mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 02:46:33 +00:00
44a2855eb3
Add methods to get and set the address pool for the stream Add method to allocate and get the multicast addresses for this stream.
162 lines
6.1 KiB
C
162 lines
6.1 KiB
C
/* 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_STREAM_H__
|
|
#define __GST_RTSP_STREAM_H__
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
/* types for the media stream */
|
|
#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
|
|
#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
|
|
#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
|
|
#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
|
|
#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
|
|
#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
|
|
#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
|
|
#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
|
|
|
|
typedef struct _GstRTSPStream GstRTSPStream;
|
|
typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
|
|
|
|
#include "rtsp-stream-transport.h"
|
|
#include "rtsp-address-pool.h"
|
|
|
|
/**
|
|
* GstRTSPStream:
|
|
* @parent: the parent instance
|
|
* @lock: mutex protecting the stream
|
|
* @idx: the stream index
|
|
* @srcpad: the srcpad of the stream
|
|
* @payloader: the payloader of the format
|
|
* @is_ipv6: should this stream be IPv6
|
|
* @buffer_size: the UDP buffer size
|
|
* @is_joined: if the stream is joined in a bin
|
|
* @send_rtp_sink: sinkpad for sending RTP buffers
|
|
* @recv_sink: sinkpad for receiving RTP/RTCP buffers
|
|
* @send_src: srcpad for sending RTP/RTCP buffers
|
|
* @session: the RTP session object
|
|
* @udpsrc: the udp source elements for RTP/RTCP
|
|
* @udpsink: the udp sink elements for RTP/RTCP
|
|
* @appsrc: the app source elements for RTP/RTCP
|
|
* @appqueue: the app queue elements for RTP/RTCP
|
|
* @appsink: the app sink elements for RTP/RTCP
|
|
* @tee: tee for the sending to udpsink and appsink
|
|
* @funnel: tee for the receiving from udpsrc and appsrc
|
|
* @server_port: the server ports for this stream
|
|
* @pool: the address pool for this stream
|
|
* @addr: the address for this stream
|
|
* @caps_sig: the signal id for detecting caps
|
|
* @caps: the caps of the stream
|
|
* @n_active: the number of active transports in @transports
|
|
* @transports: list of #GstStreamTransport being streamed to
|
|
*
|
|
* The definition of a media stream. The streams are identified by @idx.
|
|
*/
|
|
struct _GstRTSPStream {
|
|
GObject parent;
|
|
|
|
GMutex lock;
|
|
guint idx;
|
|
GstPad *srcpad;
|
|
GstElement *payloader;
|
|
gboolean is_ipv6;
|
|
guint buffer_size;
|
|
gboolean is_joined;
|
|
|
|
/* pads on the rtpbin */
|
|
GstPad *send_rtp_sink;
|
|
GstPad *recv_sink[2];
|
|
GstPad *send_src[2];
|
|
|
|
/* 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 *funnel[2];
|
|
|
|
/* server ports for sending/receiving */
|
|
GstRTSPRange server_port;
|
|
|
|
/* multicast addresses */
|
|
GstRTSPAddressPool *pool;
|
|
GstRTSPAddress *addr;
|
|
|
|
/* the caps of the stream */
|
|
gulong caps_sig;
|
|
GstCaps *caps;
|
|
|
|
/* transports we stream to */
|
|
guint n_active;
|
|
GList *transports;
|
|
};
|
|
|
|
struct _GstRTSPStreamClass {
|
|
GObjectClass parent_class;
|
|
};
|
|
|
|
GType gst_rtsp_stream_get_type (void);
|
|
|
|
GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
|
|
GstPad *srcpad);
|
|
|
|
void gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu);
|
|
guint gst_rtsp_stream_get_mtu (GstRTSPStream * stream);
|
|
|
|
void gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
|
|
GstRTSPAddressPool *
|
|
gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
|
|
|
|
GstRTSPAddress * gst_rtsp_stream_get_address (GstRTSPStream *stream);
|
|
|
|
gboolean gst_rtsp_stream_join_bin (GstRTSPStream * stream,
|
|
GstBin *bin, GstElement *rtpbin,
|
|
GstState state);
|
|
gboolean gst_rtsp_stream_leave_bin (GstRTSPStream * stream,
|
|
GstBin *bin, GstElement *rtpbin);
|
|
|
|
gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
|
|
guint *rtptime, guint * seq);
|
|
|
|
GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
|
|
GstBuffer *buffer);
|
|
GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
|
|
GstBuffer *buffer);
|
|
|
|
gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
|
|
GstRTSPStreamTransport *trans);
|
|
gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
|
|
GstRTSPStreamTransport *trans);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_STREAM_H__ */
|