mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 18:51:11 +00:00
33da3af265
Make a method to let the client handle a message and a callback when the client wants us to send a response message back. This makes it possible to also use the client object without the sockets, which should make it easier to test.
203 lines
8.5 KiB
C
203 lines
8.5 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 _GstRTSPClientState GstRTSPClientState;
|
|
|
|
#include "rtsp-server.h"
|
|
#include "rtsp-media.h"
|
|
#include "rtsp-mount-points.h"
|
|
#include "rtsp-session-pool.h"
|
|
#include "rtsp-session-media.h"
|
|
#include "rtsp-auth.h"
|
|
#include "rtsp-sdp.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))
|
|
|
|
/**
|
|
* GstRTSPClientState:
|
|
* @request: the complete request
|
|
* @uri: the complete url parsed from @request
|
|
* @method: the parsed method of @uri
|
|
* @session: the session, can be NULL
|
|
* @sessmedia: the session media for the url can be NULL
|
|
* @factory: the media factory for the url, can be NULL.
|
|
* @media: the media for the url can be NULL
|
|
* @stream: the stream for the url can be NULL
|
|
* @response: the response
|
|
*
|
|
* Information passed around containing the client state of a request.
|
|
*/
|
|
struct _GstRTSPClientState {
|
|
GstRTSPMessage *request;
|
|
GstRTSPUrl *uri;
|
|
GstRTSPMethod method;
|
|
GstRTSPSession *session;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPMediaFactory *factory;
|
|
GstRTSPMedia *media;
|
|
GstRTSPStream *stream;
|
|
GstRTSPMessage *response;
|
|
};
|
|
|
|
/**
|
|
* 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:
|
|
* @lock: lock protecting the client object
|
|
* @connection: the connection object handling the client request.
|
|
* @watch: watch for the connection
|
|
* @close_seq: sequence number of message with close header
|
|
* @server_ip: ip address of the server
|
|
* @is_ipv6: if we are IPv6
|
|
* @use_client_settings: whether to allow client transport settings for multicast
|
|
* @send_func: a #GstRTSPClientSendFunc called when an RTSP message needs to be
|
|
* sent to the client.
|
|
* @send_data: user data passed to @send_func
|
|
* @send_notify: notify called when @send_data is no longer used.
|
|
* @session_pool: handle to the session pool used by the client.
|
|
* @mount_points: handle to the mount points used by the client.
|
|
* @auth: authorization object
|
|
* @uri: cached uri
|
|
* @media: cached media
|
|
* @transports: a list of #GstRTSPStreamTransport using @connection.
|
|
* @sessions: a list of sessions managed by @connection.
|
|
*
|
|
* The client structure.
|
|
*/
|
|
struct _GstRTSPClient {
|
|
GObject parent;
|
|
|
|
GMutex lock;
|
|
GstRTSPConnection *connection;
|
|
GstRTSPWatch *watch;
|
|
guint close_seq;
|
|
gchar *server_ip;
|
|
gboolean is_ipv6;
|
|
gboolean use_client_settings;
|
|
|
|
GstRTSPClientSendFunc send_func;
|
|
gpointer send_data;
|
|
GDestroyNotify send_notify;
|
|
|
|
GstRTSPSessionPool *session_pool;
|
|
GstRTSPMountPoints *mount_points;
|
|
GstRTSPAuth *auth;
|
|
|
|
GstRTSPUrl *uri;
|
|
GstRTSPMedia *media;
|
|
|
|
GList *transports;
|
|
GList *sessions;
|
|
};
|
|
|
|
struct _GstRTSPClientClass {
|
|
GObjectClass parent_class;
|
|
|
|
GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
|
|
|
|
/* signals */
|
|
void (*closed) (GstRTSPClient *client);
|
|
void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
|
|
void (*options_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*describe_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*setup_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*play_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*pause_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*teardown_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*set_parameter_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
void (*get_parameter_request) (GstRTSPClient *client, GstRTSPClientState *state);
|
|
};
|
|
|
|
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_use_client_settings (GstRTSPClient * client,
|
|
gboolean use_client_settings);
|
|
gboolean gst_rtsp_client_get_use_client_settings (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_send_func (GstRTSPClient *client,
|
|
GstRTSPClientSendFunc func,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
|
|
GstRTSPMessage *message);
|
|
|
|
gboolean gst_rtsp_client_accept (GstRTSPClient *client,
|
|
GSocket *socket,
|
|
GCancellable *cancellable,
|
|
GError **error);
|
|
|
|
gboolean gst_rtsp_client_use_socket (GstRTSPClient * client,
|
|
GSocket *socket,
|
|
const gchar * ip,
|
|
gint port,
|
|
const gchar *initial_buffer,
|
|
GError **error);
|
|
|
|
guint gst_rtsp_client_attach (GstRTSPClient *client,
|
|
GMainContext *context);
|
|
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_CLIENT_H__ */
|