transport_enums webrtc_api_transport_sctp_transport_factory_interface webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_common_video_common_video webrtc_logging_ice_log webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_media_rtc_sdp_video_format_utils webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_rtc_p2p webrtc_pc_media_protocol_names webrtc_rtc_base_rtc_base webrtc_rtc_base_callback_list webrtc_rtc_base_checks webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_socket webrtc_rtc_base_socket_address webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_network_sent_packet webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) +target_link_libraries(webrtc_pc_rtc_pc_base PUBLIC absl::algorithm_container absl::core_headers absl::memory absl::strings absl::optional libsrtp webrtc_api_array_view webrtc_api_async_dns_resolver webrtc_api_audio_options_api webrtc_api_call_api webrtc_api_function_view webrtc_api_ice_transport_factory webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_packet_socket_factory webrtc_api_priority webrtc_api_rtc_error webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_transport_datagram_transport_interface webrtc_api_transport_enums webrtc_api_transport_sctp_transport_factory_interface webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_common_video_common_video webrtc_logging_ice_log webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_config webrtc_media_rtc_sdp_video_format_utils webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_rtc_p2p webrtc_pc_media_protocol_names webrtc_pc_proxy webrtc_pc_session_description webrtc_pc_simulcast_description webrtc_rtc_base_rtc_base webrtc_rtc_base_callback_list webrtc_rtc_base_checks webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_socket webrtc_rtc_base_socket_address webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_network_sent_packet webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics ) # pc:rtc_pc_unittests android_add_executable(TARGET webrtc_pc_rtc_pc_unittests NODISTRIBUTE SRC ${WEBRTC_ROOT}/pc/channel_manager_unittest.cc ${WEBRTC_ROOT}/pc/channel_unittest.cc ${WEBRTC_ROOT}/pc/dtls_srtp_transport_unittest.cc ${WEBRTC_ROOT}/pc/dtls_transport_unittest.cc ${WEBRTC_ROOT}/pc/ice_transport_unittest.cc ${WEBRTC_ROOT}/pc/jsep_transport_controller_unittest.cc ${WEBRTC_ROOT}/pc/jsep_transport_unittest.cc ${WEBRTC_ROOT}/pc/media_session_unittest.cc ${WEBRTC_ROOT}/pc/rtcp_mux_filter_unittest.cc ${WEBRTC_ROOT}/pc/rtp_transport_unittest.cc ${WEBRTC_ROOT}/pc/sctp_transport_unittest.cc ${WEBRTC_ROOT}/pc/session_description_unittest.cc ${WEBRTC_ROOT}/pc/srtp_filter_unittest.cc ${WEBRTC_ROOT}/pc/srtp_session_unittest.cc ${WEBRTC_ROOT}/pc/srtp_transport_unittest.cc ${WEBRTC_ROOT}/pc/test/rtp_transport_test_util.h ${WEBRTC_ROOT}/pc/test/srtp_test_util.h ${WEBRTC_ROOT}/pc/used_ids_unittest.cc ${WEBRTC_ROOT}/pc/video_rtp_receiver_unittest.cc) target_include_directories(webrtc_pc_rtc_pc_unittests PRIVATE ${WEBRTC_ROOT}/pc/../third_party/libsrtp/srtp ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_rtc_pc_unittests PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_pc_rtc_pc_unittests PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_rtc_pc_unittests PUBLIC absl::algorithm_container absl::memory absl::strings webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_ice_transport_factory webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_error webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_test_mock_recordable_encoded_frame webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_tests_utils webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_fake_ice_transport webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_libjingle_peerconnection webrtc_pc_pc_test_utils webrtc_pc_peerconnection webrtc_pc_rtc_pc webrtc_pc_rtc_pc_base webrtc_pc_video_rtp_receiver webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_threading webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_metrics webrtc_test_field_trial webrtc_test_test_main webrtc_test_test_support ) +target_link_libraries(webrtc_pc_rtc_pc_unittests PUBLIC absl::algorithm_container absl::memory absl::strings webrtc_api_array_view webrtc_api_audio_options_api webrtc_api_ice_transport_factory webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_error webrtc_api_rtp_headers webrtc_api_rtp_parameters webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_test_mock_recordable_encoded_frame webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_media_rtc_data_sctp_transport_internal webrtc_media_rtc_media_base webrtc_media_rtc_media_tests_utils webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_fake_ice_transport webrtc_p2p_fake_port_allocator webrtc_p2p_p2p_test_utils webrtc_p2p_rtc_p2p webrtc_pc_libjingle_peerconnection webrtc_pc_pc_test_utils webrtc_pc_peerconnection webrtc_pc_rtc_pc webrtc_pc_rtc_pc_base webrtc_pc_session_description webrtc_pc_video_rtp_receiver webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_gunit_helpers webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_threading webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_metrics webrtc_test_field_trial webrtc_test_test_main webrtc_test_test_support ) # pc:rtp_parameters_conversion add_library(webrtc_pc_rtp_parameters_conversion ${WEBRTC_ROOT}/pc/rtp_parameters_conversion.cc) target_include_directories(webrtc_pc_rtp_parameters_conversion PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_rtp_parameters_conversion PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_rtp_parameters_conversion PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_rtp_parameters_conversion PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_error webrtc_api_rtp_parameters webrtc_media_rtc_media_base webrtc_pc_rtc_pc_base webrtc_rtc_base_rtc_base webrtc_rtc_base_checks ) +target_link_libraries(webrtc_pc_rtp_parameters_conversion PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_error webrtc_api_rtp_parameters webrtc_media_rtc_media_base webrtc_pc_rtc_pc_base webrtc_pc_session_description webrtc_rtc_base_rtc_base webrtc_rtc_base_checks ) # pc:rtp_receiver add_library(webrtc_pc_rtp_receiver ${WEBRTC_ROOT}/pc/rtp_receiver.cc) target_include_directories(webrtc_pc_rtp_receiver PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_rtp_receiver PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_rtp_receiver PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_rtp_receiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_crypto_frame_decryptor_interface webrtc_api_video_video_frame webrtc_media_rtc_media_base webrtc_pc_media_stream webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_logging webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading ) +target_link_libraries(webrtc_pc_rtp_receiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_crypto_frame_decryptor_interface webrtc_api_video_video_frame webrtc_media_rtc_media_base webrtc_pc_media_stream webrtc_pc_rtc_pc_base webrtc_pc_video_track_source webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_logging webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading ) # pc:rtp_sender add_library(webrtc_pc_rtp_sender ${WEBRTC_ROOT}/pc/rtp_sender.cc) @@ -3306,7 +3570,7 @@ add_library(webrtc_pc_rtp_transceiver ${WEBRTC_ROOT}/pc/rtp_transceiver.cc) target_include_directories(webrtc_pc_rtp_transceiver PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_rtp_transceiver PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_rtp_transceiver PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_rtp_transceiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_error webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_task_queue_task_queue webrtc_media_rtc_media_base webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_rtc_base_checks webrtc_rtc_base_logging webrtc_rtc_base_macromagic webrtc_rtc_base_refcount webrtc_rtc_base_threading webrtc_rtc_base_third_party_sigslot_sigslot ) +target_link_libraries(webrtc_pc_rtp_transceiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_array_view webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_error webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_task_queue_task_queue webrtc_media_rtc_media_base webrtc_pc_proxy webrtc_pc_rtc_pc_base webrtc_pc_rtp_parameters_conversion webrtc_pc_rtp_receiver webrtc_pc_rtp_sender webrtc_pc_session_description webrtc_rtc_base_checks webrtc_rtc_base_logging webrtc_rtc_base_macromagic webrtc_rtc_base_refcount webrtc_rtc_base_threading webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot ) # pc:rtp_transmission_manager add_library(webrtc_pc_rtp_transmission_manager ${WEBRTC_ROOT}/pc/rtp_transmission_manager.cc) @@ -3327,6 +3591,20 @@ add_library(webrtc_pc_sdp_state_provider INTERFACE) target_link_libraries(webrtc_pc_sdp_state_provider INTERFACE webrtc_api_libjingle_peerconnection_api webrtc_pc_rtc_pc_base ) target_include_directories(webrtc_pc_sdp_state_provider INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +# pc:session_description +add_library(webrtc_pc_session_description ${WEBRTC_ROOT}/pc/session_description.cc) +target_include_directories(webrtc_pc_session_description PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_pc_session_description PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_pc_session_description PRIVATE -fno-exceptions) +target_link_libraries(webrtc_pc_session_description PUBLIC absl::algorithm_container absl::memory webrtc_api_libjingle_peerconnection_api webrtc_api_rtp_parameters webrtc_api_rtp_transceiver_direction webrtc_media_rtc_media_base webrtc_p2p_rtc_p2p webrtc_pc_media_protocol_names webrtc_pc_simulcast_description webrtc_rtc_base_checks webrtc_rtc_base_socket_address webrtc_rtc_base_system_rtc_export ) + +# pc:simulcast_description +add_library(webrtc_pc_simulcast_description ${WEBRTC_ROOT}/pc/simulcast_description.cc) +target_include_directories(webrtc_pc_simulcast_description PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +target_compile_definitions(webrtc_pc_simulcast_description PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) +target_compile_options(webrtc_pc_simulcast_description PRIVATE -fno-exceptions) +target_link_libraries(webrtc_pc_simulcast_description PUBLIC webrtc_rtc_base_checks webrtc_rtc_base_socket_address webrtc_rtc_base_system_rtc_export ) + # pc:stats_collector_interface add_library(webrtc_pc_stats_collector_interface INTERFACE) target_link_libraries(webrtc_pc_stats_collector_interface INTERFACE webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface ) @@ -3351,7 +3629,7 @@ add_library(webrtc_pc_video_rtp_receiver ${WEBRTC_ROOT}/pc/video_rtp_receiver.cc target_include_directories(webrtc_pc_video_rtp_receiver PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_pc_video_rtp_receiver PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_pc_video_rtp_receiver PRIVATE -fno-exceptions) -target_link_libraries(webrtc_pc_video_rtp_receiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_crypto_frame_decryptor_interface webrtc_api_transport_rtp_rtp_source webrtc_api_video_recordable_encoded_frame webrtc_api_video_video_frame webrtc_media_rtc_media_base webrtc_pc_jitter_buffer_delay webrtc_pc_jitter_buffer_delay_interface webrtc_pc_jitter_buffer_delay_proxy webrtc_pc_media_stream webrtc_pc_rtp_receiver webrtc_pc_video_rtp_track_source webrtc_pc_video_track webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading ) +target_link_libraries(webrtc_pc_video_rtp_receiver PUBLIC absl::algorithm_container absl::strings absl::optional webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_crypto_frame_decryptor_interface webrtc_api_transport_rtp_rtp_source webrtc_api_video_recordable_encoded_frame webrtc_api_video_video_frame webrtc_media_rtc_media_base