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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f0c047ef94
Add the first stage in monitoring the rtp manager. Make sure we don't update the state to something we don't want.
188 lines
5.8 KiB
C
188 lines
5.8 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_MEDIA_H__
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#define __GST_RTSP_MEDIA_H__
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G_BEGIN_DECLS
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/* types for the media */
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#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
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#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
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#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
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#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
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#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
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#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
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#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
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#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
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typedef struct _GstRTSPMediaStream GstRTSPMediaStream;
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typedef struct _GstRTSPMedia GstRTSPMedia;
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typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
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typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
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/**
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* GstRTSPMediaTrans:
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* @idx: a stream index
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* @transport: a transport description
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*
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* A Transport description for stream @idx
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*/
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struct _GstRTSPMediaTrans {
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guint idx;
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GstRTSPTransport *transport;
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};
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/**
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* GstRTSPMediaStream:
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*
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* @media: the owner #GstRTSPMedia
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* @srcpad: the srcpad of the stream
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* @payloader: the payloader of the format
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* @prepared: if the stream is prepared for streaming
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* @server_port: the server udp ports
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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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* @recv_rtp_src: srcpad for RTP buffers
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* @recv_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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* @caps_sig: the signal id for detecting caps
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* @caps: the caps of the stream
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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 _GstRTSPMediaStream {
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GstPad *srcpad;
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GstElement *payloader;
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gboolean prepared;
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/* pads on the rtpbin */
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GstPad *recv_rtcp_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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/* 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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};
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/**
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* GstRTSPMedia:
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* @shared: if this media can be shared between clients
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* @element: the data providing element
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* @stream: the different streams provided by @element
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* @prepared: if the media is prepared for streaming
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* @pipeline: the toplevel pipeline
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* @rtpbin: the rtpbin
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* @multifdsink: multifdsink element for TCP transport
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*
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* A class that contains the GStreamer element along with a list of
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* #GstRTSPediaStream objects that can produce data.
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*
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* This object is usually created from a #GstRTSPMediaFactory.
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*/
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struct _GstRTSPMedia {
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GObject parent;
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gboolean shared;
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gboolean complete;
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GstElement *element;
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GArray *streams;
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gboolean prepared;
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/* the pipeline for the media */
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GstElement *pipeline;
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GSource *source;
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guint id;
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gboolean is_live;
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gboolean buffering;
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GstState target_state;
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/* RTP session manager */
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GstElement *rtpbin;
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/* for TCP transport */
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GstElement *multifdsink;
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/* the range of media */
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GstRTSPTimeRange range;
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};
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/**
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* GstRTSPMediaClass:
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* @context: the main context for dispatching messages
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* @loop: the mainloop for message.
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* @thread: the thread dispatching messages.
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* @handle_message: handle a message
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*
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* The RTSP media class
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*/
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struct _GstRTSPMediaClass {
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GObjectClass parent_class;
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/* thread for the mainloop */
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GMainContext *context;
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GMainLoop *loop;
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GThread *thread;
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gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
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};
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GType gst_rtsp_media_get_type (void);
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/* creating the media */
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GstRTSPMedia * gst_rtsp_media_new (void);
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void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
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gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
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/* prepare the media for playback */
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gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
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/* dealing with the media */
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guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
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GstRTSPMediaStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
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gboolean gst_rtsp_media_play (GstRTSPMedia *media, GArray *trans);
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gboolean gst_rtsp_media_pause (GstRTSPMedia *media, GArray *trans);
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gboolean gst_rtsp_media_stop (GstRTSPMedia *media, GArray *trans);
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G_END_DECLS
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#endif /* __GST_RTSP_MEDIA_H__ */
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