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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d638b03ff9
Let the server accept the socket connection and construct a GstRTSPConnection from it. Remove the code from the client and let the client only deal with a fully configure GstRTSPConnection object. We will need this later when the server will configure the connection for TLS.
158 lines
6.6 KiB
C
158 lines
6.6 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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typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
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#include "rtsp-media.h"
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#include "rtsp-mount-points.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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* GstRTSPClientSendFunc:
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* @client: a #GstRTSPClient
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* @message: a #GstRTSPMessage
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* @close: close the connection
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* @user_data: user data when registering the callback
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*
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* This callback is called when @client wants to send @message. When @close is
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* %TRUE, the connection should be closed when the message has been sent.
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*
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* Returns: %TRUE on success.
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*/
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typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client,
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GstRTSPMessage *message,
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gboolean close,
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gpointer user_data);
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/**
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* GstRTSPClient:
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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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GstRTSPClientPrivate *priv;
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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_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_mount_points (GstRTSPClient *client,
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GstRTSPMountPoints *mounts);
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GstRTSPMountPoints * gst_rtsp_client_get_mount_points (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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GstRTSPUrl * gst_rtsp_client_get_uri (GstRTSPClient *client);
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gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
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GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
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void gst_rtsp_client_set_send_func (GstRTSPClient *client,
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GstRTSPClientSendFunc func,
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gpointer user_data,
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GDestroyNotify notify);
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GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
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GstRTSPMessage *message);
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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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