gstreamer/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-client.h

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

310 lines
14 KiB
C
Raw Normal View History

2008-10-09 12:29:12 +00:00
/* 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
2012-11-04 00:14:25 +00:00
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
2008-10-09 12:29:12 +00:00
*/
#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-server-prelude.h"
#include "rtsp-context.h"
#include "rtsp-mount-points.h"
#include "rtsp-sdp.h"
#include "rtsp-auth.h"
2008-10-09 12:29:12 +00:00
#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);
/**
* GstRTSPClientSendMessagesFunc:
* @client: a #GstRTSPClient
* @messages: #GstRTSPMessage
* @n_messages: number of messages
* @close: close the connection
* @user_data: user data when registering the callback
*
* This callback is called when @client wants to send @messages. When @close is
* %TRUE, the connection should be closed when the message has been sent.
*
* Returns: %TRUE on success.
*
* Since: 1.16
*/
typedef gboolean (*GstRTSPClientSendMessagesFunc) (GstRTSPClient *client,
GstRTSPMessage *messages,
guint n_messages,
gboolean close,
gpointer user_data);
/**
* GstRTSPClient:
*
2013-07-11 10:18:26 +00:00
* The client object represents the connection and its state with a client.
*/
2008-10-09 12:29:12 +00:00
struct _GstRTSPClient {
GObject parent;
/*< private >*/
GstRTSPClientPrivate *priv;
gpointer _gst_reserved[GST_PADDING];
2008-10-09 12:29:12 +00:00
};
/**
* GstRTSPClientClass:
2013-07-11 10:18:26 +00:00
* @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.
2013-07-11 10:18:26 +00:00
* @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()
* @make_path_from_uri: called to create path from uri.
* @adjust_play_mode: called to give the application the possibility to adjust
* the range, seek flags, rate and rate-control. Since 1.18
* @adjust_play_response: called to give the implementation the possibility to
* adjust the response to a play request, for example if extra headers were
* parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18
* @tunnel_http_response: called when a response to the GET request is about to
2019-04-23 12:09:34 +00:00
* be sent for a tunneled connection. The response can be modified. Since: 1.4
*
* The client class structure.
*/
2008-10-09 12:29:12 +00:00
struct _GstRTSPClientClass {
GObjectClass parent_class;
2011-01-12 14:35:51 +00:00
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);
GstRTSPStatusCode (*adjust_play_mode) (GstRTSPClient * client,
GstRTSPContext * context,
GstRTSPTimeRange ** range,
GstSeekFlags * flags,
gdouble * rate,
GstClockTime * trickmode_interval,
gboolean * enable_rate_control);
GstRTSPStatusCode (*adjust_play_response) (GstRTSPClient * client,
GstRTSPContext * context);
2011-01-12 14:35:51 +00:00
/* 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);
void (*send_message) (GstRTSPClient * client, GstRTSPContext *ctx,
GstRTSPMessage * response);
gboolean (*handle_sdp) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPMedia *media, GstSDPMessage *sdp);
void (*announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
void (*record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
gchar* (*check_requirements) (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr);
GstRTSPStatusCode (*pre_options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
GstRTSPStatusCode (*pre_record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
/**
* GstRTSPClientClass::adjust_error_code:
* @client: a #GstRTSPClient
* @ctx: a #GstRTSPContext
* @code: a #GstRTSPStatusCode
*
* Called before sending error response to give the application the
* possibility to adjust the error code.
*
* Returns: a #GstRTSPStatusCode, containing the adjusted error code.
*
* Since: 1.22
*/
GstRTSPStatusCode (*adjust_error_code) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPStatusCode code);
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE-19];
2008-10-09 12:29:12 +00:00
};
GST_RTSP_SERVER_API
GType gst_rtsp_client_get_type (void);
GST_RTSP_SERVER_API
GstRTSPClient * gst_rtsp_client_new (void);
2008-10-09 12:29:12 +00:00
GST_RTSP_SERVER_API
2010-12-28 17:31:26 +00:00
void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
GstRTSPSessionPool *pool);
GST_RTSP_SERVER_API
GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
2008-10-09 12:29:12 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_client_set_mount_points (GstRTSPClient *client,
GstRTSPMountPoints *mounts);
GST_RTSP_SERVER_API
GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
2008-10-09 12:29:12 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_client_set_content_length_limit (GstRTSPClient *client, guint limit);
GST_RTSP_SERVER_API
guint gst_rtsp_client_get_content_length_limit (GstRTSPClient *client);
GST_RTSP_SERVER_API
void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
GST_RTSP_SERVER_API
GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
GST_RTSP_SERVER_API
void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
GST_RTSP_SERVER_API
GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
GST_RTSP_SERVER_API
gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
GST_RTSP_SERVER_API
GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
2013-03-18 08:25:54 +00:00
GST_RTSP_SERVER_API
2013-07-11 14:57:14 +00:00
guint gst_rtsp_client_attach (GstRTSPClient *client,
GMainContext *context);
GST_RTSP_SERVER_API
void gst_rtsp_client_close (GstRTSPClient * client);
2013-07-11 14:57:14 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_client_set_send_func (GstRTSPClient *client,
GstRTSPClientSendFunc func,
gpointer user_data,
GDestroyNotify notify);
2013-07-11 14:57:14 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_client_set_send_messages_func (GstRTSPClient *client,
GstRTSPClientSendMessagesFunc func,
gpointer user_data,
GDestroyNotify notify);
GST_RTSP_SERVER_API
GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
GstRTSPMessage *message);
GST_RTSP_SERVER_API
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.
*
2013-07-11 20:12:04 +00:00
* 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);
GST_RTSP_SERVER_API
GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
GstRTSPClientSessionFilterFunc func,
gpointer user_data);
GST_RTSP_SERVER_API
GstRTSPStreamTransport * gst_rtsp_client_get_stream_transport (GstRTSPClient *client,
guint8 channel);
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPClient, gst_object_unref)
#endif
2008-10-09 12:29:12 +00:00
G_END_DECLS
#endif /* __GST_RTSP_CLIENT_H__ */