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a900866570
If we have a new enough multiudpsink with the send-duplicates property, use this instead of doing our own filtering. Our custom filtering code should eventually be removed when we can depend on a released -good.
288 lines
9.6 KiB
C
288 lines
9.6 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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typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
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typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
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/**
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* GstRTSPMediaTrans:
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* @idx: a stream index
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* @send_rtp: callback for sending RTP messages
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* @send_rtcp: callback for sending RTCP messages
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* @user_data: user data passed in the callbacks
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* @notify: free function for the user_data.
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* @keep_alive: keep alive callback
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* @ka_user_data: data passed to @keep_alive
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* @ka_notify: called when @ka_user_data is freed
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* @active: if we are actively sending
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* @timeout: if we timed out
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* @transport: a transport description
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* @rtpsource: the receiver rtp source object
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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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GstRTSPSendFunc send_rtp;
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GstRTSPSendFunc send_rtcp;
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gpointer user_data;
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GDestroyNotify notify;
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GstRTSPKeepAliveFunc keep_alive;
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gpointer ka_user_data;
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GDestroyNotify ka_notify;
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gboolean active;
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gboolean timeout;
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GstRTSPTransport *transport;
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GObject *rtpsource;
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};
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/**
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* GstRTSPMediaStream:
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*
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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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* @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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* @tranports: the current transports being streamed
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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 *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 *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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GList *transports;
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/* to filter out duplicate destinations in case multiudpsink is too old to do
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* this for us */
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gboolean filter_duplicates;
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GList *destinations;
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};
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/**
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* GstRTSPMediaStatus:
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* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
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* @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
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* @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
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* @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
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*
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* The state of the media pipeline.
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*/
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typedef enum {
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GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
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GST_RTSP_MEDIA_STATUS_PREPARING = 1,
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GST_RTSP_MEDIA_STATUS_PREPARED = 2,
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GST_RTSP_MEDIA_STATUS_ERROR = 3
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} GstRTSPMediaStatus;
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/**
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* GstRTSPMedia:
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* @lock: for protecting the object
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* @cond: for signaling the object
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* @shared: if this media can be shared between clients
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* @reusable: if this media can be reused after an unprepare
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* @protocols: the allowed lower transport for this stream
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* @reused: if this media has been reused
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* @is_ipv6: if this media is using ipv6
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* @element: the data providing element
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* @streams: the different streams provided by @element
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* @dynamic: list of dynamic elements managed by @element
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* @status: the status of the media pipeline
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* @active: the number of active connections
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* @pipeline: the toplevel pipeline
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* @fakesink: for making state changes async
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* @source: the bus watch for pipeline messages.
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* @id: the id of the watch
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* @is_live: if the pipeline is live
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* @buffering: if the pipeline is buffering
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* @target_state: the desired target state of the pipeline
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* @rtpbin: the rtpbin
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* @range: the range of the media being streamed
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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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GMutex *lock;
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GCond *cond;
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gboolean shared;
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gboolean reusable;
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GstRTSPLowerTrans protocols;
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gboolean reused;
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gboolean is_ipv6;
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GstElement *element;
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GArray *streams;
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GList *dynamic;
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GstRTSPMediaStatus status;
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gint active;
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/* the pipeline for the media */
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GstElement *pipeline;
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GstElement *fakesink;
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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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/* 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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* @unprepare: the default implementation sets the pipeline's state
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* to GST_STATE_NULL.
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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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/* vmethods */
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gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
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gboolean (*unprepare) (GstRTSPMedia *media);
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/* signals */
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gboolean (*unprepared) (GstRTSPMedia *media);
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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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void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
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gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
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void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
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GstRTSPLowerTrans gst_rtsp_media_get_protocols (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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gboolean gst_rtsp_media_is_prepared (GstRTSPMedia *media);
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gboolean gst_rtsp_media_unprepare (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_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
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GstFlowReturn gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer);
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GstFlowReturn gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer);
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gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *trans);
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void gst_rtsp_media_remove_elements (GstRTSPMedia *media);
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G_END_DECLS
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#endif /* __GST_RTSP_MEDIA_H__ */
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