docs: update docs

docs/libs/gst-rtsp-server-sections.txt

@@ -31,9 +31,20 @@ gst_rtsp_media_factory_set_shared
 gst_rtsp_media_factory_is_shared
 gst_rtsp_media_factory_set_eos_shutdown
 gst_rtsp_media_factory_is_eos_shutdown
+gst_rtsp_media_factory_set_protocols
+gst_rtsp_media_factory_get_protocols
+gst_rtsp_media_factory_set_auth
+gst_rtsp_media_factory_get_auth
+gst_rtsp_media_factory_set_buffer_size
+gst_rtsp_media_factory_get_buffer_size
+gst_rtsp_media_factory_set_multicast_group
+gst_rtsp_media_factory_get_multicast_group
 gst_rtsp_media_factory_construct
-gst_rtsp_media_factory_collect_streams
+gst_rtsp_media_factory_create_element
 <SUBSECTION Standard>
+GST_RTSP_MEDIA_FACTORY_GET_LOCK
+GST_RTSP_MEDIA_FACTORY_LOCK
+GST_RTSP_MEDIA_FACTORY_UNLOCK
 GST_RTSP_MEDIA_FACTORY_CLASS
 GST_RTSP_MEDIA_FACTORY_CAST
 GST_RTSP_MEDIA_FACTORY_CLASS_CAST
@@ -48,9 +59,6 @@ GST_RTSP_MEDIA_FACTORY_GET_CLASS
 <SECTION>
 <FILE>rtsp-media-factory-uri</FILE>
 <TITLE>GstRTSPMediaFactoryURI</TITLE>
-GST_RTSP_MEDIA_FACTORY_GET_LOCK
-GST_RTSP_MEDIA_FACTORY_LOCK
-GST_RTSP_MEDIA_FACTORY_UNLOCK
 GstRTSPMediaFactoryURI
 GstRTSPMediaFactoryURIClass
 gst_rtsp_media_factory_uri_new
@@ -72,15 +80,11 @@ gst_rtsp_media_factory_uri_get_type
 <SECTION>
 <FILE>rtsp-media</FILE>
 <TITLE>GstRTSPMedia</TITLE>
-GstRTSPMediaStream
+GstRTSPMediaStatus
 GstRTSPMedia
 GstRTSPMediaClass
-GstRTSPMediaTrans
-GstRTSPSendFunc
-GstRTSPSendListFunc
-GstRTSPKeepAliveFunc
-GstRTSPMediaStatus
 gst_rtsp_media_new
+
 gst_rtsp_media_set_shared
 gst_rtsp_media_is_shared
 gst_rtsp_media_set_reusable
@@ -89,18 +93,27 @@ gst_rtsp_media_set_protocols
 gst_rtsp_media_get_protocols
 gst_rtsp_media_set_eos_shutdown
 gst_rtsp_media_is_eos_shutdown
+gst_rtsp_media_set_auth
+gst_rtsp_media_get_auth
+gst_rtsp_media_set_buffer_size
+gst_rtsp_media_get_buffer_size
+gst_rtsp_media_set_multicast_group
+gst_rtsp_media_get_multicast_group
+gst_rtsp_media_get_mtu
+gst_rtsp_media_set_mtu
+
 gst_rtsp_media_prepare
-gst_rtsp_media_is_prepared
 gst_rtsp_media_unprepare
+
+gst_rtsp_media_collect_streams
+gst_rtsp_media_create_stream
+
 gst_rtsp_media_n_streams
 gst_rtsp_media_get_stream
+
 gst_rtsp_media_seek
 gst_rtsp_media_get_range_string
-gst_rtsp_media_stream_rtp
-gst_rtsp_media_stream_rtcp
 gst_rtsp_media_set_state
-gst_rtsp_media_remove_elements
-gst_rtsp_media_trans_cleanup
 <SUBSECTION Standard>
 GST_RTSP_MEDIA_CLASS
 GST_RTSP_MEDIA_CAST
@@ -123,6 +136,7 @@ gst_rtsp_server_set_address
 gst_rtsp_server_get_address
 gst_rtsp_server_set_service
 gst_rtsp_server_get_service
+gst_rtsp_server_get_bound_port
 gst_rtsp_server_set_backlog
 gst_rtsp_server_get_backlog
 gst_rtsp_server_set_session_pool
@@ -131,11 +145,15 @@ gst_rtsp_server_set_media_mapping
 gst_rtsp_server_get_media_mapping
 gst_rtsp_server_get_auth
 gst_rtsp_server_set_auth
+gst_rtsp_server_transfer_connection
 gst_rtsp_server_io_func
-gst_rtsp_server_get_io_channel
+gst_rtsp_server_create_socket
-gst_rtsp_server_create_watch
+gst_rtsp_server_create_source
 gst_rtsp_server_attach
 <SUBSECTION Standard>
+GST_RTSP_SERVER_GET_LOCK
+GST_RTSP_SERVER_LOCK
+GST_RTSP_SERVER_UNLOCK
 GST_RTSP_SERVER_CLASS
 GST_RTSP_SERVER_CAST
 GST_RTSP_SERVER_CLASS_CAST
@@ -182,8 +200,6 @@ GST_RTSP_SESSION_POOL_GET_CLASS
 <TITLE>GstRTSPSession</TITLE>
 GstRTSPSession
 GstRTSPSessionClass
-GstRTSPSessionStream
-GstRTSPSessionMedia
 gst_rtsp_session_new
 gst_rtsp_session_get_sessionid
 gst_rtsp_session_set_timeout
@@ -196,12 +212,6 @@ gst_rtsp_session_is_expired
 gst_rtsp_session_manage_media
 gst_rtsp_session_release_media
 gst_rtsp_session_get_media
-gst_rtsp_session_media_set_state
-gst_rtsp_session_media_get_stream
-gst_rtsp_session_media_alloc_channels
-gst_rtsp_session_stream_set_transport
-gst_rtsp_session_stream_set_callbacks
-gst_rtsp_session_stream_set_keepalive
 <SUBSECTION Standard>
 GST_RTSP_SESSION_CLASS
 GST_RTSP_SESSION_CAST
@@ -214,6 +224,27 @@ GST_IS_RTSP_SESSION_CLASS
 GST_RTSP_SESSION_GET_CLASS
 </SECTION>
 
