/* GStreamer * Copyright (C) 2008 Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #include #include #ifndef __GST_RTSP_CLIENT_H__ #define __GST_RTSP_CLIENT_H__ G_BEGIN_DECLS typedef struct _GstRTSPClient GstRTSPClient; typedef struct _GstRTSPClientClass GstRTSPClientClass; typedef struct _GstRTSPClientState GstRTSPClientState; #include "rtsp-server.h" #include "rtsp-media.h" #include "rtsp-media-mapping.h" #include "rtsp-session-pool.h" #include "rtsp-session-media.h" #include "rtsp-auth.h" #include "rtsp-sdp.h" #define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ()) #define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT)) #define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT)) #define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass)) #define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient)) #define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass)) #define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj)) #define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass)) /** * GstRTSPClientState: * @request: the complete request * @uri: the complete url parsed from @request * @method: the parsed method of @uri * @session: the session, can be NULL * @sessmedia: the session media for the url can be NULL * @factory: the media factory for the url, can be NULL. * @media: the media for the url can be NULL * @stream: the stream for the url can be NULL * @response: the response * * Information passed around containing the client state of a request. */ struct _GstRTSPClientState { GstRTSPMessage *request; GstRTSPUrl *uri; GstRTSPMethod method; GstRTSPSession *session; GstRTSPSessionMedia *sessmedia; GstRTSPMediaFactory *factory; GstRTSPMedia *media; GstRTSPStream *stream; GstRTSPMessage *response; }; /** * GstRTSPClient: * * @connection: the connection object handling the client request. * @watch: watch for the connection * @ip: ip address used by the client to connect to us * @use_client_settings: whether to allow client transport settings for multicast * @session_pool: handle to the session pool used by the client. * @media_mapping: handle to the media mapping used by the client. * @uri: cached uri * @media: cached media * @transports: a list of #GstRTSPStreamTransport using @connection. * @sessions: a list of sessions managed by @connection. * * The client structure. */ struct _GstRTSPClient { GObject parent; GstRTSPConnection *connection; GstRTSPWatch *watch; gchar *server_ip; gboolean is_ipv6; gboolean use_client_settings; GstRTSPServer *server; GstRTSPSessionPool *session_pool; GstRTSPMediaMapping *media_mapping; GstRTSPAuth *auth; GstRTSPUrl *uri; GstRTSPMedia *media; GList *transports; GList *sessions; }; struct _GstRTSPClientClass { GObjectClass parent_class; GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media); /* signals */ void (*closed) (GstRTSPClient *client); void (*new_session) (GstRTSPClient *client, GstRTSPSession *session); void (*options_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*describe_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*setup_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*play_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*pause_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*teardown_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*set_parameter_request) (GstRTSPClient *client, GstRTSPClientState *state); void (*get_parameter_request) (GstRTSPClient *client, GstRTSPClientState *state); }; GType gst_rtsp_client_get_type (void); GstRTSPClient * gst_rtsp_client_new (void); void gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server); GstRTSPServer * gst_rtsp_client_get_server (GstRTSPClient * client); void gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool); GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client); void gst_rtsp_client_set_media_mapping (GstRTSPClient *client, GstRTSPMediaMapping *mapping); GstRTSPMediaMapping * gst_rtsp_client_get_media_mapping (GstRTSPClient *client); void gst_rtsp_client_set_use_client_settings (GstRTSPClient * client, gboolean use_client_settings); gboolean gst_rtsp_client_get_use_client_settings (GstRTSPClient * client); void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth); GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client); gboolean gst_rtsp_client_accept (GstRTSPClient *client, GSocket *socket, GCancellable *cancellable, GError **error); gboolean gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket *socket, const gchar * ip, gint port, const gchar *initial_buffer, GError **error); guint gst_rtsp_client_attach (GstRTSPClient *client, GMainContext *context); G_END_DECLS #endif /* __GST_RTSP_CLIENT_H__ */