webrtc_pc_jitter_buffer_delay webrtc_pc_media_stream webrtc_pc_rtc_pc_base webrtc_pc_rtp_receiver webrtc_pc_video_rtp_track_source webrtc_pc_video_track webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_threading webrtc_rtc_base_system_no_unique_address ) # pc:video_rtp_track_source add_library(webrtc_pc_video_rtp_track_source ${WEBRTC_ROOT}/pc/video_rtp_track_source.cc) @@ -3719,7 +3997,7 @@ add_library(webrtc_rtc_base_platform_thread ${WEBRTC_ROOT}/rtc_base/platform_thr target_include_directories(webrtc_rtc_base_platform_thread PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_rtc_base_platform_thread PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_rtc_base_platform_thread PRIVATE -fno-exceptions) -target_link_libraries(webrtc_rtc_base_platform_thread PUBLIC absl::strings webrtc_api_sequence_checker webrtc_rtc_base_atomicops webrtc_rtc_base_checks webrtc_rtc_base_macromagic webrtc_rtc_base_platform_thread_types webrtc_rtc_base_rtc_event webrtc_rtc_base_timeutils ) +target_link_libraries(webrtc_rtc_base_platform_thread PUBLIC absl::memory absl::strings absl::optional webrtc_api_sequence_checker webrtc_rtc_base_atomicops webrtc_rtc_base_checks webrtc_rtc_base_macromagic webrtc_rtc_base_platform_thread_types webrtc_rtc_base_rtc_event webrtc_rtc_base_timeutils ) # rtc_base:platform_thread.headers add_library(webrtc_rtc_base_platform_thread.headers INTERFACE) @@ -3738,6 +4016,15 @@ add_library(webrtc_rtc_base_platform_thread_types.headers INTERFACE) target_link_libraries(webrtc_rtc_base_platform_thread_types.headers INTERFACE webrtc_rtc_base_macromagic.headers ) target_include_directories(webrtc_rtc_base_platform_thread_types.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) +# rtc_base:protobuf_utils +add_library(webrtc_rtc_base_protobuf_utils INTERFACE) +target_link_libraries(webrtc_rtc_base_protobuf_utils INTERFACE libprotobuf ) +target_include_directories(webrtc_rtc_base_protobuf_utils INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + +# rtc_base:protobuf_utils.headers +add_library(webrtc_rtc_base_protobuf_utils.headers INTERFACE) +target_include_directories(webrtc_rtc_base_protobuf_utils.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) + # rtc_base:rate_limiter add_library(webrtc_rtc_base_rate_limiter ${WEBRTC_ROOT}/rtc_base/rate_limiter.cc) target_include_directories(webrtc_rtc_base_rate_limiter PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) @@ -3747,12 +4034,12 @@ target_link_libraries(webrtc_rtc_base_rate_limiter PUBLIC absl::optional webrtc_ # rtc_base:refcount add_library(webrtc_rtc_base_refcount INTERFACE) -target_link_libraries(webrtc_rtc_base_refcount INTERFACE webrtc_rtc_base_macromagic ) +target_link_libraries(webrtc_rtc_base_refcount INTERFACE webrtc_api_scoped_refptr webrtc_rtc_base_macromagic ) target_include_directories(webrtc_rtc_base_refcount INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # rtc_base:refcount.headers add_library(webrtc_rtc_base_refcount.headers INTERFACE) -target_link_libraries(webrtc_rtc_base_refcount.headers INTERFACE webrtc_rtc_base_macromagic.headers ) +target_link_libraries(webrtc_rtc_base_refcount.headers INTERFACE webrtc_api_scoped_refptr.headers webrtc_rtc_base_macromagic.headers ) target_include_directories(webrtc_rtc_base_refcount.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # rtc_base @@ -3760,11 +4047,11 @@ add_library(webrtc_rtc_base_rtc_base ${WEBRTC_ROOT}/rtc_base/mac_ifaddrs_convert target_include_directories(webrtc_rtc_base_rtc_base PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_rtc_base_rtc_base PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_rtc_base_rtc_base PRIVATE -fno-exceptions) -target_link_libraries(webrtc_rtc_base_rtc_base PUBLIC absl::algorithm_container absl::flat_hash_map absl::memory absl::strings absl::optional jsoncpp ssl webrtc_api_array_view webrtc_api_function_view webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_numerics_numerics webrtc_api_task_queue_task_queue webrtc_rtc_base_async_resolver_interface webrtc_rtc_base_async_socket webrtc_rtc_base_checks webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_null_socket_server webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_socket webrtc_rtc_base_socket_address webrtc_rtc_base_socket_factory webrtc_rtc_base_socket_server webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_network_sent_packet webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_inline webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial ) +target_link_libraries(webrtc_rtc_base_rtc_base PUBLIC absl::algorithm_container absl::flat_hash_map absl::memory absl::strings absl::optional jsoncpp ssl webrtc_api_array_view webrtc_api_function_view webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_numerics_numerics webrtc_api_task_queue_task_queue webrtc_rtc_base_async_resolver_interface webrtc_rtc_base_async_socket webrtc_rtc_base_checks webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_null_socket_server webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_socket webrtc_rtc_base_socket_address webrtc_rtc_base_socket_factory webrtc_rtc_base_socket_server webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_network_sent_packet webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_file_wrapper webrtc_rtc_base_system_inline webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_base64_base64 webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_field_trial ) # rtc_base:rtc_base.headers add_library(webrtc_rtc_base_rtc_base.headers INTERFACE) -target_link_libraries(webrtc_rtc_base_rtc_base.headers INTERFACE webrtc_api_array_view.headers webrtc_api_function_view.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_numerics_numerics.headers webrtc_api_task_queue_task_queue.headers webrtc_rtc_base_async_resolver_interface.headers webrtc_rtc_base_async_socket.