mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 11:11:08 +00:00
389 lines
16 KiB
C
389 lines
16 KiB
C
/* GStreamer
|
|
* Copyright (C) <2005,2009> Wim Taymans <wim.taymans@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.
|
|
*/
|
|
/*
|
|
* Unless otherwise indicated, Source Code is licensed under MIT license.
|
|
* See further explanation attached in License Statement (distributed in the file
|
|
* LICENSE).
|
|
*
|
|
* Permission is hereby granted, free of charge, to any person obtaining a copy of
|
|
* this software and associated documentation files (the "Software"), to deal in
|
|
* the Software without restriction, including without limitation the rights to
|
|
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
|
|
* of the Software, and to permit persons to whom the Software is furnished to do
|
|
* so, subject to the following conditions:
|
|
*
|
|
* The above copyright notice and this permission notice shall be included in all
|
|
* copies or substantial portions of the Software.
|
|
*
|
|
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
|
|
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
|
|
* SOFTWARE.
|
|
*/
|
|
|
|
#ifndef __GST_RTSP_CONNECTION_H__
|
|
#define __GST_RTSP_CONNECTION_H__
|
|
|
|
#include <glib.h>
|
|
|
|
#include <gst/gstconfig.h>
|
|
#include <gst/rtsp/gstrtspdefs.h>
|
|
#include <gst/rtsp/gstrtspurl.h>
|
|
#include <gst/rtsp/gstrtspmessage.h>
|
|
#include <gio/gio.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
/**
|
|
* GstRTSPConnection:
|
|
*
|
|
* Opaque RTSP connection object.
|
|
*/
|
|
typedef struct _GstRTSPConnection GstRTSPConnection;
|
|
|
|
/* opening/closing a connection */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_create (const GstRTSPUrl *url, GstRTSPConnection **conn);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_create_from_socket (GSocket * socket,
|
|
const gchar * ip,
|
|
guint16 port,
|
|
const gchar * initial_buffer,
|
|
GstRTSPConnection ** conn);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_accept (GSocket * socket, GstRTSPConnection ** conn, GCancellable * cancellable);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_connect_usec (GstRTSPConnection * conn, gint64 timeout);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_connect_with_response_usec (GstRTSPConnection * conn, gint64 timeout, GstRTSPMessage * response);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_close (GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_free (GstRTSPConnection *conn);
|
|
|
|
/* TLS connections */
|
|
|
|
GST_RTSP_API
|
|
GTlsConnection * gst_rtsp_connection_get_tls (GstRTSPConnection * conn, GError ** error);
|
|
|
|
GST_RTSP_API
|
|
gboolean gst_rtsp_connection_set_tls_validation_flags (GstRTSPConnection * conn, GTlsCertificateFlags flags);
|
|
|
|
GST_RTSP_API
|
|
GTlsCertificateFlags gst_rtsp_connection_get_tls_validation_flags (GstRTSPConnection * conn);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_tls_database (GstRTSPConnection * conn, GTlsDatabase * database);
|
|
|
|
GST_RTSP_API
|
|
GTlsDatabase * gst_rtsp_connection_get_tls_database (GstRTSPConnection * conn);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_tls_interaction (GstRTSPConnection * conn, GTlsInteraction * interaction);
|
|
|
|
GST_RTSP_API
|
|
GTlsInteraction * gst_rtsp_connection_get_tls_interaction (GstRTSPConnection * conn);
|
|
|
|
typedef gboolean (*GstRTSPConnectionAcceptCertificateFunc) (GTlsConnection *conn,
|
|
GTlsCertificate *peer_cert,
|
|
GTlsCertificateFlags errors,
|
|
gpointer user_data);
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_accept_certificate_func (GstRTSPConnection * conn,
|
|
GstRTSPConnectionAcceptCertificateFunc func,
|
|
gpointer user_data,
|
|
GDestroyNotify destroy_notify);
|
|
|
|
/* sending/receiving raw bytes */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_read_usec (GstRTSPConnection * conn, guint8 * data,
|
|
guint size, gint64 timeout);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_write_usec (GstRTSPConnection * conn, const guint8 * data,
|
|
guint size, gint64 timeout);
|
|
|
|
/* sending/receiving messages */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_send_usec (GstRTSPConnection *conn, GstRTSPMessage *message,
