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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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de7c72dec2
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.
87 lines
3.4 KiB
C
87 lines
3.4 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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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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#include <gst/gst.h>
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#include <gst/rtsp/gstrtsptransport.h>
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#ifndef __GST_RTSP_SESSION_MEDIA_H__
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#define __GST_RTSP_SESSION_MEDIA_H__
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G_BEGIN_DECLS
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#define GST_TYPE_RTSP_SESSION_MEDIA (gst_rtsp_session_media_get_type ())
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#define GST_IS_RTSP_SESSION_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_MEDIA))
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#define GST_IS_RTSP_SESSION_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_MEDIA))
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#define GST_RTSP_SESSION_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
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#define GST_RTSP_SESSION_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMedia))
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#define GST_RTSP_SESSION_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
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#define GST_RTSP_SESSION_MEDIA_CAST(obj) ((GstRTSPSessionMedia*)(obj))
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#define GST_RTSP_SESSION_MEDIA_CLASS_CAST(klass) ((GstRTSPSessionMediaClass*)(klass))
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typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
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typedef struct _GstRTSPSessionMediaClass GstRTSPSessionMediaClass;
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/**
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* GstRTSPSessionMedia:
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* @url: the url of the media
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* @media: the pipeline for the media
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* @state: the server state
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* @counter: counter for channels
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* @transports: array of #GstRTSPStreamTransport with the configuration
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* for the transport for each selected stream from @media.
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*
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* State of a client session regarding a specific media identified by uri.
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*/
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struct _GstRTSPSessionMedia
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{
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GObject parent;
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GstRTSPUrl *url;
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GstRTSPMedia *media;
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GstRTSPState state;
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guint counter;
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GPtrArray *transports;
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};
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struct _GstRTSPSessionMediaClass
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{
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GObjectClass parent_class;
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};
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GType gst_rtsp_session_media_get_type (void);
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GstRTSPSessionMedia * gst_rtsp_session_media_new (const GstRTSPUrl *url,
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GstRTSPMedia *media);
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/* control media */
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gboolean gst_rtsp_session_media_set_state (GstRTSPSessionMedia *media,
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GstState state);
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/* get stream transport config */
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GstRTSPStreamTransport * gst_rtsp_session_media_get_transport (GstRTSPSessionMedia *media,
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guint idx);
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gboolean gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia *media,
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GstRTSPRange *range);
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G_END_DECLS
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#endif /* __GST_RTSP_SESSION_MEDIA_H__ */
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