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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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65c5ecd270
Add a limit to the amount of queued bytes or messages we allow on the watch. API: GstRTSPConnection::gst_rtsp_watch_set_send_backlog() API: GstRTSPConnection::gst_rtsp_watch_get_send_backlog()
204 lines
9.9 KiB
C
204 lines
9.9 KiB
C
/* GStreamer
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* Copyright (C) <2005,2009> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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#ifndef __GST_RTSP_CONNECTION_H__
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#define __GST_RTSP_CONNECTION_H__
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#include <glib.h>
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#include <gst/gstconfig.h>
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#include <gst/rtsp/gstrtspdefs.h>
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#include <gst/rtsp/gstrtspurl.h>
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#include <gst/rtsp/gstrtspmessage.h>
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#include <gio/gio.h>
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G_BEGIN_DECLS
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/**
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* GstRTSPConnection:
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*
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* Opaque RTSP connection object.
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*/
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typedef struct _GstRTSPConnection GstRTSPConnection;
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/* opening/closing a connection */
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GstRTSPResult gst_rtsp_connection_create (const GstRTSPUrl *url, GstRTSPConnection **conn);
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GstRTSPResult gst_rtsp_connection_create_from_socket (GSocket * socket,
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const gchar * ip,
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guint16 port,
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const gchar * initial_buffer,
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GstRTSPConnection ** conn);
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GstRTSPResult gst_rtsp_connection_accept (GSocket *socket, GstRTSPConnection **conn, GCancellable *cancellable);
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GstRTSPResult gst_rtsp_connection_connect (GstRTSPConnection *conn, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_close (GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_free (GstRTSPConnection *conn);
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/* sending/receiving raw bytes */
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GstRTSPResult gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data,
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guint size, GTimeVal * timeout);
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GstRTSPResult gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
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guint size, GTimeVal * timeout);
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/* sending/receiving messages */
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GstRTSPResult gst_rtsp_connection_send (GstRTSPConnection *conn, GstRTSPMessage *message,
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GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_receive (GstRTSPConnection *conn, GstRTSPMessage *message,
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GTimeVal *timeout);
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/* status management */
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GstRTSPResult gst_rtsp_connection_poll (GstRTSPConnection *conn, GstRTSPEvent events,
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GstRTSPEvent *revents, GTimeVal *timeout);
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/* reset the timeout */
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GstRTSPResult gst_rtsp_connection_next_timeout (GstRTSPConnection *conn, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_reset_timeout (GstRTSPConnection *conn);
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/* flushing state */
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GstRTSPResult gst_rtsp_connection_flush (GstRTSPConnection *conn, gboolean flush);
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/* HTTP proxy support */
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GstRTSPResult gst_rtsp_connection_set_proxy (GstRTSPConnection *conn,
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const gchar *host, guint port);
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/* configure authentication data */
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GstRTSPResult gst_rtsp_connection_set_auth (GstRTSPConnection *conn, GstRTSPAuthMethod method,
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const gchar *user, const gchar *pass);
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void gst_rtsp_connection_set_auth_param (GstRTSPConnection *conn,
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const gchar * param,
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const gchar *value);
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void gst_rtsp_connection_clear_auth_params (GstRTSPConnection *conn);
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/* configure DSCP */
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GstRTSPResult gst_rtsp_connection_set_qos_dscp (GstRTSPConnection *conn,
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guint qos_dscp);
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/* accessors */
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GstRTSPUrl * gst_rtsp_connection_get_url (const GstRTSPConnection *conn);
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const gchar * gst_rtsp_connection_get_ip (const GstRTSPConnection *conn);
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void gst_rtsp_connection_set_ip (GstRTSPConnection *conn, const gchar *ip);
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GSocket * gst_rtsp_connection_get_read_socket (const GstRTSPConnection *conn);
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GSocket * gst_rtsp_connection_get_write_socket (const GstRTSPConnection *conn);
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void gst_rtsp_connection_set_http_mode (GstRTSPConnection *conn,
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gboolean enable);
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/* tunneling */
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void gst_rtsp_connection_set_tunneled (GstRTSPConnection *conn, gboolean tunneled);
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gboolean gst_rtsp_connection_is_tunneled (const GstRTSPConnection *conn);
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const gchar * gst_rtsp_connection_get_tunnelid (const GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_do_tunnel (GstRTSPConnection *conn, GstRTSPConnection *conn2);
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/* async IO */
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/**
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* GstRTSPWatch:
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*
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* Opaque RTSP watch object that can be used for asynchronous RTSP
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* operations.
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*/
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typedef struct _GstRTSPWatch GstRTSPWatch;
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/**
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* GstRTSPWatchFuncs:
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* @message_received: callback when a message was received
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* @message_sent: callback when a message was sent
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* @closed: callback when the connection is closed
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* @error: callback when an error occured
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* @tunnel_start: a client started a tunneled connection. The tunnelid of the
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* connection must be saved.
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* @tunnel_complete: a client finished a tunneled connection. In this callback
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* you usually pair the tunnelid of this connection with the saved one using
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* gst_rtsp_connection_do_tunnel().
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* @error_full: callback when an error occured with more information than
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* the @error callback.
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* @tunnel_lost: callback when the post connection of a tunnel is closed.
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*
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* Callback functions from a #GstRTSPWatch.
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*/
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typedef struct {
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GstRTSPResult (*message_received) (GstRTSPWatch *watch, GstRTSPMessage *message,
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gpointer user_data);
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GstRTSPResult (*message_sent) (GstRTSPWatch *watch, guint id,
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gpointer user_data);
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GstRTSPResult (*closed) (GstRTSPWatch *watch, gpointer user_data);
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GstRTSPResult (*error) (GstRTSPWatch *watch, GstRTSPResult result,
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gpointer user_data);
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GstRTSPStatusCode (*tunnel_start) (GstRTSPWatch *watch, gpointer user_data);
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GstRTSPResult (*tunnel_complete) (GstRTSPWatch *watch, gpointer user_data);
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GstRTSPResult (*error_full) (GstRTSPWatch *watch, GstRTSPResult result,
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GstRTSPMessage *message, guint id,
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gpointer user_data);
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GstRTSPResult (*tunnel_lost) (GstRTSPWatch *watch, gpointer user_data);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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} GstRTSPWatchFuncs;
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GstRTSPWatch * gst_rtsp_watch_new (GstRTSPConnection *conn,
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GstRTSPWatchFuncs *funcs,
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gpointer user_data,
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GDestroyNotify notify);
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void gst_rtsp_watch_reset (GstRTSPWatch *watch);
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void gst_rtsp_watch_unref (GstRTSPWatch *watch);
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guint gst_rtsp_watch_attach (GstRTSPWatch *watch,
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GMainContext *context);
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void gst_rtsp_watch_set_send_backlog (GstRTSPWatch *watch,
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gsize bytes, guint messages);
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void gst_rtsp_watch_get_send_backlog (GstRTSPWatch *watch,
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gsize *bytes, guint *messages);
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GstRTSPResult gst_rtsp_watch_write_data (GstRTSPWatch *watch,
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const guint8 *data,
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guint size, guint *id);
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GstRTSPResult gst_rtsp_watch_send_message (GstRTSPWatch *watch,
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GstRTSPMessage *message,
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guint *id);
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G_END_DECLS
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#endif /* __GST_RTSP_CONNECTION_H__ */
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