*
* 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__
-#include "rtsp-media.h"
-#include "rtsp-media-mapping.h"
-#include "rtsp-session-pool.h"
-
G_BEGIN_DECLS
+typedef struct _GstRTSPClient GstRTSPClient;
+typedef struct _GstRTSPClientClass GstRTSPClientClass;
+typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
+
+#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_CAST(obj) ((GstRTSPClient*)(obj))
#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
-typedef struct _GstRTSPClient GstRTSPClient;
-typedef struct _GstRTSPClientClass GstRTSPClientClass;
+/**
+ * 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);
/**
* GstRTSPClient:
*
- * @connection: the connection object handling the client request.
- * @address: the address of the connection
- * @media: handle to the media handled by the client.
- * @pool: handle to the session pool used by the client.
- * @thread: thread to handle the client connection
- *
- * The client structure.
+ * The client object represents the connection and its state with a client.
*/
struct _GstRTSPClient {
GObject parent;
- GstRTSPConnection *connection;
- struct sockaddr_in address;
- GThread *thread;
-
- GstRTSPSessionPool *session_pool;
- GstRTSPMediaMapping *media_mapping;
-
- GstRTSPUrl *uri;
- GstRTSPMedia *media;
+ /*< 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);
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE-1];
};
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_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_mount_points (GstRTSPClient *client,
+ GstRTSPMountPoints *mounts);
+GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
+
+void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
+GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
+
+void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
+GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
+
+gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
+GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
+
+guint gst_rtsp_client_attach (GstRTSPClient *client,
+ GMainContext *context);
+
+void gst_rtsp_client_set_send_func (GstRTSPClient *client,
+ GstRTSPClientSendFunc func,
+ gpointer user_data,
+ GDestroyNotify notify);
+
+GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
+ GstRTSPMessage *message);
+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);
+
+GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
+ GstRTSPClientSessionFilterFunc func,
+ gpointer user_data);
+
-gboolean gst_rtsp_client_accept (GstRTSPClient *client,
- GIOChannel *channel);
G_END_DECLS
projects / platform / upstream / gstreamer.git / blobdiff