diff --git a/pc/BUILD.gn b/pc/BUILD.gn index e351748485..baab6f715e 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -16,6 +16,7 @@ # - rtc_pc # - session_description # - simulcast_description +# - peerconnection # - sdp_utils # - media_stream_observer # - video_track_source @@ -735,6 +736,142 @@ rtc_library("media_protocol_names") { absl_deps = [ "//third_party/abseil-cpp/absl/strings" ] } +rtc_source_set("peerconnection") { + # TODO(bugs.webrtc.org/13661): Reduce visibility if possible + visibility = [ "*" ] # Used by Chromium and others + allow_poison = [ "environment_construction" ] + cflags = [] + sources = [] + + deps = [ + ":audio_rtp_receiver", + ":audio_track", + ":connection_context", + ":data_channel_controller", + ":data_channel_utils", + ":dtmf_sender", + ":ice_server_parsing", + ":jitter_buffer_delay", + ":jsep_ice_candidate", + ":jsep_session_description", + ":legacy_stats_collector", + ":legacy_stats_collector_interface", + ":local_audio_source", + ":media_protocol_names", + ":media_stream", + ":media_stream_observer", + ":peer_connection", + ":peer_connection_factory", + ":peer_connection_internal", + ":peer_connection_message_handler", + ":proxy", + ":remote_audio_source", + ":rtc_stats_collector", + ":rtc_stats_traversal", + ":rtp_parameters_conversion", + ":rtp_receiver", + ":rtp_sender", + ":rtp_transceiver", + ":rtp_transmission_manager", + ":sctp_data_channel", + ":sdp_offer_answer", + ":sdp_state_provider", + ":sdp_utils", + ":session_description", + ":simulcast_description", + ":simulcast_sdp_serializer", + ":stream_collection", + ":track_media_info_map", + ":transceiver_list", + ":usage_pattern", + ":video_rtp_receiver", + ":video_track", + ":video_track_source", + ":webrtc_sdp", + ":webrtc_session_description_factory", + "../api:array_view", + "../api:async_dns_resolver", + "../api:audio_options_api", + "../api:call_api", + "../api:fec_controller_api", + "../api:field_trials_view", + "../api:frame_transformer_interface", + "../api:ice_transport_factory", + "../api:libjingle_logging_api", + "../api:libjingle_peerconnection_api", + "../api:media_stream_interface", + "../api:network_state_predictor_api", + "../api:packet_socket_factory", + "../api:priority", + "../api:rtc_error", + "../api:rtc_event_log_output_file", + "../api:rtc_stats_api", + "../api:rtp_parameters", + "../api:rtp_transceiver_direction", + "../api:scoped_refptr", + "../api:sequence_checker", + "../api/adaptation:resource_adaptation_api", + "../api/audio_codecs:audio_codecs_api", + "../api/crypto:frame_decryptor_interface", + "../api/crypto:options", + "../api/neteq:neteq_api", + "../api/rtc_event_log", + "../api/task_queue", + "../api/task_queue:pending_task_safety_flag", + "../api/transport:bitrate_settings", + "../api/transport:datagram_transport_interface", + "../api/transport:enums", + "../api/transport:field_trial_based_config", + "../api/transport:network_control", + "../api/transport:sctp_transport_factory_interface", + "../api/units:data_rate", + "../api/video:builtin_video_bitrate_allocator_factory", + "../api/video:video_bitrate_allocator_factory", + "../api/video:video_codec_constants", + "../api/video:video_frame", + "../api/video:video_rtp_headers", + "../api/video_codecs:video_codecs_api", + "../call:call_interfaces", + "../call:rtp_interfaces", + "../call:rtp_sender", + "../common_video", + "../logging:ice_log", + "../media:rtc_data_sctp_transport_internal", + "../media:rtc_media_base", + "../media:rtc_media_config", + "../modules/audio_processing:audio_processing_statistics", + "../modules/rtp_rtcp:rtp_rtcp_format", + "../p2p:rtc_p2p", + "../rtc_base:callback_list", + "../rtc_base:checks", + "../rtc_base:ip_address", + "../rtc_base:network_constants", + "../rtc_base:rtc_operations_chain", + "../rtc_base:safe_minmax", + "../rtc_base:socket_address", + "../rtc_base:threading", + "../rtc_base:weak_ptr", + "../rtc_base/experiments:field_trial_parser", + "../rtc_base/network:sent_packet", + "../rtc_base/synchronization:mutex", + "../rtc_base/system:file_wrapper", + "../rtc_base/system:no_unique_address", + "../rtc_base/system:rtc_export", + "../rtc_base/system:unused", + "../rtc_base/third_party/base64", + "../rtc_base/third_party/sigslot", + "../stats", + "../system_wrappers", + "../system_wrappers:field_trial", + "../system_wrappers:metrics", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/algorithm:container", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + rtc_library("sctp_data_channel") { visibility = [ ":*" ] sources = [ @@ -1982,6 +2119,7 @@ if (rtc_include_tests && !build_with_chromium) { ":media_protocol_names", ":media_session", ":pc_test_utils", + ":peerconnection", ":rtc_pc", ":rtcp_mux_filter", ":rtp_media_utils", @@ -2082,6 +2220,7 @@ if (rtc_include_tests && !build_with_chromium) { deps = [ ":pc_test_utils", ":peer_connection", + ":peerconnection", ":peerconnection_wrapper", "../api:audio_options_api", "../api:create_peerconnection_factory", @@ -2134,6 +2273,7 @@ if (rtc_include_tests && !build_with_chromium) { ] deps = [ ":pc_test_utils", + ":peerconnection", ":sdp_utils", "../api:function_view", "../api:libjingle_peerconnection_api", @@ -2491,6 +2631,7 @@ if (rtc_include_tests && !build_with_chromium) { ":peer_connection", ":peer_connection_factory", ":peer_connection_proxy", + ":peerconnection", ":remote_audio_source", ":rtp_media_utils", ":rtp_parameters_conversion", @@ -2649,6 +2790,7 @@ if (rtc_include_tests && !build_with_chromium) { ":jitter_buffer_delay", ":libjingle_peerconnection", ":peer_connection_internal", + ":peerconnection", ":rtp_receiver", ":rtp_sender", ":sctp_data_channel", diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 1f672662b7..88fd1e712c 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -828,6 +828,7 @@ if (current_os == "linux" || is_android) { ":generated_metrics_jni", ":native_api_jni", ":peerconnection_jni", + "../../pc:peerconnection", "../../rtc_base:stringutils", "../../system_wrappers:metrics", ]