gstreamer/gst/rtsp-server/rtsp-client.h
David Svensson Fors faf0b31cbb rtsp-client: vmethod for modifying tunnel GET response
Add a vmethod tunnel_http_response where the response to the HTTP GET
for tunneled connections can be modified.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2014-03-11 11:17:04 +01:00

193 lines
8.9 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/gst.h>
#include <gst/rtsp/gstrtspconnection.h>
#ifndef __GST_RTSP_CLIENT_H__
#define __GST_RTSP_CLIENT_H__
G_BEGIN_DECLS
typedef struct _GstRTSPClient GstRTSPClient;
typedef struct _GstRTSPClientClass GstRTSPClientClass;
typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
#include "rtsp-context.h"
#include "rtsp-mount-points.h"
#include "rtsp-sdp.h"
#include "rtsp-auth.h"
#define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ())
#define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
#define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT))
#define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
#define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient))
#define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
#define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj))
#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
/**
* GstRTSPClientSendFunc:
* @client: a #GstRTSPClient
* @message: a #GstRTSPMessage
* @close: close the connection
* @user_data: user data when registering the callback
*
* This callback is called when @client wants to send @message. When @close is
* %TRUE, the connection should be closed when the message has been sent.
*
* Returns: %TRUE on success.
*/
typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client,
GstRTSPMessage *message,
gboolean close,
gpointer user_data);
/**
* GstRTSPClient:
*
* The client object represents the connection and its state with a client.
*/
struct _GstRTSPClient {
GObject parent;
/*< private >*/
GstRTSPClientPrivate *priv;
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstRTSPClientClass:
* @create_sdp: called when the SDP needs to be created for media.
* @configure_client_media: called when the stream in media needs to be configured.
* The default implementation will configure the blocksize on the payloader when
* spcified in the request headers.
* @configure_client_transport: called when the client transport needs to be
* configured.
* @params_set: set parameters. This function should also initialize the
* RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
* @params_get: get parameters. This function should also initialize the
* RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
* @tunnel_http_response: called when a response to the GET request is about to
* be sent for a tunneled connection. The response can be modified. Since 1.4
*
* The client class structure.
*/
struct _GstRTSPClientClass {
GObjectClass parent_class;
GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
gboolean (*configure_client_media) (GstRTSPClient * client,
GstRTSPMedia * media, GstRTSPStream * stream,
GstRTSPContext * ctx);
gboolean (*configure_client_transport) (GstRTSPClient * client,
GstRTSPContext * ctx,
GstRTSPTransport * ct);
GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
/* signals */
void (*closed) (GstRTSPClient *client);
void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request,
GstRTSPMessage * response);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE-1];
};
GType gst_rtsp_client_get_type (void);
GstRTSPClient * gst_rtsp_client_new (void);
void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
GstRTSPSessionPool *pool);
GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
void gst_rtsp_client_set_mount_points (GstRTSPClient *client,
GstRTSPMountPoints *mounts);
GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
guint gst_rtsp_client_attach (GstRTSPClient *client,
GMainContext *context);
void gst_rtsp_client_set_send_func (GstRTSPClient *client,
GstRTSPClientSendFunc func,
gpointer user_data,
GDestroyNotify notify);
GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
GstRTSPMessage *message);
GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client,
GstRTSPSession *session,
GstRTSPMessage *message);
/**
* GstRTSPClientSessionFilterFunc:
* @client: a #GstRTSPClient object
* @sess: a #GstRTSPSession in @client
* @user_data: user data that has been given to gst_rtsp_client_session_filter()
*
* This function will be called by the gst_rtsp_client_session_filter(). An
* implementation should return a value of #GstRTSPFilterResult.
*
* When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
* from @client.
*
* A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
* @client.
*
* A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
* gst_rtsp_client_session_filter().
*
* Returns: a #GstRTSPFilterResult.
*/
typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client,
GstRTSPSession *sess,
gpointer user_data);
GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
GstRTSPClientSessionFilterFunc func,
gpointer user_data);
G_END_DECLS
#endif /* __GST_RTSP_CLIENT_H__ */