mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-28 12:41:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtsp-server/rtsp-media.h
Wim Taymans 7797023fda media: enable per factory authorisations 
Allow for adding a GstRTSPAuth on the factory and media level and check
permissions when accessing the factory.
Add hints to the auth methods for future more fine grained authorisation.
Add example application for per factory authentication.
2011-01-12 13:57:09 +01:00

309 lines
11 KiB
C
Raw Blame History

/* 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;
typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
typedef gboolean (*GstRTSPSendListFunc) (GstBufferList *blist, guint8 channel, gpointer user_data);
typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
/**
* GstRTSPMediaTrans:
* @idx: a 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 _GstRTSPMediaTrans {
guint idx;
GstRTSPSendFunc send_rtp;
GstRTSPSendFunc send_rtcp;
GstRTSPSendListFunc send_rtp_list;
GstRTSPSendListFunc send_rtcp_list;
gpointer user_data;
GDestroyNotify notify;
GstRTSPKeepAliveFunc keep_alive;
gpointer ka_user_data;
GDestroyNotify ka_notify;
gboolean active;
gboolean timeout;
GstRTSPTransport *transport;
GObject *rtpsource;
};
#include "rtsp-auth.h"
/**
* GstRTSPMediaStream:
* @srcpad: the srcpad of the stream
* @payloader: the payloader of the format
* @prepared: if the stream is prepared for streaming
* @recv_rtp_sink: sinkpad for RTP buffers
* @recv_rtcp_sink: sinkpad for RTCP buffers
* @send_rtp_src: srcpad for RTP buffers
* @send_rtcp_src: srcpad for RTCP buffers
* @udpsrc: the udp source elements for RTP/RTCP
* @udpsink: the udp sink elements for RTP/RTCP
* @appsrc: the app source elements for RTP/RTCP
* @appsink: the app sink elements for RTP/RTCP
* @server_port: the server ports for this stream
* @caps_sig: the signal id for detecting caps
* @caps: the caps of the stream
* @tranports: the current transports being streamed
*
* 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 *recv_rtp_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];
/* for TCP transport */
GstElement *appsrc[2];
GstElement *appsink[2];
GstElement *tee[2];
GstElement *selector[2];
/* server ports for sending/receiving */
GstRTSPRange server_port;
/* the caps of the stream */
gulong caps_sig;
GstCaps *caps;
/* transports we stream to */
GList *transports;
/* to filter out duplicate destinations in case multiudpsink is too old to do
* this for us */
gboolean filter_duplicates;
GList *destinations;
};
/**
* GstRTSPMediaStatus:
* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
* @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
* @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
* @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
*
* The state of the media pipeline.
*/
typedef enum {
GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
GST_RTSP_MEDIA_STATUS_PREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARED = 2,
GST_RTSP_MEDIA_STATUS_ERROR = 3
} GstRTSPMediaStatus;
/**
* GstRTSPMedia:
* @lock: for protecting the object
* @cond: for signaling the object
* @shared: if this media can be shared between clients
* @reusable: if this media can be reused after an unprepare
* @protocols: the allowed lower transport for this stream
* @reused: if this media has been reused
* @is_ipv6: if this media is using ipv6
* @element: the data providing element
* @streams: the different streams provided by @element
* @dynamic: list of dynamic elements managed by @element
* @status: the status of the media pipeline
* @active: the number of active connections
* @pipeline: the toplevel pipeline
* @fakesink: for making state changes async
* @source: the bus watch for pipeline messages.
* @id: the id of the watch
* @is_live: if the pipeline is live
* @buffering: if the pipeline is buffering
* @target_state: the desired target state of the pipeline
* @rtpbin: the rtpbin
* @range: the range of the media being streamed
*
* A class that contains the GStreamer element along with a list of
* #GstRTSPMediaStream objects that can produce data.
*
* This object is usually created from a #GstRTSPMediaFactory.
*/
struct _GstRTSPMedia {
GObject parent;
GMutex *lock;
GCond *cond;
gboolean shared;
gboolean reusable;
GstRTSPLowerTrans protocols;
gboolean reused;
gboolean is_ipv6;
gboolean eos_shutdown;
GstRTSPAuth *auth;
GstElement *element;
GArray *streams;
GList *dynamic;
GstRTSPMediaStatus status;
gint active;
gboolean eos_pending;
gboolean adding;
/* the pipeline for the media */
GstElement *pipeline;
GstElement *fakesink;
GSource *source;
guint id;
gboolean is_live;
gboolean buffering;
GstState target_state;
/* RTP session manager */
GstElement *rtpbin;
/* 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
* @unprepare: the default implementation sets the pipeline's state
* to GST_STATE_NULL.
*
* The RTSP media class
*/
struct _GstRTSPMediaClass {
GObjectClass parent_class;
/* thread for the mainloop */
GMainContext *context;
GMainLoop *loop;
GThread *thread;
/* vmethods */
gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
gboolean (*unprepare) (GstRTSPMedia *media);
/* signals */
gboolean (*prepared) (GstRTSPMedia *media);
gboolean (*unprepared) (GstRTSPMedia *media);
gboolean (*new_state) (GstRTSPMedia *media, GstState state);
};
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);
void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
void gst_rtsp_media_set_auth (GstRTSPMedia *media, GstRTSPAuth *auth);
GstRTSPAuth * gst_rtsp_media_get_auth (GstRTSPMedia *media);
/* prepare the media for playback */
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
gboolean gst_rtsp_media_is_prepared (GstRTSPMedia *media);
gboolean gst_rtsp_media_unprepare (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_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media, gboolean play);
GstFlowReturn gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer);
GstFlowReturn gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer);
gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transports);
void gst_rtsp_media_remove_elements (GstRTSPMedia *media);
void gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans *trans);
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */
Reference in a new issue View git blame Copy permalink