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_ip_address.headers webrtc_rtc_base_network_constants.headers webrtc_rtc_base_null_socket_server.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_numerics.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_socket.headers webrtc_rtc_base_socket_address.headers webrtc_rtc_base_socket_factory.headers webrtc_rtc_base_socket_server.headers webrtc_rtc_base_stringutils.headers webrtc_rtc_base_threading.headers webrtc_rtc_base_network_sent_packet.headers webrtc_rtc_base_synchronization_mutex.headers webrtc_rtc_base_system_file_wrapper.headers webrtc_rtc_base_system_inline.headers webrtc_rtc_base_system_no_unique_address.headers webrtc_rtc_base_system_rtc_export.headers webrtc_rtc_base_task_utils_pending_task_safety_flag.headers webrtc_rtc_base_task_utils_repeating_task.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_rtc_base_third_party_base64_base64.headers webrtc_rtc_base_third_party_sigslot_sigslot.headers webrtc_system_wrappers_field_trial.headers ) +target_link_libraries(webrtc_rtc_base_rtc_base.headers INTERFACE webrtc_api_array_view.headers webrtc_api_function_view.headers webrtc_api_refcountedbase.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_numerics_numerics.headers webrtc_api_task_queue_task_queue.headers webrtc_rtc_base_async_resolver_interface.headers webrtc_rtc_base_async_socket.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_ip_address.headers webrtc_rtc_base_network_constants.headers webrtc_rtc_base_null_socket_server.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_numerics.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_socket.headers webrtc_rtc_base_socket_address.headers webrtc_rtc_base_socket_factory.headers webrtc_rtc_base_socket_server.headers webrtc_rtc_base_stringutils.headers webrtc_rtc_base_threading.headers webrtc_rtc_base_network_sent_packet.headers webrtc_rtc_base_synchronization_mutex.headers webrtc_rtc_base_system_file_wrapper.headers webrtc_rtc_base_system_inline.headers webrtc_rtc_base_system_no_unique_address.headers webrtc_rtc_base_system_rtc_export.headers webrtc_rtc_base_task_utils_pending_task_safety_flag.headers webrtc_rtc_base_task_utils_repeating_task.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_rtc_base_third_party_base64_base64.headers webrtc_rtc_base_third_party_sigslot_sigslot.headers webrtc_system_wrappers_field_trial.headers ) target_include_directories(webrtc_rtc_base_rtc_base.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # rtc_base:rtc_base_approved @@ -4105,11 +4392,11 @@ add_library(webrtc_rtc_base_task_utils_pending_task_safety_flag ${WEBRTC_ROOT}/r target_include_directories(webrtc_rtc_base_task_utils_pending_task_safety_flag PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_rtc_base_task_utils_pending_task_safety_flag PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_rtc_base_task_utils_pending_task_safety_flag PRIVATE -fno-exceptions) -target_link_libraries(webrtc_rtc_base_task_utils_pending_task_safety_flag PUBLIC webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_rtc_base_checks webrtc_rtc_base_refcount webrtc_rtc_base_system_no_unique_address ) +target_link_libraries(webrtc_rtc_base_task_utils_pending_task_safety_flag PUBLIC webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_rtc_base_checks webrtc_rtc_base_system_no_unique_address ) # rtc_base/task_utils:pending_task_safety_flag.headers add_library(webrtc_rtc_base_task_utils_pending_task_safety_flag.headers INTERFACE) -target_link_libraries(webrtc_rtc_base_task_utils_pending_task_safety_flag.headers INTERFACE webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_refcount.headers webrtc_rtc_base_system_no_unique_address.headers ) +target_link_libraries(webrtc_rtc_base_task_utils_pending_task_safety_flag.headers INTERFACE webrtc_api_refcountedbase.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_system_no_unique_address.headers ) target_include_directories(webrtc_rtc_base_task_utils_pending_task_safety_flag.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # rtc_base/task_utils:repeating_task @@ -4117,11 +4404,11 @@ add_library(webrtc_rtc_base_task_utils_repeating_task ${WEBRTC_ROOT}/rtc_base/ta target_include_directories(webrtc_rtc_base_task_utils_repeating_task PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_rtc_base_task_utils_repeating_task PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_rtc_base_task_utils_repeating_task PRIVATE -fno-exceptions) -target_link_libraries(webrtc_rtc_base_task_utils_repeating_task PUBLIC absl::memory webrtc_api_sequence_checker webrtc_api_task_queue_task_queue webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_rtc_base_logging webrtc_rtc_base_timeutils webrtc_rtc_base_task_utils_to_queued_task webrtc_system_wrappers_system_wrappers ) +target_link_libraries(webrtc_rtc_base_task_utils_repeating_task PUBLIC absl::memory webrtc_api_sequence_checker webrtc_api_task_queue_task_queue webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_rtc_base_logging webrtc_rtc_base_timeutils webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_system_wrappers_system_wrappers ) # rtc_base/task_utils:repeating_task.headers add_library(webrtc_rtc_base_task_utils_repeating_task.headers INTERFACE) -target_link_libraries(webrtc_rtc_base_task_utils_repeating_task.headers INTERFACE webrtc_api_sequence_checker.headers webrtc_api_task_queue_task_queue.headers webrtc_api_units_time_delta.headers webrtc_api_units_timestamp.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_timeutils.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_system_wrappers_system_wrappers.headers ) +target_link_libraries(webrtc_rtc_base_task_utils_repeating_task.headers INTERFACE webrtc_api_sequence_checker.headers webrtc_api_task_queue_task_queue.headers webrtc_api_units_time_delta.headers webrtc_api_units_timestamp.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_timeutils.headers webrtc_rtc_base_task_utils_pending_task_safety_flag.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_system_wrappers_system_wrappers.headers ) target_include_directories(webrtc_rtc_base_task_utils_repeating_task.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # rtc_base/task_utils:to_queued_task @@ -4161,11 +4448,11 @@ add_library(webrtc_rtc_base_threading ${WEBRTC_ROOT}/rtc_base/async_resolver.cc target_include_directories(webrtc_rtc_base_threading PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_rtc_base_threading PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_rtc_base_threading PRIVATE -fno-exceptions) -target_link_libraries(webrtc_rtc_base_threading PUBLIC webrtc_rtc_base_system_cocoa_threading absl::algorithm_container webrtc_api_function_view webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_task_queue_task_queue webrtc_rtc_base_async_resolver_interface webrtc_rtc_base_atomicops webrtc_rtc_base_checks webrtc_rtc_base_criticalsection webrtc_rtc_base_ip_address webrtc_rtc_base_logging webrtc_rtc_base_macromagic webrtc_rtc_base_network_constants webrtc_rtc_base_null_socket_server webrtc_rtc_base_platform_thread_types webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_event webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_socket_address webrtc_rtc_base_socket_server webrtc_rtc_base_timeutils webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot ) +target_link_libraries(webrtc_rtc_base_threading PUBLIC webrtc_rtc_base_system_cocoa_threading absl::algorithm_container webrtc_api_function_view webrtc_api_refcountedbase webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_task_queue_task_queue webrtc_rtc_base_async_resolver_interface webrtc_rtc_base_atomicops webrtc_rtc_base_checks webrtc_rtc_base_criticalsection webrtc_rtc_base_ip_address webrtc_rtc_base_logging webrtc_rtc_base_macromagic webrtc_rtc_base_network_constants webrtc_rtc_base_null_socket_server webrtc_rtc_base_platform_thread_types webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_event webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_socket_address webrtc_rtc_base_socket_server webrtc_rtc_base_timeutils webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_rtc_export webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot ) # rtc_base:threading.headers add_library(webrtc_rtc_base_threading.headers INTERFACE) -target_link_libraries(webrtc_rtc_base_threading.headers INTERFACE webrtc_rtc_base_system_cocoa_threading.headers webrtc_api_function_view.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_task_queue_task_queue.headers webrtc_rtc_base_async_resolver_interface.headers webrtc_rtc_base_atomicops.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_criticalsection.headers webrtc_rtc_base_ip_address.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_macromagic.headers webrtc_rtc_base_network_constants.headers webrtc_rtc_base_null_socket_server.headers webrtc_rtc_base_platform_thread_types.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_event.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_socket_address.headers webrtc_rtc_base_socket_server.headers webrtc_rtc_base_timeutils.headers webrtc_rtc_base_synchronization_mutex.headers webrtc_rtc_base_system_no_unique_address.headers webrtc_rtc_base_system_rtc_export.headers webrtc_rtc_base_task_utils_pending_task_safety_flag.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_rtc_base_third_party_sigslot_sigslot.headers ) +target_link_libraries(webrtc_rtc_base_threading.headers INTERFACE webrtc_rtc_base_system_cocoa_threading.headers webrtc_api_function_view.headers webrtc_api_refcountedbase.headers webrtc_api_scoped_refptr.headers webrtc_api_sequence_checker.headers webrtc_api_task_queue_task_queue.headers webrtc_rtc_base_async_resolver_interface.headers webrtc_rtc_base_atomicops.headers webrtc_rtc_base_checks.headers webrtc_rtc_base_criticalsection.headers webrtc_rtc_base_ip_address.headers webrtc_rtc_base_logging.headers webrtc_rtc_base_macromagic.headers webrtc_rtc_base_network_constants.headers webrtc_rtc_base_null_socket_server.headers webrtc_rtc_base_platform_thread_types.headers webrtc_rtc_base_rtc_base_approved.headers webrtc_rtc_base_rtc_event.headers webrtc_rtc_base_rtc_task_queue.headers webrtc_rtc_base_socket_address.headers webrtc_rtc_base_socket_server.headers webrtc_rtc_base_timeutils.headers webrtc_rtc_base_synchronization_mutex.headers webrtc_rtc_base_system_no_unique_address.headers webrtc_rtc_base_system_rtc_export.headers webrtc_rtc_base_task_utils_pending_task_safety_flag.headers webrtc_rtc_base_task_utils_to_queued_task.headers webrtc_rtc_base_third_party_sigslot_sigslot.headers ) target_include_directories(webrtc_rtc_base_threading.headers INTERFACE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) # rtc_base/time:timestamp_extrapolator @@ -4349,7 +4636,7 @@ add_library(webrtc_sdk_videosource_objc ${WEBRTC_ROOT}/sdk/objc/api/peerconnecti target_include_directories(webrtc_sdk_videosource_objc PRIVATE ${WEBRTC_ROOT}/sdk/objc ${WEBRTC_ROOT}/sdk/objc/base ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_sdk_videosource_objc PRIVATE WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_MAC WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_POSIX RTC_SUPPORTS_METAL) target_compile_options(webrtc_sdk_videosource_objc PRIVATE -Wimplicit-retain-self -Wmissing-field-initializers -Wstrict-overflow -fno-exceptions -fobjc-weak) -target_link_libraries(webrtc_sdk_videosource_objc PUBLIC webrtc-yuv webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_threading webrtc_sdk_base_objc webrtc_sdk_mediasource_objc webrtc_sdk_native_video webrtc_sdk_videoframebuffer_objc ) +target_link_libraries(webrtc_sdk_videosource_objc PUBLIC webrtc-yuv webrtc_api_libjingle_peerconnection_api webrtc_api_media_stream_interface webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_pc_rtc_pc_base webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_threading webrtc_sdk_base_objc webrtc_sdk_mediasource_objc webrtc_sdk_native_video webrtc_sdk_videoframebuffer_objc ) target_link_libraries(webrtc_sdk_videosource_objc PRIVATE "-framework Foundation") # sdk:videotoolbox_objc @@ -4511,7 +4798,7 @@ add_library(webrtc_test_network_emulated_network ${WEBRTC_ROOT}/test/network/cro target_include_directories(webrtc_test_network_emulated_network PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_test_network_emulated_network PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_test_network_emulated_network PRIVATE -fno-exceptions) -target_link_libraries(webrtc_test_network_emulated_network PUBLIC absl::algorithm_container absl::memory absl::optional webrtc_api_array_view webrtc_api_network_emulation_manager_api webrtc_api_packet_socket_factory webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_simulated_network_api webrtc_api_time_controller webrtc_api_numerics_numerics webrtc_api_test_network_emulation_network_emulation webrtc_api_transport_stun_types webrtc_api_units_data_rate webrtc_api_units_data_size webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_call_simulated_network webrtc_p2p_p2p_server_utils webrtc_rtc_base_rtc_base webrtc_rtc_base_async_socket webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_socket_address webrtc_rtc_base_socket_server webrtc_rtc_base_stringutils webrtc_rtc_base_task_queue_for_test webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_third_party_sigslot_sigslot webrtc_system_wrappers_system_wrappers webrtc_test_scenario_column_printer webrtc_test_time_controller_time_controller ) +target_link_libraries(webrtc_test_network_emulated_network PUBLIC absl::algorithm_container absl::memory absl::optional webrtc_api_array_view webrtc_api_network_emulation_manager_api webrtc_api_packet_socket_factory webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_simulated_network_api webrtc_api_time_controller webrtc_api_numerics_numerics webrtc_api_test_network_emulation_network_emulation webrtc_api_transport_stun_types webrtc_api_units_data_rate webrtc_api_units_data_size webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_call_simulated_network webrtc_p2p_p2p_server_utils webrtc_rtc_base_rtc_base webrtc_rtc_base_async_socket webrtc_rtc_base_ip_address webrtc_rtc_base_network_constants webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_socket_address webrtc_rtc_base_socket_server webrtc_rtc_base_stringutils webrtc_rtc_base_task_queue_for_test webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_task_utils_to_queued_task webrtc_system_wrappers_system_wrappers webrtc_test_scenario_column_printer webrtc_test_time_controller_time_controller ) # test/pc/sctp:fake_sctp_transport add_library(webrtc_test_pc_sctp_fake_sctp_transport INTERFACE) @@ -4523,7 +4810,7 @@ add_library(webrtc_test_peer_scenario_peer_scenario ${WEBRTC_ROOT}/test/peer_sce target_include_directories(webrtc_test_peer_scenario_peer_scenario PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_test_peer_scenario_peer_scenario PRIVATE WEBRTC_MAC WEBRTC_POSIX HAVE_WEBRTC_VIDEO WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_test_peer_scenario_peer_scenario PRIVATE -fno-exceptions) -target_link_libraries(webrtc_test_peer_scenario_peer_scenario PUBLIC absl::flags absl::memory webrtc_api_create_time_controller webrtc_api_libjingle_peerconnection_api webrtc_api_network_emulation_manager_api webrtc_api_rtc_stats_api webrtc_api_time_controller webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_task_queue_default_task_queue_factory webrtc_api_transport_field_trial_based_config webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_media_rtc_audio_video webrtc_media_rtc_media_base webrtc_modules_audio_device_audio_device_impl webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_rtc_p2p webrtc_pc_pc_test_utils webrtc_pc_rtc_pc_base webrtc_rtc_base_rtc_base webrtc_rtc_base_null_socket_server webrtc_rtc_base_stringutils webrtc_test_fake_video_codecs webrtc_test_fileutils webrtc_test_test_support webrtc_test_video_test_common webrtc_test_logging_log_writer webrtc_test_network_emulated_network webrtc_test_scenario_scenario webrtc_test_time_controller_time_controller ) +target_link_libraries(webrtc_test_peer_scenario_peer_scenario PUBLIC absl::flags absl::memory webrtc_api_create_time_controller webrtc_api_libjingle_peerconnection_api webrtc_api_network_emulation_manager_api webrtc_api_rtc_stats_api webrtc_api_time_controller webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_task_queue_default_task_queue_factory webrtc_api_transport_field_trial_based_config webrtc_api_video_codecs_builtin_video_decoder_factory webrtc_api_video_codecs_builtin_video_encoder_factory webrtc_media_rtc_audio_video webrtc_media_rtc_media_base webrtc_modules_audio_device_audio_device_impl webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_p2p_rtc_p2p webrtc_pc_pc_test_utils webrtc_pc_rtc_pc_base webrtc_pc_session_description webrtc_rtc_base_rtc_base webrtc_rtc_base_null_socket_server webrtc_rtc_base_stringutils webrtc_test_fake_video_codecs webrtc_test_fileutils webrtc_test_test_support webrtc_test_video_test_common webrtc_test_logging_log_writer webrtc_test_network_emulated_network webrtc_test_scenario_scenario webrtc_test_time_controller_time_controller ) # test:perf_test add_library(webrtc_test_perf_test ${WEBRTC_ROOT}/test/testsupport/perf_result_reporter.cc ${WEBRTC_ROOT}/test/testsupport/perf_test.cc ${WEBRTC_ROOT}/test/testsupport/perf_test_histogram_writer.cc) @@ -4549,7 +4836,7 @@ add_library(webrtc_test_rtp_test_utils ${WEBRTC_ROOT}/test/rtcp_packet_parser.cc target_include_directories(webrtc_test_rtp_test_utils PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_test_rtp_test_utils PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_test_rtp_test_utils PRIVATE -fno-exceptions) -target_link_libraries(webrtc_test_rtp_test_utils PUBLIC webrtc_api_array_view webrtc_api_rtp_parameters webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_criticalsection webrtc_rtc_base_logging webrtc_rtc_base_macromagic webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_arch ) +target_link_libraries(webrtc_test_rtp_test_utils PUBLIC absl::optional webrtc_api_array_view webrtc_api_rtp_parameters webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_rtc_base_checks webrtc_rtc_base_criticalsection webrtc_rtc_base_logging webrtc_rtc_base_macromagic webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_arch ) # test/scenario:column_printer add_library(webrtc_test_scenario_column_printer ${WEBRTC_ROOT}/test/scenario/column_printer.cc) @@ -4563,7 +4850,7 @@ add_library(webrtc_test_scenario_scenario ${WEBRTC_ROOT}/test/scenario/audio_str target_include_directories(webrtc_test_scenario_scenario PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_test_scenario_scenario PRIVATE WEBRTC_MAC WEBRTC_POSIX HAVE_WEBRTC_VIDEO WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1 WEBRTC_UNIT_TEST) target_compile_options(webrtc_test_scenario_scenario PRIVATE -fno-exceptions) -target_link_libraries(webrtc_test_scenario_scenario PUBLIC webrtc_modules_video_coding_objc_codec_factory_helper absl::flags absl::flags_parse absl::memory absl::strings absl::optional webrtc_api_create_frame_generator webrtc_api_fec_controller_api webrtc_api_frame_generator_api webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_event_log_output_file webrtc_api_rtp_parameters webrtc_api_sequence_checker webrtc_api_time_controller webrtc_api_transport_api webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_rtc_event_log_rtc_event_log webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_test_video_function_video_factory webrtc_api_transport_network_control webrtc_api_units_data_rate webrtc_api_units_data_size webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_audio_audio webrtc_call_call webrtc_call_call_interfaces webrtc_call_rtp_sender webrtc_call_simulated_network webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_audio_video webrtc_media_rtc_internal_video_codecs webrtc_media_rtc_media_base webrtc_modules_audio_coding_ana_config_proto_bridge webrtc_modules_audio_device_audio_device webrtc_modules_audio_device_audio_device_impl webrtc_modules_audio_device_mock_audio_device webrtc_modules_audio_mixer_audio_mixer_impl webrtc_modules_audio_processing_audio_processing webrtc_modules_congestion_controller_goog_cc_test_goog_cc_printer webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_mock_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_modules_video_coding_webrtc_h264 webrtc_modules_video_coding_webrtc_multiplex webrtc_modules_video_coding_webrtc_vp8 webrtc_modules_video_coding_webrtc_vp9 webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_numerics webrtc_rtc_base_rtc_stats_counters webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_socket_address webrtc_rtc_base_task_queue_for_test webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_repeating_task webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_test_fake_video_codecs webrtc_test_fileutils webrtc_test_rtp_test_utils webrtc_test_test_common webrtc_test_test_support webrtc_test_video_test_common webrtc_test_logging_log_writer webrtc_test_network_emulated_network webrtc_test_scenario_column_printer webrtc_test_time_controller_time_controller webrtc_video_video ) +target_link_libraries(webrtc_test_scenario_scenario PUBLIC webrtc_modules_video_coding_objc_codec_factory_helper absl::flags absl::flags_parse absl::memory absl::strings absl::optional webrtc_api_create_frame_generator webrtc_api_fec_controller_api webrtc_api_frame_generator_api webrtc_api_libjingle_peerconnection_api webrtc_api_rtc_event_log_output_file webrtc_api_rtp_parameters webrtc_api_sequence_checker webrtc_api_time_controller webrtc_api_transport_api webrtc_api_audio_codecs_builtin_audio_decoder_factory webrtc_api_audio_codecs_builtin_audio_encoder_factory webrtc_api_rtc_event_log_rtc_event_log webrtc_api_rtc_event_log_rtc_event_log_factory webrtc_api_test_video_function_video_factory webrtc_api_transport_network_control webrtc_api_units_data_rate webrtc_api_units_data_size webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_api_video_builtin_video_bitrate_allocator_factory webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_codecs_video_codecs_api webrtc_audio_audio webrtc_call_call webrtc_call_call_interfaces webrtc_call_rtp_sender webrtc_call_simulated_network webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_audio_video webrtc_media_rtc_internal_video_codecs webrtc_media_rtc_media_base webrtc_modules_audio_coding_ana_config_proto webrtc_modules_audio_device_audio_device webrtc_modules_audio_device_audio_device_impl webrtc_modules_audio_device_mock_audio_device webrtc_modules_audio_mixer_audio_mixer_impl webrtc_modules_audio_processing_audio_processing webrtc_modules_congestion_controller_goog_cc_test_goog_cc_printer webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_mock_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_modules_video_coding_webrtc_h264 webrtc_modules_video_coding_webrtc_multiplex webrtc_modules_video_coding_webrtc_vp8 webrtc_modules_video_coding_webrtc_vp9 webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_base_tests_utils webrtc_rtc_base_rtc_numerics webrtc_rtc_base_rtc_stats_counters webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_safe_minmax webrtc_rtc_base_socket_address webrtc_rtc_base_task_queue_for_test webrtc_rtc_base_threading webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_task_utils_repeating_task webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_test_fake_video_codecs webrtc_test_fileutils webrtc_test_rtp_test_utils webrtc_test_test_common webrtc_test_test_support