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6eedcfbc8c
Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps): Add pads after setting them up. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_combine_flows), (gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: Fix interleaved mode. - Protect streaming with lock. - Combine flows - set caps on outgoing buffers. - strip trailing \0 from data packets. - Configure RTP/RTCP in stream. Use DEBUG_OBJECT more.
112 lines
2.7 KiB
C
112 lines
2.7 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_RTSPSRC_H__
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#define __GST_RTSPSRC_H__
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#include <gst/gst.h>
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G_BEGIN_DECLS
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#include "gstrtsp.h"
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#include "rtsp.h"
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#define GST_TYPE_RTSPSRC \
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(gst_rtspsrc_get_type())
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#define GST_RTSPSRC(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc))
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#define GST_RTSPSRC_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass))
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#define GST_IS_RTSPSRC(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC))
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#define GST_IS_RTSPSRC_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC))
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typedef struct _GstRTSPSrc GstRTSPSrc;
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typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
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/* flags with allowed protocols */
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typedef enum
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{
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GST_RTSP_PROTO_UDP_UNICAST = (1 << 0),
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GST_RTSP_PROTO_UDP_MULTICAST = (1 << 1),
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GST_RTSP_PROTO_TCP = (1 << 2),
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} GstRTSPProto;
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typedef struct _GstRTSPStream GstRTSPStream;
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struct _GstRTSPStream {
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gint id;
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GstRTSPSrc *parent;
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GstFlowReturn last_ret;
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/* for interleaved mode */
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gint rtpchannel;
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gint rtcpchannel;
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GstCaps *caps;
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/* our udp sources for RTP */
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GstElement *rtpsrc;
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GstElement *rtcpsrc;
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/* our udp sink back to the server */
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GstElement *rtcpsink;
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/* the RTP decoder */
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GstElement *rtpdec;
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GstPad *rtpdecrtp;
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GstPad *rtpdecrtcp;
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};
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struct _GstRTSPSrc {
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GstElement element;
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/* task and mutex for interleaved mode */
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gboolean interleaved;
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GstTask *task;
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GStaticRecMutex *stream_rec_lock;
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gint numstreams;
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GList *streams;
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gchar *location;
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gboolean debug;
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guint retry;
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GstRTSPProto protocols;
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/* supported options */
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gint options;
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RTSPConnection *connection;
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RTSPMessage *request;
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RTSPMessage *response;
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};
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struct _GstRTSPSrcClass {
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GstElementClass parent_class;
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};
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GType gst_rtspsrc_get_type(void);
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G_END_DECLS
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#endif /* __GST_RTSPSRC_H__ */
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