gstreamer/gst-libs/gst/rtsp/gstrtsptransport.h
2012-11-03 23:05:09 +00:00

177 lines
5.8 KiB
C

/* GStreamer
* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.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_TRANSPORT_H__
#define __GST_RTSP_TRANSPORT_H__
#include <gst/gstconfig.h>
#include <gst/rtsp/gstrtspdefs.h>
G_BEGIN_DECLS
/**
* GstRTSPTransMode:
* @GST_RTSP_TRANS_UNKNOWN: invalid tansport mode
* @GST_RTSP_TRANS_RTP: transfer RTP data
* @GST_RTSP_TRANS_RDT: transfer RDT (RealMedia) data
*
* The transfer mode to use.
*/
typedef enum {
GST_RTSP_TRANS_UNKNOWN = 0,
GST_RTSP_TRANS_RTP = (1 << 0),
GST_RTSP_TRANS_RDT = (1 << 1)
} GstRTSPTransMode;
/**
* GstRTSPProfile:
* @GST_RTSP_PROFILE_UNKNOWN: invalid profile
* @GST_RTSP_PROFILE_AVP: the Audio/Visual profile
* @GST_RTSP_PROFILE_SAVP: the secure Audio/Visual profile
*
* The transfer profile to use.
*/
typedef enum {
GST_RTSP_PROFILE_UNKNOWN = 0,
GST_RTSP_PROFILE_AVP = (1 << 0),
GST_RTSP_PROFILE_SAVP = (1 << 1)
} GstRTSPProfile;
/**
* GstRTSPLowerTrans:
* @GST_RTSP_LOWER_TRANS_UNKNOWN: invalid transport flag
* @GST_RTSP_LOWER_TRANS_UDP: stream data over UDP
* @GST_RTSP_LOWER_TRANS_UDP_MCAST: stream data over UDP multicast
* @GST_RTSP_LOWER_TRANS_TCP: stream data over TCP
* @GST_RTSP_LOWER_TRANS_HTTP: stream data tunneled over HTTP.
*
* The different transport methods.
*/
typedef enum {
GST_RTSP_LOWER_TRANS_UNKNOWN = 0,
GST_RTSP_LOWER_TRANS_UDP = (1 << 0),
GST_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
GST_RTSP_LOWER_TRANS_TCP = (1 << 2),
GST_RTSP_LOWER_TRANS_HTTP = (1 << 4)
} GstRTSPLowerTrans;
#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
GType gst_rtsp_lower_trans_get_type (void);
typedef struct _GstRTSPRange GstRTSPRange;
typedef struct _GstRTSPTransport GstRTSPTransport;
/**
* GstRTSPRange:
* @min: minimum value of the range
* @max: maximum value of the range
*
* A type to specify a range.
*/
struct _GstRTSPRange {
gint min;
gint max;
};
/**
* GstRTSPTransport:
* @trans: the transport mode
* @profile: the tansport profile
* @lower_transport: the lower transport
* @destination: the destination ip/hostname
* @source: the source ip/hostname
* @layers: the number of layers
* @mode_play: if play mode was selected
* @mode_record: if record mode was selected
* @append: is append mode was selected
* @interleaved: the interleave range
* @ttl: the time to live for multicast UDP
* @port: the port pair for multicast sessions
* @client_port: the client port pair for receiving data
* @server_port: the server port pair for receiving data
* @ssrc: the ssrc that the sender/receiver will use
*
* A structure holding the RTSP transport values.
*/
struct _GstRTSPTransport {
GstRTSPTransMode trans;
GstRTSPProfile profile;
GstRTSPLowerTrans lower_transport;
gchar *destination;
gchar *source;
guint layers;
gboolean mode_play;
gboolean mode_record;
gboolean append;
GstRTSPRange interleaved;
/* multicast specific */
guint ttl;
/* UDP specific */
GstRTSPRange port;
GstRTSPRange client_port;
GstRTSPRange server_port;
/* RTP specific */
guint ssrc;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport);
GstRTSPResult gst_rtsp_transport_init (GstRTSPTransport *transport);
GstRTSPResult gst_rtsp_transport_parse (const gchar *str, GstRTSPTransport *transport);
gchar* gst_rtsp_transport_as_text (GstRTSPTransport *transport);
GstRTSPResult gst_rtsp_transport_get_mime (GstRTSPTransMode trans, const gchar **mime);
GstRTSPResult gst_rtsp_transport_get_manager (GstRTSPTransMode trans, const gchar **manager, guint option);
GstRTSPResult gst_rtsp_transport_free (GstRTSPTransport *transport);
G_END_DECLS
#endif /* __GST_RTSP_TRANSPORT_H__ */