mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 00:36:51 +00:00
docs: update docs
This commit is contained in:
Wim Taymans
parent
6b7ff45ca6
commit
348b7f9c21
11 changed files with 174 additions and 36 deletions
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@ -31,9 +31,20 @@ gst_rtsp_media_factory_set_shared
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gst_rtsp_media_factory_is_shared
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gst_rtsp_media_factory_set_eos_shutdown
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gst_rtsp_media_factory_is_eos_shutdown
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gst_rtsp_media_factory_set_protocols
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gst_rtsp_media_factory_get_protocols
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gst_rtsp_media_factory_set_auth
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gst_rtsp_media_factory_get_auth
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gst_rtsp_media_factory_set_buffer_size
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gst_rtsp_media_factory_get_buffer_size
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gst_rtsp_media_factory_set_multicast_group
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gst_rtsp_media_factory_get_multicast_group
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gst_rtsp_media_factory_construct
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gst_rtsp_media_factory_collect_streams
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gst_rtsp_media_factory_create_element
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<SUBSECTION Standard>
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GST_RTSP_MEDIA_FACTORY_GET_LOCK
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GST_RTSP_MEDIA_FACTORY_LOCK
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GST_RTSP_MEDIA_FACTORY_UNLOCK
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GST_RTSP_MEDIA_FACTORY_CLASS
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GST_RTSP_MEDIA_FACTORY_CAST
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GST_RTSP_MEDIA_FACTORY_CLASS_CAST
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@ -48,9 +59,6 @@ GST_RTSP_MEDIA_FACTORY_GET_CLASS
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<SECTION>
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<FILE>rtsp-media-factory-uri</FILE>
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<TITLE>GstRTSPMediaFactoryURI</TITLE>
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GST_RTSP_MEDIA_FACTORY_GET_LOCK
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GST_RTSP_MEDIA_FACTORY_LOCK
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GST_RTSP_MEDIA_FACTORY_UNLOCK
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GstRTSPMediaFactoryURI
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GstRTSPMediaFactoryURIClass
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gst_rtsp_media_factory_uri_new
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@ -72,15 +80,11 @@ gst_rtsp_media_factory_uri_get_type
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<SECTION>
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<FILE>rtsp-media</FILE>
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<TITLE>GstRTSPMedia</TITLE>
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GstRTSPMediaStream
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GstRTSPMediaStatus
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GstRTSPMedia
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GstRTSPMediaClass
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GstRTSPMediaTrans
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GstRTSPSendFunc
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GstRTSPSendListFunc
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GstRTSPKeepAliveFunc
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GstRTSPMediaStatus
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gst_rtsp_media_new
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gst_rtsp_media_set_shared
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gst_rtsp_media_is_shared
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gst_rtsp_media_set_reusable
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@ -89,18 +93,27 @@ gst_rtsp_media_set_protocols
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gst_rtsp_media_get_protocols
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gst_rtsp_media_set_eos_shutdown
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gst_rtsp_media_is_eos_shutdown
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gst_rtsp_media_set_auth
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gst_rtsp_media_get_auth
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gst_rtsp_media_set_buffer_size
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gst_rtsp_media_get_buffer_size
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gst_rtsp_media_set_multicast_group
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gst_rtsp_media_get_multicast_group
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gst_rtsp_media_get_mtu
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gst_rtsp_media_set_mtu
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gst_rtsp_media_prepare
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gst_rtsp_media_is_prepared
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gst_rtsp_media_unprepare
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gst_rtsp_media_collect_streams
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gst_rtsp_media_create_stream
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gst_rtsp_media_n_streams
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gst_rtsp_media_get_stream
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gst_rtsp_media_seek
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gst_rtsp_media_get_range_string
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gst_rtsp_media_stream_rtp
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gst_rtsp_media_stream_rtcp
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gst_rtsp_media_set_state
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gst_rtsp_media_remove_elements
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gst_rtsp_media_trans_cleanup
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<SUBSECTION Standard>
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GST_RTSP_MEDIA_CLASS
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GST_RTSP_MEDIA_CAST
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@ -123,6 +136,7 @@ gst_rtsp_server_set_address
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gst_rtsp_server_get_address
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gst_rtsp_server_set_service
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gst_rtsp_server_get_service
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gst_rtsp_server_get_bound_port
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gst_rtsp_server_set_backlog
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gst_rtsp_server_get_backlog
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gst_rtsp_server_set_session_pool
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@ -131,11 +145,15 @@ gst_rtsp_server_set_media_mapping
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gst_rtsp_server_get_media_mapping
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gst_rtsp_server_get_auth
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gst_rtsp_server_set_auth
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gst_rtsp_server_transfer_connection
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gst_rtsp_server_io_func
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gst_rtsp_server_get_io_channel
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gst_rtsp_server_create_watch
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gst_rtsp_server_create_socket
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gst_rtsp_server_create_source
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gst_rtsp_server_attach
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<SUBSECTION Standard>
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GST_RTSP_SERVER_GET_LOCK
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GST_RTSP_SERVER_LOCK
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GST_RTSP_SERVER_UNLOCK
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GST_RTSP_SERVER_CLASS
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GST_RTSP_SERVER_CAST
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GST_RTSP_SERVER_CLASS_CAST
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@ -182,8 +200,6 @@ GST_RTSP_SESSION_POOL_GET_CLASS
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<TITLE>GstRTSPSession</TITLE>
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GstRTSPSession
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GstRTSPSessionClass
