mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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de7c72dec2
Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
142 lines
5.2 KiB
C
142 lines
5.2 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/gstrtsprange.h>
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#include <gst/rtsp/gstrtspurl.h>
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#ifndef __GST_RTSP_STREAM_H__
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#define __GST_RTSP_STREAM_H__
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G_BEGIN_DECLS
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/* types for the media stream */
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#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
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#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
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#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
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#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
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#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
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#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
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#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
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#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
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typedef struct _GstRTSPStream GstRTSPStream;
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typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
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#include "rtsp-stream-transport.h"
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/**
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* GstRTSPStream:
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* @parent: the parent instance
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* @idx: the stream index
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* @srcpad: the srcpad of the stream
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* @payloader: the payloader of the format
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* @is_ipv6: should this stream be IPv6
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* @buffer_size: the UDP buffer size
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* @is_joined: if the stream is joined in a bin
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* @recv_rtp_sink: sinkpad for RTP buffers
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* @recv_rtcp_sink: sinkpad for RTCP buffers
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* @send_rtp_src: srcpad for RTP buffers
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* @send_rtcp_src: srcpad for RTCP buffers
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* @udpsrc: the udp source elements for RTP/RTCP
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* @udpsink: the udp sink elements for RTP/RTCP
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* @appsrc: the app source elements for RTP/RTCP
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* @appsink: the app sink elements for RTP/RTCP
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* @server_port: the server ports for this stream
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* @caps_sig: the signal id for detecting caps
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* @caps: the caps of the stream
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* @n_active: the number of active transports in @transports
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* @tranports: list of #GstStreamTransport being streamed to
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*
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* The definition of a media stream. The streams are identified by @id.
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*/
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struct _GstRTSPStream {
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GObject parent;
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guint idx;
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GstPad *srcpad;
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GstElement *payloader;
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gboolean is_ipv6;
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guint buffer_size;
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gboolean is_joined;
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/* pads on the rtpbin */
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GstPad *recv_rtcp_sink;
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GstPad *recv_rtp_sink;
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GstPad *send_rtp_sink;
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GstPad *send_rtp_src;
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GstPad *send_rtcp_src;
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/* the RTPSession object */
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GObject *session;
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/* sinks used for sending and receiving RTP and RTCP, they share
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* sockets */
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GstElement *udpsrc[2];
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GstElement *udpsink[2];
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/* for TCP transport */
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GstElement *appsrc[2];
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GstElement *appqueue[2];
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GstElement *appsink[2];
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GstElement *tee[2];
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GstElement *selector[2];
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/* server ports for sending/receiving */
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GstRTSPRange server_port;
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/* the caps of the stream */
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gulong caps_sig;
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GstCaps *caps;
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/* transports we stream to */
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guint n_active;
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GList *transports;
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};
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struct _GstRTSPStreamClass {
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GObjectClass parent_class;
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};
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GType gst_rtsp_stream_get_type (void);
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GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
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GstPad *srcpad);
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gboolean gst_rtsp_stream_join_bin (GstRTSPStream * stream,
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GstBin *bin, GstElement *rtpbin);
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gboolean gst_rtsp_stream_leave_bin (GstRTSPStream * stream,
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GstBin *bin, GstElement *rtpbin);
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gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
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guint *rtptime, guint * seq);
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GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
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GstBuffer *buffer);
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GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
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GstBuffer *buffer);
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gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
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GstRTSPStreamTransport *trans);
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gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
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GstRTSPStreamTransport *trans);
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G_END_DECLS
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#endif /* __GST_RTSP_STREAM_H__ */
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