+<SECTION>
+<FILE>rtsp-session-media</FILE>
+<TITLE>GstRTSPSessionMedia</TITLE>
+GstRTSPSessionMedia
+GstRTSPSessionMediaClass
+gst_rtsp_session_media_new
+gst_rtsp_session_media_set_state
+gst_rtsp_session_media_get_transport
+gst_rtsp_session_media_alloc_channels
+<SUBSECTION Standard>
+GST_RTSP_SESSION_MEDIA_CAST
+GST_RTSP_SESSION_MEDIA_CLASS_CAST
+GST_IS_RTSP_SESSION_MEDIA
+GST_IS_RTSP_SESSION_MEDIA_CLASS
+GST_RTSP_SESSION_MEDIA
+GST_RTSP_SESSION_MEDIA_CLASS
+GST_RTSP_SESSION_MEDIA_GET_CLASS
+GST_TYPE_RTSP_SESSION_MEDIA
+gst_rtsp_session_media_get_type
+</SECTION>
+
 <SECTION>
 <FILE>rtsp-auth</FILE>
 <TITLE>GstRTSPAuth</TITLE>
@@ -222,7 +253,7 @@ GstRTSPAuthClass
 gst_rtsp_auth_new
 gst_rtsp_auth_set_basic
 gst_rtsp_auth_setup_auth
-gst_rtsp_auth_check_method
+gst_rtsp_auth_check
 gst_rtsp_auth_make_basic
 <SUBSECTION Standard>
 GST_IS_RTSP_AUTH
@@ -239,6 +270,7 @@ gst_rtsp_auth_get_type
 <SECTION>
 <FILE>rtsp-client</FILE>
 <TITLE>GstRTSPClient</TITLE>
+GstRTSPClientState
 GstRTSPClient
 GstRTSPClientClass
 gst_rtsp_client_new
@@ -248,9 +280,12 @@ gst_rtsp_client_set_session_pool
 gst_rtsp_client_get_session_pool
 gst_rtsp_client_set_media_mapping
 gst_rtsp_client_get_media_mapping
+gst_rtsp_client_set_use_client_settings
+gst_rtsp_client_get_use_client_settings
 gst_rtsp_client_set_auth
 gst_rtsp_client_get_auth
 gst_rtsp_client_accept
+gst_rtsp_client_create_from_socket
 <SUBSECTION Standard>
 GST_RTSP_CLIENT_CLASS
 GST_RTSP_CLIENT_CAST
@@ -275,3 +310,53 @@ GstSDPInfo
 gst_rtsp_sdp_from_media
 </SECTION>
 
