gstreamer/gst/rtsp-server/rtsp-stream.h

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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/gst.h>
#include <gst/rtsp/gstrtsprange.h>
#include <gst/rtsp/gstrtspurl.h>
#ifndef __GST_RTSP_STREAM_H__
#define __GST_RTSP_STREAM_H__
G_BEGIN_DECLS
/* types for the media stream */
#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
typedef struct _GstRTSPStream GstRTSPStream;
typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
#include "rtsp-stream-transport.h"
#include "rtsp-address-pool.h"
/**
* GstRTSPStream:
* @parent: the parent instance
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* @lock: mutex protecting the stream
* @idx: the stream index
* @srcpad: the srcpad of the stream
* @payloader: the payloader of the format
* @is_ipv6: should this stream be IPv6
* @buffer_size: the UDP buffer size
* @is_joined: if the stream is joined in a bin
* @send_rtp_sink: sinkpad for sending RTP buffers
* @recv_sink: sinkpad for receiving RTP/RTCP buffers
* @send_src: srcpad for sending RTP/RTCP buffers
* @session: the RTP session object
* @udpsrc: the udp source elements for RTP/RTCP
* @udpsink: the udp sink elements for RTP/RTCP
* @appsrc: the app source elements for RTP/RTCP
* @appqueue: the app queue elements for RTP/RTCP
* @appsink: the app sink elements for RTP/RTCP
* @tee: tee for the sending to udpsink and appsink
* @funnel: tee for the receiving from udpsrc and appsrc
* @server_port: the server ports for this stream
* @pool: the address pool for this stream
* @addr: the address for this stream
* @caps_sig: the signal id for detecting caps
* @caps: the caps of the stream
* @n_active: the number of active transports in @transports
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* @transports: list of #GstStreamTransport being streamed to
*
* The definition of a media stream. The streams are identified by @idx.
*/
struct _GstRTSPStream {
GObject parent;
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GMutex lock;
guint idx;
GstPad *srcpad;
GstElement *payloader;
gboolean is_ipv6;
guint buffer_size;
gboolean is_joined;
/* pads on the rtpbin */
GstPad *send_rtp_sink;
GstPad *recv_sink[2];
GstPad *send_src[2];
/* the RTPSession object */
GObject *session;
/* sinks used for sending and receiving RTP and RTCP, they share
* sockets */
GstElement *udpsrc[2];
GstElement *udpsink[2];
/* for TCP transport */
GstElement *appsrc[2];
GstElement *appqueue[2];
GstElement *appsink[2];
GstElement *tee[2];
GstElement *funnel[2];
/* server ports for sending/receiving */
GstRTSPRange server_port;
/* multicast addresses */
GstRTSPAddressPool *pool;
GstRTSPAddress *addr;
/* the caps of the stream */
gulong caps_sig;
GstCaps *caps;
/* transports we stream to */
guint n_active;
GList *transports;
};
struct _GstRTSPStreamClass {
GObjectClass parent_class;
};
GType gst_rtsp_stream_get_type (void);
GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
GstPad *srcpad);
void gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu);
guint gst_rtsp_stream_get_mtu (GstRTSPStream * stream);
void gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
GstRTSPAddressPool *
gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
GstRTSPAddress * gst_rtsp_stream_get_address (GstRTSPStream *stream);
gboolean gst_rtsp_stream_join_bin (GstRTSPStream * stream,
GstBin *bin, GstElement *rtpbin,
GstState state);
gboolean gst_rtsp_stream_leave_bin (GstRTSPStream * stream,
GstBin *bin, GstElement *rtpbin);
gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
guint *rtptime, guint * seq);
GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
GstBuffer *buffer);
GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
GstBuffer *buffer);
gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
GstRTSPStreamTransport *trans);
gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
GstRTSPStreamTransport *trans);
G_END_DECLS
#endif /* __GST_RTSP_STREAM_H__ */