mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-03 21:25:54 +00:00
0f38451f20
Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: * sys/oss/gstosssink.c: (gst_oss_sink_open), (gst_oss_sink_prepare), (gst_oss_sink_unprepare): Small documentation updates.
120 lines
2.9 KiB
C
120 lines
2.9 KiB
C
/* GStreamer
|
|
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
|
*
|
|
* 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., 59 Temple Place - Suite 330,
|
|
* Boston, MA 02111-1307, USA.
|
|
*/
|
|
|
|
|
|
#ifndef __GST_RTSPSRC_H__
|
|
#define __GST_RTSPSRC_H__
|
|
|
|
#include <gst/gst.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#include "gstrtsp.h"
|
|
#include "rtsp.h"
|
|
|
|
#define GST_TYPE_RTSPSRC \
|
|
(gst_rtspsrc_get_type())
|
|
#define GST_RTSPSRC(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc))
|
|
#define GST_RTSPSRC_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass))
|
|
#define GST_IS_RTSPSRC(obj) \
|
|
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC))
|
|
#define GST_IS_RTSPSRC_CLASS(klass) \
|
|
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC))
|
|
|
|
typedef struct _GstRTSPSrc GstRTSPSrc;
|
|
typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
|
|
|
|
/**
|
|
* GstRTSPProto:
|
|
* @GST_RTSP_PROTO_UDP_UNICAST: Use unicast UDP transfer.
|
|
* @GST_RTSP_PROTO_UDP_MULTICAST: Use multicast UDP transfer
|
|
* @GST_RTSP_PROTO_TCP: Use TCP transfer.
|
|
*
|
|
* Flags with allowed protocols for the datatransfer.
|
|
*/
|
|
typedef enum
|
|
{
|
|
GST_RTSP_PROTO_UDP_UNICAST = (1 << 0),
|
|
GST_RTSP_PROTO_UDP_MULTICAST = (1 << 1),
|
|
GST_RTSP_PROTO_TCP = (1 << 2),
|
|
} GstRTSPProto;
|
|
|
|
typedef struct _GstRTSPStream GstRTSPStream;
|
|
|
|
struct _GstRTSPStream {
|
|
gint id;
|
|
|
|
GstRTSPSrc *parent;
|
|
|
|
GstFlowReturn last_ret;
|
|
|
|
/* for interleaved mode */
|
|
gint rtpchannel;
|
|
gint rtcpchannel;
|
|
GstCaps *caps;
|
|
|
|
/* our udp sources for RTP */
|
|
GstElement *rtpsrc;
|
|
GstElement *rtcpsrc;
|
|
|
|
/* our udp sink back to the server */
|
|
GstElement *rtcpsink;
|
|
|
|
/* the RTP decoder */
|
|
GstElement *rtpdec;
|
|
GstPad *rtpdecrtp;
|
|
GstPad *rtpdecrtcp;
|
|
};
|
|
|
|
struct _GstRTSPSrc {
|
|
GstElement element;
|
|
|
|
/* task and mutex for interleaved mode */
|
|
gboolean interleaved;
|
|
GstTask *task;
|
|
GStaticRecMutex *stream_rec_lock;
|
|
|
|
gint numstreams;
|
|
GList *streams;
|
|
|
|
gchar *location;
|
|
gboolean debug;
|
|
guint retry;
|
|
|
|
GstRTSPProto protocols;
|
|
/* supported options */
|
|
gint options;
|
|
|
|
RTSPConnection *connection;
|
|
RTSPMessage *request;
|
|
RTSPMessage *response;
|
|
};
|
|
|
|
struct _GstRTSPSrcClass {
|
|
GstElementClass parent_class;
|
|
};
|
|
|
|
GType gst_rtspsrc_get_type(void);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSPSRC_H__ */
|