+<SECTION>
+<FILE>rtsp-stream</FILE>
+<TITLE>GstRTSPStream</TITLE>
+GstRTSPStream
+GstRTSPStreamClass
+gst_rtsp_stream_new
+gst_rtsp_stream_get_mtu
+gst_rtsp_stream_set_mtu
+gst_rtsp_stream_join_bin
+gst_rtsp_stream_leave_bin
+gst_rtsp_stream_get_rtpinfo
+gst_rtsp_stream_recv_rtcp
+gst_rtsp_stream_recv_rtp
+gst_rtsp_stream_add_transport
+gst_rtsp_stream_remove_transport
+<SUBSECTION Standard>
+GST_RTSP_STREAM_CAST
+GST_RTSP_STREAM_CLASS_CAST
+GST_IS_RTSP_STREAM
+GST_IS_RTSP_STREAM_CLASS
+GST_RTSP_STREAM
+GST_RTSP_STREAM_CLASS
+GST_RTSP_STREAM_GET_CLASS
+GST_TYPE_RTSP_STREAM
+gst_rtsp_stream_get_type
+</SECTION>
+
+<SECTION>
+<FILE>rtsp-stream-transport</FILE>
+<TITLE>GstRTSPStreamTransport</TITLE>
+GstRTSPKeepAliveFunc
+GstRTSPSendFunc
+GstRTSPStreamTransport
+GstRTSPStreamTransportClass
+gst_rtsp_stream_transport_new
+gst_rtsp_stream_transport_set_callbacks
+gst_rtsp_stream_transport_set_keepalive
+gst_rtsp_stream_transport_set_transport
+<SUBSECTION Standard>
+GST_RTSP_STREAM_TRANSPORT_CAST
+GST_RTSP_STREAM_TRANSPORT_CLASS_CAST
+GST_IS_RTSP_STREAM_TRANSPORT
+GST_IS_RTSP_STREAM_TRANSPORT_CLASS
+GST_RTSP_STREAM_TRANSPORT
+GST_RTSP_STREAM_TRANSPORT_CLASS
+GST_RTSP_STREAM_TRANSPORT_GET_CLASS
+GST_TYPE_RTSP_STREAM_TRANSPORT
+gst_rtsp_stream_transport_get_type
+</SECTION>
+

gst/rtsp-server/rtsp-media-factory.h

@@ -47,6 +47,7 @@ typedef struct _GstRTSPMediaFactoryClass GstRTSPMediaFactoryClass;
 
 /**
  * GstRTSPMediaFactory:
+ * @parent: the parent GObject
  * @lock: mutex protecting the datastructure.
  * @launch: the launch description
  * @shared: if media from this factory can be shared between clients

gst/rtsp-server/rtsp-media-mapping.c

@@ -65,6 +65,13 @@ gst_rtsp_media_mapping_finalize (GObject * obj)
   G_OBJECT_CLASS (gst_rtsp_media_mapping_parent_class)->finalize (obj);
 }
 
+/**
+ * gst_rtsp_media_mapping_new:
+ *
+ * Make a new media mapping object.
+ *
+ * Returns: a new #GstRTSPMediaMapping
+ */
 GstRTSPMediaMapping *
 gst_rtsp_media_mapping_new (void)
 {

gst/rtsp-server/rtsp-media.h

@@ -63,6 +63,7 @@ typedef enum {
 
 /**
  * GstRTSPMedia:
+ * @parent: parent GObject
  * @lock: for protecting the object
  * @cond: for signaling the object
  * @shared: if this media can be shared between clients
@@ -70,11 +71,17 @@ typedef enum {
  * @protocols: the allowed lower transport for this stream
  * @reused: if this media has been reused
  * @is_ipv6: if this media is using ipv6
+ * @eos_shutdown: if EOS should be sent on shutdown
+ * @buffer_size: The UDP buffer size
+ * @auth: the authentication service in use
+ * @multicast_group: the multicast group to use
+ * @mtu: the MTU of the payloaders
  * @element: the data providing element
  * @streams: the different #GstRTSPStream provided by @element
  * @dynamic: list of dynamic elements managed by @element
  * @status: the status of the media pipeline
  * @n_active: the number of active connections
+ * @adding: when elements are added to the pipeline
  * @pipeline: the toplevel pipeline
  * @fakesink: for making state changes async
  * @source: the bus watch for pipeline messages.
@@ -140,8 +147,7 @@ struct _GstRTSPMedia {
  * @thread: the thread dispatching messages.
  * @handle_message: handle a message
  * @unprepare: the default implementation sets the pipeline's state
- *             to GST_STATE_NULL.
+ *             to GST_STATE_NULL and removes all elements.
- * @handle_mtu: handle a mtu
  *
  * The RTSP media class
  */
@@ -196,7 +202,6 @@ guint                 gst_rtsp_media_get_mtu          (GstRTSPMedia *media);
 
