X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtsp-server%2Frtsp-client.h;h=01f820826d67ecd731a7fe7d6d605a9658329209;hb=faf0b31cbbdeba09217027fb9133946a3a72cf05;hp=a896ea6993b03c98218b95c5604f8f44879db819;hpb=fb117a4f751f7860c9d0d96e3291855bdba9bdfe;p=platform%2Fupstream%2Fgstreamer.git diff --git a/gst/rtsp-server/rtsp-client.h b/gst/rtsp-server/rtsp-client.h index a896ea6..01f8208 100644 --- a/gst/rtsp-server/rtsp-client.h +++ b/gst/rtsp-server/rtsp-client.h @@ -13,8 +13,8 @@ * * 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 @@ -27,15 +27,12 @@ G_BEGIN_DECLS typedef struct _GstRTSPClient GstRTSPClient; typedef struct _GstRTSPClientClass GstRTSPClientClass; -typedef struct _GstRTSPClientState GstRTSPClientState; +typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate; -#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-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)) @@ -47,122 +44,148 @@ typedef struct _GstRTSPClientState GstRTSPClientState; #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 + * GstRTSPClientSendFunc: + * @client: a #GstRTSPClient + * @message: a #GstRTSPMessage + * @close: close the connection + * @user_data: user data when registering the callback * - * Information passed around containing the client state of a request. + * 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. */ -struct _GstRTSPClientState { - GstRTSPMessage *request; - GstRTSPUrl *uri; - GstRTSPMethod method; - GstRTSPSession *session; - GstRTSPSessionMedia *sessmedia; - GstRTSPMediaFactory *factory; - GstRTSPMedia *media; - GstRTSPStream *stream; - GstRTSPMessage *response; -}; +typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client, + GstRTSPMessage *message, + gboolean close, + gpointer user_data); /** * GstRTSPClient: * - * @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 - * @transports: a list of #GstRTSPStreamTransport using @connection. - * @sessions: a list of sessions managed by @connection. - * - * The client structure. + * The client object represents the connection and its state with a client. */ struct _GstRTSPClient { GObject parent; - GstRTSPConnection *connection; - GstRTSPWatch *watch; - guint watchid; - 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; + /*< 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, 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); + 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_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_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, - GSocket *socket, - GCancellable *cancellable, - GError **error); -gboolean gst_rtsp_client_create_from_socket(GstRTSPClient * client, - GSocket *socket, - const gchar * ip, - gint port, - const gchar *initial_buffer, - GError **error); G_END_DECLS