From: Wim Taymans Date: Fri, 26 Oct 2012 10:33:21 +0000 (+0200) Subject: docs: update docs X-Git-Tag: 1.6.0~721 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=348b7f9c21f63cb9f4d895f62034884ba33bd648;p=platform%2Fupstream%2Fgst-rtsp-server.git docs: update docs --- diff --git a/docs/libs/gst-rtsp-server-sections.txt b/docs/libs/gst-rtsp-server-sections.txt index 9ac66c1..561f664 100644 --- a/docs/libs/gst-rtsp-server-sections.txt +++ b/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 +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
rtsp-media-factory-uri GstRTSPMediaFactoryURI -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
rtsp-media GstRTSPMedia -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 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 +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 GstRTSPSession 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 GST_RTSP_SESSION_CLASS GST_RTSP_SESSION_CAST @@ -215,6 +225,27 @@ GST_RTSP_SESSION_GET_CLASS
+rtsp-session-media +GstRTSPSessionMedia +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 + +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 +
+ +
rtsp-auth GstRTSPAuth GstRTSPAuth @@ -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 GST_IS_RTSP_AUTH @@ -239,6 +270,7 @@ gst_rtsp_auth_get_type
rtsp-client GstRTSPClient +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 GST_RTSP_CLIENT_CLASS GST_RTSP_CLIENT_CAST @@ -275,3 +310,53 @@ GstSDPInfo gst_rtsp_sdp_from_media
+
+rtsp-stream +GstRTSPStream +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 + +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 +
+ +
+rtsp-stream-transport +GstRTSPStreamTransport +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 + +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 +
+ diff --git a/gst/rtsp-server/rtsp-media-factory.h b/gst/rtsp-server/rtsp-media-factory.h index d20e502..7a80fe0 100644 --- a/gst/rtsp-server/rtsp-media-factory.h +++ b/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 diff --git a/gst/rtsp-server/rtsp-media-mapping.c b/gst/rtsp-server/rtsp-media-mapping.c index 8a28db9..6344394 100644 --- a/gst/rtsp-server/rtsp-media-mapping.c +++ b/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) { diff --git a/gst/rtsp-server/rtsp-media.h b/gst/rtsp-server/rtsp-media.h index 88c5d81..b7ba346 100644 --- a/gst/rtsp-server/rtsp-media.h +++ b/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. - * @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 */ diff --git a/gst/rtsp-server/rtsp-params.c b/gst/rtsp-server/rtsp-params.c index 95614fd..a5d46b3 100644 --- a/gst/rtsp-server/rtsp-params.c +++ b/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) { diff --git a/gst/rtsp-server/rtsp-server.c b/gst/rtsp-server/rtsp-server.c index c622981..1b4bbf6 100644 --- a/gst/rtsp-server/rtsp-server.c +++ b/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. diff --git a/gst/rtsp-server/rtsp-session-pool.h b/gst/rtsp-server/rtsp-session-pool.h index 642242f..645dab3 100644 --- a/gst/rtsp-server/rtsp-session-pool.h +++ b/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. diff --git a/gst/rtsp-server/rtsp-session.c b/gst/rtsp-server/rtsp-session.c index 7b8cac0..58abc4b 100644 --- a/gst/rtsp-server/rtsp-session.c +++ b/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) { diff --git a/gst/rtsp-server/rtsp-session.h b/gst/rtsp-server/rtsp-session.h index f7815c9..b6236dc 100644 --- a/gst/rtsp-server/rtsp-session.h +++ b/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 diff --git a/gst/rtsp-server/rtsp-stream-transport.h b/gst/rtsp-server/rtsp-stream-transport.h index cd17332..fe39b98 100644 --- a/gst/rtsp-server/rtsp-stream-transport.h +++ b/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 diff --git a/gst/rtsp-server/rtsp-stream.h b/gst/rtsp-server/rtsp-stream.h index 9840313..db923d0 100644 --- a/gst/rtsp-server/rtsp-stream.h +++ b/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. */