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GstRTSPSessionStream
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GstRTSPSessionMedia
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gst_rtsp_session_new
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gst_rtsp_session_get_sessionid
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gst_rtsp_session_set_timeout
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@ -196,12 +212,6 @@ gst_rtsp_session_is_expired
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gst_rtsp_session_manage_media
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gst_rtsp_session_release_media
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gst_rtsp_session_get_media
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gst_rtsp_session_media_set_state
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gst_rtsp_session_media_get_stream
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gst_rtsp_session_media_alloc_channels
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gst_rtsp_session_stream_set_transport
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gst_rtsp_session_stream_set_callbacks
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gst_rtsp_session_stream_set_keepalive
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<SUBSECTION Standard>
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GST_RTSP_SESSION_CLASS
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GST_RTSP_SESSION_CAST
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@ -214,6 +224,27 @@ GST_IS_RTSP_SESSION_CLASS
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GST_RTSP_SESSION_GET_CLASS
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</SECTION>
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<SECTION>
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<FILE>rtsp-session-media</FILE>
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<TITLE>GstRTSPSessionMedia</TITLE>
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GstRTSPSessionMedia
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GstRTSPSessionMediaClass
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gst_rtsp_session_media_new
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gst_rtsp_session_media_set_state
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gst_rtsp_session_media_get_transport
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gst_rtsp_session_media_alloc_channels
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<SUBSECTION Standard>
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GST_RTSP_SESSION_MEDIA_CAST
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GST_RTSP_SESSION_MEDIA_CLASS_CAST
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GST_IS_RTSP_SESSION_MEDIA
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GST_IS_RTSP_SESSION_MEDIA_CLASS
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GST_RTSP_SESSION_MEDIA
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GST_RTSP_SESSION_MEDIA_CLASS
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GST_RTSP_SESSION_MEDIA_GET_CLASS
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GST_TYPE_RTSP_SESSION_MEDIA
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gst_rtsp_session_media_get_type
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</SECTION>
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<SECTION>
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<FILE>rtsp-auth</FILE>
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<TITLE>GstRTSPAuth</TITLE>
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@ -222,7 +253,7 @@ GstRTSPAuthClass
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gst_rtsp_auth_new
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gst_rtsp_auth_set_basic
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gst_rtsp_auth_setup_auth
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gst_rtsp_auth_check_method
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gst_rtsp_auth_check
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gst_rtsp_auth_make_basic
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<SUBSECTION Standard>
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GST_IS_RTSP_AUTH
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@ -239,6 +270,7 @@ gst_rtsp_auth_get_type
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<SECTION>
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<FILE>rtsp-client</FILE>
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<TITLE>GstRTSPClient</TITLE>
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GstRTSPClientState
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GstRTSPClient
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GstRTSPClientClass
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gst_rtsp_client_new
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@ -248,9 +280,12 @@ gst_rtsp_client_set_session_pool
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gst_rtsp_client_get_session_pool
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gst_rtsp_client_set_media_mapping
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gst_rtsp_client_get_media_mapping
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gst_rtsp_client_set_use_client_settings
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gst_rtsp_client_get_use_client_settings
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gst_rtsp_client_set_auth
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gst_rtsp_client_get_auth
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gst_rtsp_client_accept
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gst_rtsp_client_create_from_socket
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<SUBSECTION Standard>
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GST_RTSP_CLIENT_CLASS
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GST_RTSP_CLIENT_CAST
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@ -275,3 +310,53 @@ GstSDPInfo
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gst_rtsp_sdp_from_media
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</SECTION>
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<SECTION>
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<FILE>rtsp-stream</FILE>
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<TITLE>GstRTSPStream</TITLE>
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GstRTSPStream
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GstRTSPStreamClass
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gst_rtsp_stream_new
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gst_rtsp_stream_get_mtu
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gst_rtsp_stream_set_mtu
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gst_rtsp_stream_join_bin
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gst_rtsp_stream_leave_bin
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gst_rtsp_stream_get_rtpinfo
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gst_rtsp_stream_recv_rtcp
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gst_rtsp_stream_recv_rtp
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gst_rtsp_stream_add_transport
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gst_rtsp_stream_remove_transport
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<SUBSECTION Standard>
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GST_RTSP_STREAM_CAST
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GST_RTSP_STREAM_CLASS_CAST
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GST_IS_RTSP_STREAM
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GST_IS_RTSP_STREAM_CLASS
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GST_RTSP_STREAM
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GST_RTSP_STREAM_CLASS
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GST_RTSP_STREAM_GET_CLASS
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GST_TYPE_RTSP_STREAM
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gst_rtsp_stream_get_type
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</SECTION>
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<SECTION>
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<FILE>rtsp-stream-transport</FILE>
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<TITLE>GstRTSPStreamTransport</TITLE>
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GstRTSPKeepAliveFunc
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GstRTSPSendFunc
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GstRTSPStreamTransport
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GstRTSPStreamTransportClass
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gst_rtsp_stream_transport_new
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gst_rtsp_stream_transport_set_callbacks
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gst_rtsp_stream_transport_set_keepalive
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gst_rtsp_stream_transport_set_transport
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<SUBSECTION Standard>
