gstreamer/gst/rtsp-server/rtsp-client.h
Ognyan Tonchev d581b7bd4e client: Use client transport settings for multicast if allowed.
This patch makes it possible for the client to send transport settings for
multicast (destination && ttl). Client settings must be explicitly allowed or
the server will use its own settings.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
2012-10-10 11:07:59 +02:00

166 lines
6.8 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#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"
#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_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))
/**
* 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 session media 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;
GstRTSPMessage *response;
};
/**
* 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
* @streams: a list of streams using @connection.
* @sessions: a list of sessions managed by @connection.
*
* The client structure.
*/
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 *streams;
GList *sessions;
};
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_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_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_create_from_socket(GstRTSPClient * client,
GSocket *socket,
const gchar * ip,
gint port,
const gchar *initial_buffer,
GError **error);
G_END_DECLS
#endif /* __GST_RTSP_CLIENT_H__ */