mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-22 17:51:16 +00:00
ae8a5e110c
Confuses the documentation builder, since it's documented twice it complains about a missing "Since:" marker whereas it's present in the documentation comment further down Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
309 lines
14 KiB
C
309 lines
14 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-server-prelude.h"
|
|
#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);
|
|
|
|
/**
|
|
* 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:
|
|
*
|
|
* 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()
|
|
* @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
|
|
* 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);
|
|
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);
|
|
|
|
/* 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];
|
|
};
|
|
|
|
GST_RTSP_SERVER_API
|
|
GType gst_rtsp_client_get_type (void);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPClient * gst_rtsp_client_new (void);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
|
|
GstRTSPSessionPool *pool);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
|
|
|
|
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);
|
|
|
|
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);
|
|
|
|
GST_RTSP_SERVER_API
|
|
guint gst_rtsp_client_attach (GstRTSPClient *client,
|
|
GMainContext *context);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_client_close (GstRTSPClient * client);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_client_set_send_func (GstRTSPClient *client,
|
|
GstRTSPClientSendFunc func,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
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.
|
|
*
|
|
* 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
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_CLIENT_H__ */
|