gstreamer/gst/rtsp-server/rtsp-stream-transport.h
Wim Taymans de7c72dec2 rtsp: massive refactoring
Make GObjects from the remaining simple structures.
Remove GstRTSPSessionStream, it's not needed.
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
  a GstRTSPStream should be transported to a client.
Rename GstRTSPMediaFactory::get_element -> create_element because that
  more accurately describes what it does.
Make nice methods instead of poking in the structures.
Move some methods inside the relevant object source code.
Use GPtrArray to store objects instead of plain arrays, it is more
  natural and allows us to more easily clean up.
Move the allocation of udp ports to the Stream object. The Stream object
  contains the elements needed to stream the media to a client.
Improve the prepare and unprepare methods. Unprepare should now undo
  everything prepare did. Improve also async unprepare when doing EOS on
  shutdown. Make sure we always unprepare correctly.
2012-10-25 21:29:58 +02:00

111 lines
4.7 KiB
C

/* 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
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
#include <gst/rtsp/gstrtsprange.h>
#include <gst/rtsp/gstrtspurl.h>
#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
#define __GST_RTSP_STREAM_TRANSPORT_H__
G_BEGIN_DECLS
/* types for the media */
#define GST_TYPE_RTSP_STREAM_TRANSPORT (gst_rtsp_stream_transport_get_type ())
#define GST_IS_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
#define GST_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
#include "rtsp-stream.h"
typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
/**
* GstRTSPStreamTransport:
* @parent: parent instance
* @idx: the stream index
* @send_rtp: callback for sending RTP messages
* @send_rtcp: callback for sending RTCP messages
* @send_rtp_list: callback for sending RTP messages
* @send_rtcp_list: callback for sending RTCP messages
* @user_data: user data passed in the callbacks
* @notify: free function for the user_data.
* @keep_alive: keep alive callback
* @ka_user_data: data passed to @keep_alive
* @ka_notify: called when @ka_user_data is freed
* @active: if we are actively sending
* @timeout: if we timed out
* @transport: a transport description
* @rtpsource: the receiver rtp source object
*
* A Transport description for stream @idx
*/
struct _GstRTSPStreamTransport {
GObject parent;
GstRTSPStream *stream;
GstRTSPSendFunc send_rtp;
GstRTSPSendFunc send_rtcp;
gpointer user_data;
GDestroyNotify notify;
GstRTSPKeepAliveFunc keep_alive;
gpointer ka_user_data;
GDestroyNotify ka_notify;
gboolean active;
gboolean timeout;
GstRTSPTransport *transport;
GObject *rtpsource;
};
struct _GstRTSPStreamTransportClass {
GObjectClass parent_class;
};
GType gst_rtsp_stream_transport_get_type (void);
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream);
GstRTSPTransport * gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
GstRTSPTransport * ct);
void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
GstRTSPSendFunc send_rtp,
GstRTSPSendFunc send_rtcp,
gpointer user_data,
GDestroyNotify notify);
void gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
GstRTSPKeepAliveFunc keep_alive,
gpointer user_data,
GDestroyNotify notify);
G_END_DECLS
#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */