mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-15 04:46:32 +00:00
201 lines
6.7 KiB
C
201 lines
6.7 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>
|
|
#include <gst/rtsp/gstrtsp-enumtypes.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 (RFC 3551)
|
|
* @GST_RTSP_PROFILE_SAVP: the secure Audio/Visual profile (RFC 3711)
|
|
* @GST_RTSP_PROFILE_AVPF: the Audio/Visual profile with feedback (RFC 4585)
|
|
* @GST_RTSP_PROFILE_SAVPF: the secure Audio/Visual profile with feedback (RFC 5124)
|
|
*
|
|
* The transfer profile to use.
|
|
*/
|
|
/* FIXME 2.0: This should probably be an enum, not flags and maybe be replaced
|
|
* by GstRTPTransport */
|
|
typedef enum {
|
|
GST_RTSP_PROFILE_UNKNOWN = 0,
|
|
GST_RTSP_PROFILE_AVP = (1 << 0),
|
|
GST_RTSP_PROFILE_SAVP = (1 << 1),
|
|
GST_RTSP_PROFILE_AVPF = (1 << 2),
|
|
GST_RTSP_PROFILE_SAVPF = (1 << 3),
|
|
} 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.
|
|
* @GST_RTSP_LOWER_TRANS_TLS: encrypt TCP and HTTP with TLS
|
|
*
|
|
* 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),
|
|
GST_RTSP_LOWER_TRANS_TLS = (1 << 5)
|
|
} GstRTSPLowerTrans;
|
|
|
|
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. For TCP
|
|
* based transports, applications can use this field to store the
|
|
* sender and receiver ports of the client.
|
|
* @server_port: the server port pair for receiving data. For TCP
|
|
* based transports, applications can use this field to store the
|
|
* sender and receiver ports of the server.
|
|
* @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;
|
|
GstRTSPRange port;
|
|
|
|
/* UDP/TCP specific */
|
|
GstRTSPRange client_port;
|
|
GstRTSPRange server_port;
|
|
/* RTP specific */
|
|
guint ssrc;
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_transport_init (GstRTSPTransport *transport);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_transport_parse (const gchar *str, GstRTSPTransport *transport);
|
|
|
|
GST_RTSP_API
|
|
gchar* gst_rtsp_transport_as_text (GstRTSPTransport *transport);
|
|
|
|
GST_RTSP_DEPRECATED_FOR(gst_rtsp_transport_get_media_type)
|
|
GstRTSPResult gst_rtsp_transport_get_mime (GstRTSPTransMode trans, const gchar **mime);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_transport_get_manager (GstRTSPTransMode trans, const gchar **manager, guint option);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_transport_get_media_type (GstRTSPTransport *transport,
|
|
const gchar **media_type);
|
|
|
|
GST_RTSP_API
|
|
GstRTSPResult gst_rtsp_transport_free (GstRTSPTransport *transport);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_TRANSPORT_H__ */
|