*
* 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 <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <errno.h>
-#include <string.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <netinet/in.h>
-#include <netdb.h>
-#include <sys/socket.h>
-#include <sys/wait.h>
-#include <fcntl.h>
-#include <arpa/inet.h>
-
#include <gst/gst.h>
#include <gst/rtsp/gstrtspconnection.h>
#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"
-
-G_BEGIN_DECLS
+#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_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj))
#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
-typedef struct _GstRTSPClient GstRTSPClient;
-typedef struct _GstRTSPClientClass GstRTSPClientClass;
+/**
+ * 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
- * @watchid: id of the watch
- * @timeout: the session timeout
+ * @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;
- guint timeout;
+ GstRTSPServer *server;
GstRTSPSessionPool *session_pool;
GstRTSPMediaMapping *media_mapping;
+ GstRTSPAuth *auth;
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 (*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_session_pool (GstRTSPClient *client,
+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,
+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_timeout (GstRTSPClient *client, guint timeout);
-guint gst_rtsp_client_get_timeout (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);
-gboolean gst_rtsp_client_accept (GstRTSPClient *client,
- GIOChannel *channel);
G_END_DECLS
projects / platform / upstream / gstreamer.git / blobdiff