 /* prepare the media for playback */
 gboolean              gst_rtsp_media_prepare          (GstRTSPMedia *media);
-gboolean              gst_rtsp_media_is_prepared      (GstRTSPMedia *media);
 gboolean              gst_rtsp_media_unprepare        (GstRTSPMedia *media);
 
 /* creating streams */

gst/rtsp-server/rtsp-params.c

@@ -20,6 +20,15 @@
 
 #include "rtsp-params.h"
 
+/**
+ * gst_rtsp_params_set:
+ * @client: a #GstRTSPClient
+ * @state: a #GstRTSPClientState
+ *
+ * Set parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
 GstRTSPResult
 gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
 {
@@ -35,6 +44,15 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
   return GST_RTSP_OK;
 }
 
+/**
+ * gst_rtsp_params_get:
+ * @client: a #GstRTSPClient
+ * @state: a #GstRTSPClientState
+ *
+ * Get parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
 GstRTSPResult
 gst_rtsp_params_get (GstRTSPClient * client, GstRTSPClientState * state)
 {

gst/rtsp-server/rtsp-server.c

@@ -255,6 +255,14 @@ gst_rtsp_server_get_address (GstRTSPServer * server)
   return result;
 }
 
+/**
+ * gst_rtsp_server_get_bound_port:
+ * @server: a #GstRTSPServer
+ *
+ * Get the port number where the server was bound to.
+ *
+ * Returns: the port number
+ */
 int
 gst_rtsp_server_get_bound_port (GstRTSPServer * server)
 {
@@ -877,6 +885,7 @@ transfer_failed:
  * gst_rtsp_server_io_func:
  * @socket: a #GSocket
  * @condition: the condition on @source
+ * @server: a #GstRTSPServer
  *
  * A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
  * new connection on @socket or @server.

gst/rtsp-server/rtsp-session-pool.h

@@ -43,7 +43,7 @@ typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
  * GstRTSPSessionPool:
  * @max_sessions: the maximum number of sessions.
  * @lock: locking the session hashtable
- * @session: hashtable of sessions indexed by the session id.
+ * @sessions: hashtable of sessions indexed by the session id.
  *
  * An object that keeps track of the active sessions. This object is usually
  * attached to a #GstRTSPServer object to manage the sessions in that server.

gst/rtsp-server/rtsp-session.c

@@ -223,8 +223,9 @@ gst_rtsp_session_get_media (GstRTSPSession * sess, const GstRTSPUrl * url)
 
 /**
  * gst_rtsp_session_new:
+ * @sessionid: a session id
  *
- * Create a new #GstRTSPSession instance.
+ * Create a new #GstRTSPSession instance with @sessionid.
  */
 GstRTSPSession *
 gst_rtsp_session_new (const gchar * sessionid)
@@ -301,6 +302,12 @@ gst_rtsp_session_touch (GstRTSPSession * session)
   g_get_current_time (&session->last_access);
 }
 
+/**
+ * gst_rtsp_session_prevent_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Prevent @session from expiring.
+ */
 void
 gst_rtsp_session_prevent_expire (GstRTSPSession * session)
 {
@@ -309,6 +316,13 @@ gst_rtsp_session_prevent_expire (GstRTSPSession * session)
   g_atomic_int_add (&session->expire_count, 1);
 }
 
+/**
+ * gst_rtsp_session_allow_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Allow @session to expire. This method must be called an equal
+ * amount of time as gst_rtsp_session_prevent_expire().
+ */
 void
 gst_rtsp_session_allow_expire (GstRTSPSession * session)
 {

gst/rtsp-server/rtsp-session.h

@@ -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

gst/rtsp-server/rtsp-stream-transport.h

@@ -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

gst/rtsp-server/rtsp-stream.h

@@ -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.
  */