*
* 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., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#include <gst/gst.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))
* @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 session media 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{
+struct _GstRTSPClientState {
GstRTSPMessage *request;
GstRTSPUrl *uri;
GstRTSPMethod method;
GstRTSPSessionMedia *sessmedia;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
+ GstRTSPStream *stream;
GstRTSPMessage *response;
};
*
* @connection: the connection object handling the client request.
* @watch: watch for the connection
- * @watchid: id of the watch
* @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
- * @streams: a list of streams using @connection.
+ * @transports: a list of #GstRTSPStreamTransport using @connection.
* @sessions: a list of sessions managed by @connection.
*
* The client structure.
GstRTSPConnection *connection;
GstRTSPWatch *watch;
- guint watchid;
gchar *server_ip;
gboolean is_ipv6;
+ gboolean use_client_settings;
GstRTSPServer *server;
GstRTSPSessionPool *session_pool;
GstRTSPUrl *uri;
GstRTSPMedia *media;
- GList *streams;
+ GList *transports;
GList *sessions;
+
+ guint teardown_response_seq;
};
struct _GstRTSPClientClass {
GObjectClass parent_class;
+ GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
+
/* signals */
- void (*closed) (GstRTSPClient *client);
+ 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);
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);
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__ */