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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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de7c72dec2
Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
167 lines
6.9 KiB
C
167 lines
6.9 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-session-media.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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* @transports: a list of #GstRTSPStreamTransport 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 *transports;
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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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