|
|
gint64 timeout);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_send_messages_usec (GstRTSPConnection *conn, GstRTSPMessage *messages, guint n_messages,
|
|
gint64 timeout);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_receive_usec (GstRTSPConnection *conn, GstRTSPMessage *message,
|
|
gint64 timeout);
|
|
|
|
/* status management */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_poll_usec (GstRTSPConnection *conn, GstRTSPEvent events,
|
|
GstRTSPEvent *revents, gint64 timeout);
|
|
|
|
/* reset the timeout */
|
|
|
|
GST_RTSP_API
|
|
gint64 gst_rtsp_connection_next_timeout_usec (GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_reset_timeout (GstRTSPConnection *conn);
|
|
|
|
/* flushing state */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_flush (GstRTSPConnection *conn, gboolean flush);
|
|
|
|
/* HTTP proxy support */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_set_proxy (GstRTSPConnection *conn,
|
|
const gchar *host, guint port);
|
|
|
|
/* configure authentication data */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_set_auth (GstRTSPConnection *conn, GstRTSPAuthMethod method,
|
|
const gchar *user, const gchar *pass);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_auth_param (GstRTSPConnection *conn,
|
|
const gchar * param,
|
|
const gchar *value);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_clear_auth_params (GstRTSPConnection *conn);
|
|
|
|
/* configure DSCP */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_set_qos_dscp (GstRTSPConnection *conn,
|
|
guint qos_dscp);
|
|
|
|
/* Content-Length limit */
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_content_length_limit (GstRTSPConnection *conn,
|
|
guint limit);
|
|
|
|
/* accessors */
|
|
|
|
GST_RTSP_API
|
|
GstRTSPUrl * gst_rtsp_connection_get_url (const GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
const gchar * gst_rtsp_connection_get_ip (const GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_ip (GstRTSPConnection *conn, const gchar *ip);
|
|
|
|
GST_RTSP_API
|
|
GSocket * gst_rtsp_connection_get_read_socket (const GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
GSocket * gst_rtsp_connection_get_write_socket (const GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_http_mode (GstRTSPConnection *conn,
|
|
gboolean enable);
|
|
|
|
/* tunneling */
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_tunneled (GstRTSPConnection *conn, gboolean tunneled);
|
|
|
|
GST_RTSP_API
|
|
gboolean gst_rtsp_connection_is_tunneled (const GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
const gchar * gst_rtsp_connection_get_tunnelid (const GstRTSPConnection *conn);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_connection_do_tunnel (GstRTSPConnection *conn, GstRTSPConnection *conn2);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_connection_set_remember_session_id (GstRTSPConnection *conn, gboolean remember);
|
|
|
|
GST_RTSP_API
|
|
gboolean gst_rtsp_connection_get_remember_session_id (GstRTSPConnection *conn);
|
|
|
|
/* async IO */
|
|
|
|
/**
|
|
* GstRTSPWatch:
|
|
*
|
|
* Opaque RTSP watch object that can be used for asynchronous RTSP
|
|
* operations.
|
|
*/
|
|
typedef struct _GstRTSPWatch GstRTSPWatch;
|
|
|
|
/**
|
|
* GstRTSPWatchFuncs:
|
|
* @message_received: callback when a message was received
|
|
* @message_sent: callback when a message was sent
|
|
* @closed: callback when the connection is closed
|
|
* @error: callback when an error occurred
|
|
* @tunnel_start: a client started a tunneled connection. The tunnelid of the
|
|
* connection must be saved.
|
|
* @tunnel_complete: a client finished a tunneled connection. In this callback
|
|
* you usually pair the tunnelid of this connection with the saved one using
|
|
* gst_rtsp_connection_do_tunnel().
|
|
* @error_full: callback when an error occurred with more information than
|
|
* the @error callback.
|
|
* @tunnel_lost: callback when the post connection of a tunnel is closed.
|
|
* @tunnel_http_response: callback when an HTTP response to the GET request
|
|
* is about to be sent for a tunneled connection. The response can be
|
|
* modified in the callback. Since: 1.4.
|
|
*
|
|
* Callback functions from a #GstRTSPWatch.