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GST_RTSP_STREAM_TRANSPORT_CAST
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GST_RTSP_STREAM_TRANSPORT_CLASS_CAST
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GST_IS_RTSP_STREAM_TRANSPORT
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GST_IS_RTSP_STREAM_TRANSPORT_CLASS
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GST_RTSP_STREAM_TRANSPORT
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GST_RTSP_STREAM_TRANSPORT_CLASS
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GST_RTSP_STREAM_TRANSPORT_GET_CLASS
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GST_TYPE_RTSP_STREAM_TRANSPORT
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gst_rtsp_stream_transport_get_type
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</SECTION>
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@ -47,6 +47,7 @@ typedef struct _GstRTSPMediaFactoryClass GstRTSPMediaFactoryClass;
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/**
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* GstRTSPMediaFactory:
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* @parent: the parent GObject
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* @lock: mutex protecting the datastructure.
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* @launch: the launch description
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* @shared: if media from this factory can be shared between clients
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@ -65,6 +65,13 @@ gst_rtsp_media_mapping_finalize (GObject * obj)
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G_OBJECT_CLASS (gst_rtsp_media_mapping_parent_class)->finalize (obj);
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}
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/**
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* gst_rtsp_media_mapping_new:
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*
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* Make a new media mapping object.
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*
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* Returns: a new #GstRTSPMediaMapping
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*/
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GstRTSPMediaMapping *
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gst_rtsp_media_mapping_new (void)
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{
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@ -63,6 +63,7 @@ typedef enum {
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/**
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* GstRTSPMedia:
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* @parent: parent GObject
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* @lock: for protecting the object
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* @cond: for signaling the object
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* @shared: if this media can be shared between clients
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@ -70,11 +71,17 @@ typedef enum {
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* @protocols: the allowed lower transport for this stream
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* @reused: if this media has been reused
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* @is_ipv6: if this media is using ipv6
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* @eos_shutdown: if EOS should be sent on shutdown
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* @buffer_size: The UDP buffer size
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* @auth: the authentication service in use
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* @multicast_group: the multicast group to use
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* @mtu: the MTU of the payloaders
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* @element: the data providing element
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* @streams: the different #GstRTSPStream provided by @element
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* @dynamic: list of dynamic elements managed by @element
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* @status: the status of the media pipeline
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* @n_active: the number of active connections
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* @adding: when elements are added to the pipeline
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* @pipeline: the toplevel pipeline
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* @fakesink: for making state changes async
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* @source: the bus watch for pipeline messages.
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@ -140,8 +147,7 @@ struct _GstRTSPMedia {
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* @thread: the thread dispatching messages.
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* @handle_message: handle a message
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* @unprepare: the default implementation sets the pipeline's state
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* to GST_STATE_NULL.
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* @handle_mtu: handle a mtu
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* to GST_STATE_NULL and removes all elements.
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*
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* The RTSP media class
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*/
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@ -196,7 +202,6 @@ guint gst_rtsp_media_get_mtu (GstRTSPMedia *media);
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/* prepare the media for playback */
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gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
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gboolean gst_rtsp_media_is_prepared (GstRTSPMedia *media);
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gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
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/* creating streams */
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@ -20,6 +20,15 @@
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#include "rtsp-params.h"
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/**
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* gst_rtsp_params_set:
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* @client: a #GstRTSPClient
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* @state: a #GstRTSPClientState
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*
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* Set parameters (not implemented yet)
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*
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* Returns: a #GstRTSPResult
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*/
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GstRTSPResult
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gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
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{
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@ -35,6 +44,15 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
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return GST_RTSP_OK;
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}
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/**
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* gst_rtsp_params_get:
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* @client: a #GstRTSPClient
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* @state: a #GstRTSPClientState
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*
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* Get parameters (not implemented yet)
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*
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* Returns: a #GstRTSPResult
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*/
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GstRTSPResult
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gst_rtsp_params_get (GstRTSPClient * client, GstRTSPClientState * state)
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{
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@ -255,6 +255,14 @@ gst_rtsp_server_get_address (GstRTSPServer * server)
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return result;
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}
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/**
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* gst_rtsp_server_get_bound_port:
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* @server: a #GstRTSPServer
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*
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* Get the port number where the server was bound to.
