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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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fde25cd9c3
Fix caps. Remove bufferlist stuff Update for new API. Add queue before appsink now that preroll-queue-len is gone. Update for request pad changes.
166 lines
6.6 KiB
C
166 lines
6.6 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_H__
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#define __GST_RTSP_SESSION_H__
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G_BEGIN_DECLS
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#define GST_TYPE_RTSP_SESSION (gst_rtsp_session_get_type ())
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#define GST_IS_RTSP_SESSION(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION))
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#define GST_IS_RTSP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION))
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#define GST_RTSP_SESSION_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
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#define GST_RTSP_SESSION(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSession))
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#define GST_RTSP_SESSION_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
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#define GST_RTSP_SESSION_CAST(obj) ((GstRTSPSession*)(obj))
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#define GST_RTSP_SESSION_CLASS_CAST(klass) ((GstRTSPSessionClass*)(klass))
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typedef struct _GstRTSPSession GstRTSPSession;
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typedef struct _GstRTSPSessionClass GstRTSPSessionClass;
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typedef struct _GstRTSPSessionStream GstRTSPSessionStream;
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typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
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#include "rtsp-media.h"
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/**
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* GstRTSPSessionStream:
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* @trans: the media transport
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* @media_stream: the controlled media stream
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*
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* Configuration of a stream. A stream is an audio or video stream related to a
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* media.
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*/
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struct _GstRTSPSessionStream
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{
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GstRTSPMediaTrans trans;
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/* the stream of the media */
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GstRTSPMediaStream *media_stream;
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};
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/**
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* GstRTSPSessionMedia:
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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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/* the url of the media */
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GstRTSPUrl *url;
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/* the pipeline for the media */
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GstRTSPMedia *media;
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/* the server state */
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GstRTSPState state;
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/* counter for channels */
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guint counter;
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/* configuration for the different streams */
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GArray *streams;
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};
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/**
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* GstRTSPSession:
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* @sessionid: the session id of the session
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* @timeout: the timeout of the session
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* @create_time: the time when the session was created
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* @last_access: the time the session was last accessed
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* @expire_count: the expire prevention counter
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* @media: a list of #GstRTSPSessionMedia managed in this session
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*
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* Session information kept by the server for a specific client.
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* One client session, identified with a session id, can handle multiple medias
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* identified with the url of a media.
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*/
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struct _GstRTSPSession {
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GObject parent;
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gchar *sessionid;
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guint timeout;
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GTimeVal create_time;
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GTimeVal last_access;
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gint expire_count;
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GList *medias;
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};
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struct _GstRTSPSessionClass {
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GObjectClass parent_class;
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};
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GType gst_rtsp_session_get_type (void);
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/* create a new session */
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GstRTSPSession * gst_rtsp_session_new (const gchar *sessionid);
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const gchar * gst_rtsp_session_get_sessionid (GstRTSPSession *session);
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void gst_rtsp_session_set_timeout (GstRTSPSession *session, guint timeout);
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guint gst_rtsp_session_get_timeout (GstRTSPSession *session);
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/* session timeout stuff */
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void gst_rtsp_session_touch (GstRTSPSession *session);
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void gst_rtsp_session_prevent_expire (GstRTSPSession *session);
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void gst_rtsp_session_allow_expire (GstRTSPSession *session);
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gint gst_rtsp_session_next_timeout (GstRTSPSession *session, GTimeVal *now);
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gboolean gst_rtsp_session_is_expired (GstRTSPSession *session, GTimeVal *now);
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/* handle media in a session */
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GstRTSPSessionMedia * gst_rtsp_session_manage_media (GstRTSPSession *sess,
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const GstRTSPUrl *uri,
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GstRTSPMedia *media);
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gboolean gst_rtsp_session_release_media (GstRTSPSession *sess,
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GstRTSPSessionMedia *media);
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/* get media in a session */
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GstRTSPSessionMedia * gst_rtsp_session_get_media (GstRTSPSession *sess,
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const GstRTSPUrl *url);
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/* control media */
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gboolean gst_rtsp_session_media_set_state (GstRTSPSessionMedia *media, GstState state);
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/* get stream config */
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GstRTSPSessionStream * gst_rtsp_session_media_get_stream (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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/* configure transport */
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GstRTSPTransport * gst_rtsp_session_stream_set_transport (GstRTSPSessionStream *stream,
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GstRTSPTransport *ct);
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void gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStream *stream,
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GstRTSPSendFunc send_rtp,
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GstRTSPSendFunc send_rtcp,
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gpointer user_data,
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GDestroyNotify notify);
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void gst_rtsp_session_stream_set_keepalive (GstRTSPSessionStream *stream,
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GstRTSPKeepAliveFunc keep_alive,
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gpointer user_data,
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GDestroyNotify notify);
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G_END_DECLS
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#endif /* __GST_RTSP_SESSION_H__ */
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