/* 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 _GstRTSPClientPrivate GstRTSPClientPrivate; #include "rtsp-server-prelude.h" #include "rtsp-context.h" #include "rtsp-mount-points.h" #include "rtsp-sdp.h" #include "rtsp-auth.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)) /** * GstRTSPClientSendFunc: * @client: a #GstRTSPClient * @message: a #GstRTSPMessage * @close: close the connection * @user_data: user data when registering the callback * * This callback is called when @client wants to send @message. When @close is * %TRUE, the connection should be closed when the message has been sent. * * Returns: %TRUE on success. */ typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client, GstRTSPMessage *message, gboolean close, gpointer user_data); /** * GstRTSPClientSendMessagesFunc: * @client: a #GstRTSPClient * @messages: #GstRTSPMessage * @n_messages: number of messages * @close: close the connection * @user_data: user data when registering the callback * * This callback is called when @client wants to send @messages. When @close is * %TRUE, the connection should be closed when the message has been sent. * * Returns: %TRUE on success. * * Since: 1.16 */ typedef gboolean (*GstRTSPClientSendMessagesFunc) (GstRTSPClient *client, GstRTSPMessage *messages, guint n_messages, gboolean close, gpointer user_data); /** * GstRTSPClient: * * The client object represents the connection and its state with a client. */ struct _GstRTSPClient { GObject parent; /*< private >*/ GstRTSPClientPrivate *priv; gpointer _gst_reserved[GST_PADDING]; }; /** * GstRTSPClientClass: * @create_sdp: called when the SDP needs to be created for media. * @configure_client_media: called when the stream in media needs to be configured. * The default implementation will configure the blocksize on the payloader when * spcified in the request headers. * @configure_client_transport: called when the client transport needs to be * configured. * @params_set: set parameters. This function should also initialize the * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() * @params_get: get parameters. This function should also initialize the * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() * @tunnel_http_response: called when a response to the GET request is about to * be sent for a tunneled connection. The response can be modified. Since 1.4 * * The client class structure. */ struct _GstRTSPClientClass { GObjectClass parent_class; GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media); gboolean (*configure_client_media) (GstRTSPClient * client, GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx); gboolean (*configure_client_transport) (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPTransport * ct); GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx); gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri); /* signals */ void (*closed) (GstRTSPClient *client); void (*new_session) (GstRTSPClient *client, GstRTSPSession *session); void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx); void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request, GstRTSPMessage * response); void (*send_message) (GstRTSPClient * client, GstRTSPContext *ctx, GstRTSPMessage * response); gboolean (*handle_sdp) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPMedia *media, GstSDPMessage *sdp); void (*announce_request) (GstRTSPClient *client, GstRTSPContext *ctx); void (*record_request) (GstRTSPClient *client, GstRTSPContext *ctx); gchar* (*check_requirements) (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr); GstRTSPStatusCode (*pre_options_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_describe_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_setup_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_play_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_pause_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_announce_request) (GstRTSPClient *client, GstRTSPContext *ctx); GstRTSPStatusCode (*pre_record_request) (GstRTSPClient *client, GstRTSPContext *ctx); /*< private >*/ gpointer _gst_reserved[GST_PADDING_LARGE-16]; }; GST_RTSP_SERVER_API GType gst_rtsp_client_get_type (void); GST_RTSP_SERVER_API GstRTSPClient * gst_rtsp_client_new (void); GST_RTSP_SERVER_API void gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool); GST_RTSP_SERVER_API GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client); GST_RTSP_SERVER_API void gst_rtsp_client_set_mount_points (GstRTSPClient *client, GstRTSPMountPoints *mounts); GST_RTSP_SERVER_API GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client); GST_RTSP_SERVER_API void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth); GST_RTSP_SERVER_API GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client); GST_RTSP_SERVER_API void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool); GST_RTSP_SERVER_API GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client); GST_RTSP_SERVER_API gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn); GST_RTSP_SERVER_API GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client); GST_RTSP_SERVER_API guint gst_rtsp_client_attach (GstRTSPClient *client, GMainContext *context); GST_RTSP_SERVER_API void gst_rtsp_client_close (GstRTSPClient * client); GST_RTSP_SERVER_API void gst_rtsp_client_set_send_func (GstRTSPClient *client, GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify); GST_RTSP_SERVER_API void gst_rtsp_client_set_send_messages_func (GstRTSPClient *client, GstRTSPClientSendMessagesFunc func, gpointer user_data, GDestroyNotify notify); GST_RTSP_SERVER_API GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client, GstRTSPMessage *message); GST_RTSP_SERVER_API GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession *session, GstRTSPMessage *message); /** * GstRTSPClientSessionFilterFunc: * @client: a #GstRTSPClient object * @sess: a #GstRTSPSession in @client * @user_data: user data that has been given to gst_rtsp_client_session_filter() * * This function will be called by the gst_rtsp_client_session_filter(). An * implementation should return a value of #GstRTSPFilterResult. * * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed * from @client. * * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in * @client. * * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of * gst_rtsp_client_session_filter(). * * Returns: a #GstRTSPFilterResult. */ typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client, GstRTSPSession *sess, gpointer user_data); GST_RTSP_SERVER_API GList * gst_rtsp_client_session_filter (GstRTSPClient *client, GstRTSPClientSessionFilterFunc func, gpointer user_data); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPClient, gst_object_unref) #endif G_END_DECLS #endif /* __GST_RTSP_CLIENT_H__ */