webrtc_test_video_test_common webrtc_test_logging_log_writer webrtc_test_network_emulated_network webrtc_test_scenario_column_printer webrtc_test_time_controller_time_controller webrtc_video_video ) # test:test_common add_library(webrtc_test_test_common ${WEBRTC_ROOT}/test/call_test.cc ${WEBRTC_ROOT}/test/drifting_clock.cc ${WEBRTC_ROOT}/test/layer_filtering_transport.cc ${WEBRTC_ROOT}/test/run_loop.cc) @@ -4624,7 +4911,7 @@ add_library(webrtc_video_video ${WEBRTC_ROOT}/video/buffered_frame_decryptor.cc target_include_directories(webrtc_video_video PRIVATE ${WEBRTC_ROOT} ${CMAKE_CURRENT_BINARY_DIR}) target_compile_definitions(webrtc_video_video PRIVATE WEBRTC_MAC WEBRTC_POSIX WEBRTC_ABSL_MUTEX WEBRTC_ENABLE_AVX2 WEBRTC_ENABLE_PROTOBUF=1 WEBRTC_HAVE_SCTP WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1) target_compile_options(webrtc_video_video PRIVATE -fno-exceptions) -target_link_libraries(webrtc_video_video PUBLIC absl::algorithm_container absl::core_headers absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_transport_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_units_timestamp webrtc_api_video_encoded_image webrtc_api_video_recordable_encoded_frame webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_bitrate_allocator webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_video_stream_encoder webrtc_api_video_codecs_video_codecs_api webrtc_call_bitrate_allocator webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_call_rtp_sender webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_modules_module_api webrtc_modules_module_api_public webrtc_modules_pacing_pacing webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_rtp_rtcp_rtp_video_header webrtc_modules_utility_utility webrtc_modules_video_coding_video_coding webrtc_modules_video_coding_codec_globals_headers webrtc_modules_video_coding_nack_module webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_modules_video_processing_video_processing webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rate_limiter webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_weak_ptr webrtc_rtc_base_experiments_alr_experiment webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_experiments_keyframe_interval_settings_experiment webrtc_rtc_base_experiments_min_video_bitrate_experiment webrtc_rtc_base_experiments_quality_scaling_experiment webrtc_rtc_base_experiments_rate_control_settings webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_thread_registry webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_time_timestamp_extrapolator webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics webrtc_video_frame_dumping_decoder webrtc_video_video_stream_encoder_impl webrtc_video_adaptation_video_adaptation ) +target_link_libraries(webrtc_video_video PUBLIC absl::algorithm_container absl::core_headers absl::memory absl::strings absl::optional webrtc_api_array_view webrtc_api_fec_controller_api webrtc_api_frame_transformer_interface webrtc_api_libjingle_peerconnection_api webrtc_api_rtp_parameters webrtc_api_scoped_refptr webrtc_api_sequence_checker webrtc_api_transport_api webrtc_api_crypto_frame_decryptor_interface webrtc_api_crypto_options webrtc_api_rtc_event_log_rtc_event_log webrtc_api_task_queue_task_queue webrtc_api_units_time_delta webrtc_api_units_timestamp webrtc_api_video_encoded_image webrtc_api_video_recordable_encoded_frame webrtc_api_video_video_bitrate_allocation webrtc_api_video_video_bitrate_allocator webrtc_api_video_video_codec_constants webrtc_api_video_video_frame webrtc_api_video_video_rtp_headers webrtc_api_video_video_stream_encoder webrtc_api_video_codecs_video_codecs_api webrtc_call_bitrate_allocator webrtc_call_call_interfaces webrtc_call_rtp_interfaces webrtc_call_rtp_receiver webrtc_call_rtp_sender webrtc_call_video_stream_api webrtc_common_video_common_video webrtc_media_rtc_media_base webrtc_modules_module_api webrtc_modules_module_api_public webrtc_modules_pacing_pacing webrtc_modules_remote_bitrate_estimator_remote_bitrate_estimator webrtc_modules_rtp_rtcp_rtp_rtcp webrtc_modules_rtp_rtcp_rtp_rtcp_format webrtc_modules_rtp_rtcp_rtp_video_header webrtc_modules_utility_utility webrtc_modules_video_coding_video_coding webrtc_modules_video_coding_codec_globals_headers webrtc_modules_video_coding_nack_module webrtc_modules_video_coding_video_codec_interface webrtc_modules_video_coding_video_coding_utility webrtc_modules_video_processing_video_processing webrtc_rtc_base_rtc_base webrtc_rtc_base_checks webrtc_rtc_base_rate_limiter webrtc_rtc_base_rtc_base_approved webrtc_rtc_base_rtc_numerics webrtc_rtc_base_rtc_task_queue webrtc_rtc_base_stringutils webrtc_rtc_base_threading webrtc_rtc_base_weak_ptr webrtc_rtc_base_experiments_alr_experiment webrtc_rtc_base_experiments_field_trial_parser webrtc_rtc_base_experiments_keyframe_interval_settings_experiment webrtc_rtc_base_experiments_min_video_bitrate_experiment webrtc_rtc_base_experiments_quality_scaling_experiment webrtc_rtc_base_experiments_rate_control_settings webrtc_rtc_base_synchronization_mutex webrtc_rtc_base_system_no_unique_address webrtc_rtc_base_system_thread_registry webrtc_rtc_base_task_utils_pending_task_safety_flag webrtc_rtc_base_task_utils_repeating_task webrtc_rtc_base_task_utils_to_queued_task webrtc_rtc_base_time_timestamp_extrapolator webrtc_system_wrappers_system_wrappers webrtc_system_wrappers_field_trial webrtc_system_wrappers_metrics webrtc_video_frame_dumping_decoder webrtc_video_video_stream_encoder_impl webrtc_video_adaptation_video_adaptation ) # video:video_stream_encoder_impl add_library(webrtc_video_video_stream_encoder_impl ${WEBRTC_ROOT}/video/alignment_adjuster.cc ${WEBRTC_ROOT}/video/encoder_bitrate_adjuster.cc ${WEBRTC_ROOT}/video/encoder_overshoot_detector.cc ${WEBRTC_ROOT}/video/frame_encode_metadata_writer.cc ${WEBRTC_ROOT}/video/video_source_sink_controller.cc ${WEBRTC_ROOT}/video/video_stream_encoder.cc) |