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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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65042a9551
Responses can be sent async so we need to wait until the TEARDOWN response has been written before we close the connection to the client. This avoids the risk of writing/polling closed sockets. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
173 lines
7.1 KiB
C
173 lines
7.1 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 media for the url can be NULL
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* @stream: the stream 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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GstRTSPStream *stream;
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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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* @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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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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guint teardown_response_seq;
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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_use_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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guint gst_rtsp_client_attach (GstRTSPClient *client,
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GMainContext *context);
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G_END_DECLS
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#endif /* __GST_RTSP_CLIENT_H__ */
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