|
|
*/
|
|
typedef struct {
|
|
GstRTSPResult (*message_received) (GstRTSPWatch *watch, GstRTSPMessage *message,
|
|
gpointer user_data);
|
|
GstRTSPResult (*message_sent) (GstRTSPWatch *watch, guint id,
|
|
gpointer user_data);
|
|
GstRTSPResult (*closed) (GstRTSPWatch *watch, gpointer user_data);
|
|
GstRTSPResult (*error) (GstRTSPWatch *watch, GstRTSPResult result,
|
|
gpointer user_data);
|
|
GstRTSPStatusCode (*tunnel_start) (GstRTSPWatch *watch, gpointer user_data);
|
|
GstRTSPResult (*tunnel_complete) (GstRTSPWatch *watch, gpointer user_data);
|
|
GstRTSPResult (*error_full) (GstRTSPWatch *watch, GstRTSPResult result,
|
|
GstRTSPMessage *message, guint id,
|
|
gpointer user_data);
|
|
GstRTSPResult (*tunnel_lost) (GstRTSPWatch *watch, gpointer user_data);
|
|
GstRTSPResult (*tunnel_http_response) (GstRTSPWatch *watch,
|
|
GstRTSPMessage *request,
|
|
GstRTSPMessage *response,
|
|
gpointer user_data);
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING-1];
|
|
} GstRTSPWatchFuncs;
|
|
|
|
GST_RTSP_API
|
|
GstRTSPWatch * gst_rtsp_watch_new (GstRTSPConnection *conn,
|
|
GstRTSPWatchFuncs *funcs,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_watch_reset (GstRTSPWatch *watch);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_watch_unref (GstRTSPWatch *watch);
|
|
|
|
GST_RTSP_API
|
|
guint gst_rtsp_watch_attach (GstRTSPWatch *watch,
|
|
GMainContext *context);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_watch_set_send_backlog (GstRTSPWatch *watch,
|
|
gsize bytes, guint messages);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_watch_get_send_backlog (GstRTSPWatch *watch,
|
|
gsize *bytes, guint *messages);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_watch_write_data (GstRTSPWatch *watch,
|
|
const guint8 *data,
|
|
guint size, guint *id);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_watch_send_message (GstRTSPWatch *watch,
|
|
GstRTSPMessage *message,
|
|
guint *id);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_watch_send_messages (GstRTSPWatch *watch,
|
|
GstRTSPMessage *messages,
|
|
guint n_messages,
|
|
guint *id);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_watch_wait_backlog_usec (GstRTSPWatch * watch,
|
|
gint64 timeout);
|
|
|
|
GST_RTSP_API
|
|
void gst_rtsp_watch_set_flushing (GstRTSPWatch * watch,
|
|
gboolean flushing);
|
|
|
|
#ifndef GST_DISABLE_DEPRECATED
|
|
|
|
/* Deprecated */
|
|
G_GNUC_BEGIN_IGNORE_DEPRECATIONS
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_connect_usec)
|
|
GstRTSPResult gst_rtsp_connection_connect (GstRTSPConnection * conn, GTimeVal * timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_connect_with_response_usec)
|
|
GstRTSPResult gst_rtsp_connection_connect_with_response (GstRTSPConnection * conn, GTimeVal * timeout, GstRTSPMessage * response);
|
|
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_read_usec)
|
|
GstRTSPResult gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data,
|
|
guint size, GTimeVal * timeout);
|
|
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_write_usec)
|
|
GstRTSPResult gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
|
|
guint size, GTimeVal * timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_send_usec)
|
|
GstRTSPResult gst_rtsp_connection_send (GstRTSPConnection *conn, GstRTSPMessage *message,
|
|
GTimeVal *timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_send_messages_usec)
|
|
GstRTSPResult gst_rtsp_connection_send_messages (GstRTSPConnection *conn, GstRTSPMessage *messages, guint n_messages,
|
|
GTimeVal *timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_receive_usec)
|
|
GstRTSPResult gst_rtsp_connection_receive (GstRTSPConnection *conn, GstRTSPMessage *message,
|
|
GTimeVal *timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_poll_usec)
|
|
GstRTSPResult gst_rtsp_connection_poll (GstRTSPConnection *conn, GstRTSPEvent events,
|
|
GstRTSPEvent *revents, GTimeVal *timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_connection_next_timeout_usec)
|
|
GstRTSPResult gst_rtsp_connection_next_timeout (GstRTSPConnection *conn, GTimeVal *timeout);
|
|
|
|
GST_RTSP_DEPRECATED_FOR (gst_rtsp_watch_wait_backlog_usec)
|
|
GstRTSPResult gst_rtsp_watch_wait_backlog (GstRTSPWatch * watch,
|
|
GTimeVal *timeout);
|
|
|
|
G_GNUC_END_IGNORE_DEPRECATIONS
|
|
|
|
#endif /* GST_DISABLE_DEPRECATED */
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_CONNECTION_H__ */
|