gstreamer/gst/rtsp-server/rtsp-media.h
Wim Taymans f0c047ef94 Add suport for RTP manager monitoring
Add the first stage in monitoring the rtp manager.
Make sure we don't update the state to something we don't want.
2009-02-13 19:54:18 +01:00

188 lines
5.8 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_MEDIA_H__
#define __GST_RTSP_MEDIA_H__
G_BEGIN_DECLS
/* types for the media */
#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
typedef struct _GstRTSPMediaStream GstRTSPMediaStream;
typedef struct _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
/**
* GstRTSPMediaTrans:
* @idx: a stream index
* @transport: a transport description
*
* A Transport description for stream @idx
*/
struct _GstRTSPMediaTrans {
guint idx;
GstRTSPTransport *transport;
};
/**
* GstRTSPMediaStream:
*
* @media: the owner #GstRTSPMedia
* @srcpad: the srcpad of the stream
* @payloader: the payloader of the format
* @prepared: if the stream is prepared for streaming
* @server_port: the server udp ports
* @recv_rtp_sink: sinkpad for RTP buffers
* @recv_rtcp_sink: sinkpad for RTCP buffers
* @recv_rtp_src: srcpad for RTP buffers
* @recv_rtcp_src: srcpad for RTCP buffers
* @udpsrc: the udp source elements for RTP/RTCP
* @udpsink: the udp sink elements for RTP/RTCP
* @caps_sig: the signal id for detecting caps
* @caps: the caps of the stream
*
* The definition of a media stream. The streams are identified by @id.
*/
struct _GstRTSPMediaStream {
GstPad *srcpad;
GstElement *payloader;
gboolean prepared;
/* pads on the rtpbin */
GstPad *recv_rtcp_sink;
GstPad *send_rtp_sink;
GstPad *send_rtp_src;
GstPad *send_rtcp_src;
/* the RTPSession object */
GObject *session;
/* sinks used for sending and receiving RTP and RTCP, they share
* sockets */
GstElement *udpsrc[2];
GstElement *udpsink[2];
/* server ports for sending/receiving */
GstRTSPRange server_port;
/* the caps of the stream */
gulong caps_sig;
GstCaps *caps;
};
/**
* GstRTSPMedia:
* @shared: if this media can be shared between clients
* @element: the data providing element
* @stream: the different streams provided by @element
* @prepared: if the media is prepared for streaming
* @pipeline: the toplevel pipeline
* @rtpbin: the rtpbin
* @multifdsink: multifdsink element for TCP transport
*
* A class that contains the GStreamer element along with a list of
* #GstRTSPediaStream objects that can produce data.
*
* This object is usually created from a #GstRTSPMediaFactory.
*/
struct _GstRTSPMedia {
GObject parent;
gboolean shared;
gboolean complete;
GstElement *element;
GArray *streams;
gboolean prepared;
/* the pipeline for the media */
GstElement *pipeline;
GSource *source;
guint id;
gboolean is_live;
gboolean buffering;
GstState target_state;
/* RTP session manager */
GstElement *rtpbin;
/* for TCP transport */
GstElement *multifdsink;
/* the range of media */
GstRTSPTimeRange range;
};
/**
* GstRTSPMediaClass:
* @context: the main context for dispatching messages
* @loop: the mainloop for message.
* @thread: the thread dispatching messages.
* @handle_message: handle a message
*
* The RTSP media class
*/
struct _GstRTSPMediaClass {
GObjectClass parent_class;
/* thread for the mainloop */
GMainContext *context;
GMainLoop *loop;
GThread *thread;
gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
};
GType gst_rtsp_media_get_type (void);
/* creating the media */
GstRTSPMedia * gst_rtsp_media_new (void);
void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
/* prepare the media for playback */
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
/* dealing with the media */
guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
GstRTSPMediaStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
gboolean gst_rtsp_media_play (GstRTSPMedia *media, GArray *trans);
gboolean gst_rtsp_media_pause (GstRTSPMedia *media, GArray *trans);
gboolean gst_rtsp_media_stop (GstRTSPMedia *media, GArray *trans);
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */