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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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2df75442d0
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
243 lines
11 KiB
C
243 lines
11 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 _GstRTSPClientPrivate GstRTSPClientPrivate;
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#include "rtsp-server-prelude.h"
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#include "rtsp-context.h"
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#include "rtsp-mount-points.h"
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#include "rtsp-sdp.h"
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#include "rtsp-auth.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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* 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 object represents the connection and its state with a client.
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*/
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struct _GstRTSPClient {
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GObject parent;
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/*< private >*/
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GstRTSPClientPrivate *priv;
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstRTSPClientClass:
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* @create_sdp: called when the SDP needs to be created for media.
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* @configure_client_media: called when the stream in media needs to be configured.
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* The default implementation will configure the blocksize on the payloader when
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* spcified in the request headers.
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* @configure_client_transport: called when the client transport needs to be
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* configured.
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* @params_set: set parameters. This function should also initialize the
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* RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
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* @params_get: get parameters. This function should also initialize the
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* RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
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* @tunnel_http_response: called when a response to the GET request is about to
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* be sent for a tunneled connection. The response can be modified. Since 1.4
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*
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* The client class structure.
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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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gboolean (*configure_client_media) (GstRTSPClient * client,
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GstRTSPMedia * media, GstRTSPStream * stream,
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GstRTSPContext * ctx);
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gboolean (*configure_client_transport) (GstRTSPClient * client,
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GstRTSPContext * ctx,
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GstRTSPTransport * ct);
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GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
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gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
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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, GstRTSPContext *ctx);
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void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request,
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GstRTSPMessage * response);
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void (*send_message) (GstRTSPClient * client, GstRTSPContext *ctx,
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GstRTSPMessage * response);
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gboolean (*handle_sdp) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPMedia *media, GstSDPMessage *sdp);
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void (*announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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gchar* (*check_requirements) (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr);
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GstRTSPStatusCode (*pre_options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE-16];
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};
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GST_RTSP_SERVER_API
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GType gst_rtsp_client_get_type (void);
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GST_RTSP_SERVER_API
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GstRTSPClient * gst_rtsp_client_new (void);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
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GstRTSPSessionPool *pool);
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GST_RTSP_SERVER_API
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GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_mount_points (GstRTSPClient *client,
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GstRTSPMountPoints *mounts);
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GST_RTSP_SERVER_API
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GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
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GST_RTSP_SERVER_API
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GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
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GST_RTSP_SERVER_API
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GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
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GST_RTSP_SERVER_API
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GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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guint gst_rtsp_client_attach (GstRTSPClient *client,
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GMainContext *context);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_close (GstRTSPClient * client);
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GST_RTSP_SERVER_API
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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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GST_RTSP_SERVER_API
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GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
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GstRTSPMessage *message);
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GST_RTSP_SERVER_API
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GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client,
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GstRTSPSession *session,
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GstRTSPMessage *message);
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/**
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* GstRTSPClientSessionFilterFunc:
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* @client: a #GstRTSPClient object
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* @sess: a #GstRTSPSession in @client
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* @user_data: user data that has been given to gst_rtsp_client_session_filter()
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*
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* This function will be called by the gst_rtsp_client_session_filter(). An
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* implementation should return a value of #GstRTSPFilterResult.
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*
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* When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
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* from @client.
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*
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* A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
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* @client.
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*
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* A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
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* gst_rtsp_client_session_filter().
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*
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* Returns: a #GstRTSPFilterResult.
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*/
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typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client,
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GstRTSPSession *sess,
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gpointer user_data);
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GST_RTSP_SERVER_API
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GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
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GstRTSPClientSessionFilterFunc func,
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gpointer user_data);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPClient, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* __GST_RTSP_CLIENT_H__ */
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