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gstreamer/gst/rtsp-server/rtsp-media.h
Wim Taymans c34f5d1c1a media: add signal for new streams 
This allows applications to listen for new streams and configure properties on
them, like the address pool.
2012-11-15 15:41:42 +01:00

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/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, 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 _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
#include "rtsp-stream.h"
#include "rtsp-auth.h"
#include "rtsp-address-pool.h"
/**
* GstRTSPMediaStatus:
* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
* @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
* shutdown.
* @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_UNPREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARING = 2,
GST_RTSP_MEDIA_STATUS_PREPARED = 3,
GST_RTSP_MEDIA_STATUS_ERROR = 4
} GstRTSPMediaStatus;
/**
* GstRTSPMedia:
* @parent: parent GObject
* @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
* @eos_shutdown: if EOS should be sent on shutdown
* @buffer_size: The UDP buffer size
* @auth: the authentication service in use
* @multicast_group: the multicast group to use
* @element: the data providing element
* @streams: the different #GstRTSPStream provided by @element
* @dynamic: list of dynamic elements managed by @element
* @status: the status of the media pipeline
* @n_active: the number of active connections
* @adding: when elements are added to the pipeline
* @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
* @seekable: if the pipeline can perform a seek
* @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
* #GstRTSPStream 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;
guint buffer_size;
GstRTSPAuth *auth;
GstRTSPAddressPool*pool;
GstElement *element;
GRecMutex state_lock;
GPtrArray *streams;
GList *dynamic;
GstRTSPMediaStatus status;
gint n_active;
gboolean adding;
/* the pipeline for the media */
GstElement *pipeline;
GstElement *fakesink;
GSource *source;
guint id;
gboolean is_live;
gboolean seekable;
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 and removes all elements.
*
* 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 (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
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);
void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
/* prepare the media for playback */
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
/* creating streams */
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
GstElement *payloader,
GstPad *srcpad);
/* dealing with the media */
guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
GstRTSPStream * 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);
gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
GPtrArray *transports);
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */
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