mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-25 03:01:03 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtsp-server/rtsp-media.h
Wim Taymans fde25cd9c3 rtsp-server: port some more to 0.11 
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.
2011-12-09 10:53:30 +01:00

313 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 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;
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 *appqueue[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;
};
/**
* 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
* @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
* #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;
guint buffer_size;
GstRTSPAuth *auth;
gchar *multicast_group;
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 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.
*
* 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);
void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
void gst_rtsp_media_set_multicast_group (GstRTSPMedia *media, const gchar * mc);
gchar * gst_rtsp_media_get_multicast_group (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