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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d581b7bd4e
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
166 lines
6.8 KiB
C
166 lines
6.8 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <gst/gst.h>
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#include <gst/rtsp/gstrtspconnection.h>
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#ifndef __GST_RTSP_CLIENT_H__
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#define __GST_RTSP_CLIENT_H__
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G_BEGIN_DECLS
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typedef struct _GstRTSPClient GstRTSPClient;
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typedef struct _GstRTSPClientClass GstRTSPClientClass;
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typedef struct _GstRTSPClientState GstRTSPClientState;
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#include "rtsp-server.h"
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#include "rtsp-media.h"
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#include "rtsp-media-mapping.h"
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#include "rtsp-session-pool.h"
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#include "rtsp-auth.h"
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#include "rtsp-sdp.h"
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#define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ())
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#define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
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#define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT))
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#define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
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#define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient))
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#define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
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#define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj))
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#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
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/**
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* GstRTSPClientState:
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* @request: the complete request
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* @uri: the complete url parsed from @request
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* @method: the parsed method of @uri
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* @session: the session, can be NULL
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* @sessmedia: the session media for the url can be NULL
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* @factory: the media factory for the url, can be NULL.
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* @media: the session media for the url can be NULL
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* @response: the response
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*
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* Information passed around containing the client state of a request.
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*/
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struct _GstRTSPClientState{
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GstRTSPMessage *request;
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GstRTSPUrl *uri;
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GstRTSPMethod method;
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GstRTSPSession *session;
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GstRTSPSessionMedia *sessmedia;
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GstRTSPMediaFactory *factory;
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GstRTSPMedia *media;
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GstRTSPMessage *response;
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};
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/**
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* GstRTSPClient:
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*
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* @connection: the connection object handling the client request.
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* @watch: watch for the connection
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* @watchid: id of the watch
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* @ip: ip address used by the client to connect to us
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* @use_client_settings: whether to allow client transport settings for multicast
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* @session_pool: handle to the session pool used by the client.
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* @media_mapping: handle to the media mapping used by the client.
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* @uri: cached uri
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* @media: cached media
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* @streams: a list of streams using @connection.
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* @sessions: a list of sessions managed by @connection.
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*
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* The client structure.
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*/
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struct _GstRTSPClient {
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GObject parent;
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GstRTSPConnection *connection;
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GstRTSPWatch *watch;
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guint watchid;
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gchar *server_ip;
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gboolean is_ipv6;
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gboolean use_client_settings;
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GstRTSPServer *server;
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GstRTSPSessionPool *session_pool;
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GstRTSPMediaMapping *media_mapping;
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GstRTSPAuth *auth;
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GstRTSPUrl *uri;
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GstRTSPMedia *media;
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GList *streams;
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GList *sessions;
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};
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struct _GstRTSPClientClass {
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GObjectClass parent_class;
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GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
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/* signals */
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void (*closed) (GstRTSPClient *client);
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void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
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void (*options_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*describe_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*setup_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*play_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*pause_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*teardown_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*set_parameter_request) (GstRTSPClient *client, GstRTSPClientState *state);
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void (*get_parameter_request) (GstRTSPClient *client, GstRTSPClientState *state);
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};
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GType gst_rtsp_client_get_type (void);
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GstRTSPClient * gst_rtsp_client_new (void);
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void gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server);
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GstRTSPServer * gst_rtsp_client_get_server (GstRTSPClient * client);
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void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
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GstRTSPSessionPool *pool);
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GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
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void gst_rtsp_client_set_media_mapping (GstRTSPClient *client,
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GstRTSPMediaMapping *mapping);
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GstRTSPMediaMapping * gst_rtsp_client_get_media_mapping (GstRTSPClient *client);
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void gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
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gboolean use_client_settings);
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gboolean gst_rtsp_client_get_use_client_settings (GstRTSPClient * client);
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void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
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GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
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gboolean gst_rtsp_client_accept (GstRTSPClient *client,
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GSocket *socket,
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GCancellable *cancellable,
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GError **error);
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gboolean gst_rtsp_client_create_from_socket(GstRTSPClient * client,
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GSocket *socket,
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const gchar * ip,
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gint port,
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const gchar *initial_buffer,
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GError **error);
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G_END_DECLS
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#endif /* __GST_RTSP_CLIENT_H__ */
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