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*
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* Returns: the port number
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*/
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int
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gst_rtsp_server_get_bound_port (GstRTSPServer * server)
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{
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@ -877,6 +885,7 @@ transfer_failed:
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* gst_rtsp_server_io_func:
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* @socket: a #GSocket
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* @condition: the condition on @source
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* @server: a #GstRTSPServer
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*
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* A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
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* new connection on @socket or @server.
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@ -43,7 +43,7 @@ typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
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* GstRTSPSessionPool:
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* @max_sessions: the maximum number of sessions.
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* @lock: locking the session hashtable
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* @session: hashtable of sessions indexed by the session id.
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* @sessions: hashtable of sessions indexed by the session id.
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*
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* An object that keeps track of the active sessions. This object is usually
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* attached to a #GstRTSPServer object to manage the sessions in that server.
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@ -223,8 +223,9 @@ gst_rtsp_session_get_media (GstRTSPSession * sess, const GstRTSPUrl * url)
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/**
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* gst_rtsp_session_new:
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* @sessionid: a session id
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*
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* Create a new #GstRTSPSession instance.
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* Create a new #GstRTSPSession instance with @sessionid.
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*/
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GstRTSPSession *
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gst_rtsp_session_new (const gchar * sessionid)
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@ -301,6 +302,12 @@ gst_rtsp_session_touch (GstRTSPSession * session)
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g_get_current_time (&session->last_access);
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}
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/**
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* gst_rtsp_session_prevent_expire:
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* @session: a #GstRTSPSession
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*
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* Prevent @session from expiring.
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*/
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void
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gst_rtsp_session_prevent_expire (GstRTSPSession * session)
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{
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@ -309,6 +316,13 @@ gst_rtsp_session_prevent_expire (GstRTSPSession * session)
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g_atomic_int_add (&session->expire_count, 1);
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}
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/**
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* gst_rtsp_session_allow_expire:
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* @session: a #GstRTSPSession
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*
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* Allow @session to expire. This method must be called an equal
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* amount of time as gst_rtsp_session_prevent_expire().
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*/
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||||
void
|
||||
gst_rtsp_session_allow_expire (GstRTSPSession * session)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -43,12 +43,13 @@ typedef struct _GstRTSPSessionClass GstRTSPSessionClass;
|
|||
|
||||
/**
|
||||
* GstRTSPSession:
|
||||
* @parent: the parent GObject
|
||||
* @sessionid: the session id of the session
|
||||
* @timeout: the timeout of the session
|
||||
* @create_time: the time when the session was created
|
||||
* @last_access: the time the session was last accessed
|
||||
* @expire_count: the expire prevention counter
|
||||
* @media: a list of #GstRTSPSessionMedia managed in this session
|
||||
* @medias: a list of #GstRTSPSessionMedia managed in this session
|
||||
*
|
||||
* Session information kept by the server for a specific client.
|
||||
* One client session, identified with a session id, can handle multiple medias
|
||||
|
|
|
|||
|
|
@ -47,11 +47,9 @@ typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
|
|||
/**
|
||||
* GstRTSPStreamTransport:
|
||||
* @parent: parent instance
|
||||
* @idx: the stream index
|
||||
* @stream: the GstRTSPStream we manage
|
||||
* @send_rtp: callback for sending RTP messages
|
||||
* @send_rtcp: callback for sending RTCP messages
|
||||
* @send_rtp_list: callback for sending RTP messages
|
||||
* @send_rtcp_list: callback for sending RTCP messages
|
||||
* @user_data: user data passed in the callbacks
|
||||
* @notify: free function for the user_data.
|
||||
* @keep_alive: keep alive callback
|
||||
|
|
|
|||
|
|
@ -62,7 +62,7 @@ typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
|
|||
* @caps_sig: the signal id for detecting caps
|
||||
* @caps: the caps of the stream
|
||||
* @n_active: the number of active transports in @transports
|
||||
* @tranports: list of #GstStreamTransport being streamed to
|
||||
* @transports: list of #GstStreamTransport being streamed to
|
||||
*
|
||||
* The definition of a media stream. The streams are identified by @id.
|
||||
*/
|
||||
|
|
|
|||
Loading…
Reference in a new issue