mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-03 01:02:19 +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
|
|
@ -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_watch
|
||||
gst_rtsp_server_create_socket
|
||||
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>
|
||||
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
* @handle_mtu: handle a mtu
|
||||
* to GST_STATE_NULL and removes all elements.
|
||||
*
|
||||
* 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 */
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -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