gstreamer/gst/rtsp/gstrtspsrc.h

/* GStreamer
 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
 *               <2006> Wim Taymans <wim@fluendo.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.
 */
/*
 * Unless otherwise indicated, Source Code is licensed under MIT license.
 * See further explanation attached in License Statement (distributed in the file
 * LICENSE).
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy of
 * this software and associated documentation files (the "Software"), to deal in
 * the Software without restriction, including without limitation the rights to
 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
 * of the Software, and to permit persons to whom the Software is furnished to do
 * so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in all
 * copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
 * SOFTWARE.
 */

#ifndef __GST_RTSPSRC_H__
#define __GST_RTSPSRC_H__

#include <gst/gst.h>

G_BEGIN_DECLS

#include "gstrtsp.h"
#include "rtsp.h"
#include "rtspext.h"

#define GST_TYPE_RTSPSRC \
  (gst_rtspsrc_get_type())
#define GST_RTSPSRC(obj) \
  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc))
#define GST_RTSPSRC_CLASS(klass) \
  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass))
#define GST_IS_RTSPSRC(obj) \
  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC))
#define GST_IS_RTSPSRC_CLASS(klass) \
  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC))
#define GST_RTSPSRC_CAST(obj) \
  ((GstRTSPSrc *)(obj))

typedef struct _GstRTSPSrc GstRTSPSrc;
typedef struct _GstRTSPSrcClass GstRTSPSrcClass;

#define GST_RTSP_LOOP_GET_COND(rtsp)     (GST_RTSPSRC_CAST(rtsp)->loop_cond)
#define GST_RTSP_LOOP_WAIT(rtsp)         (g_cond_wait(GST_RTSP_LOOP_GET_COND (rtsp), GST_OBJECT_GET_LOCK (rtsp)))
#define GST_RTSP_LOOP_SIGNAL(rtsp)       (g_cond_signal(GST_RTSP_LOOP_GET_COND (rtsp)))

/**
 * GstRTSPProto:
 * @GST_RTSP_PROTO_UDP_UNICAST: Use unicast UDP transfer.
 * @GST_RTSP_PROTO_UDP_MULTICAST: Use multicast UDP transfer
 * @GST_RTSP_PROTO_TCP: Use TCP transfer.
 *
 * Flags with allowed protocols for the datatransfer.
 */
typedef enum
{
  GST_RTSP_PROTO_UDP_UNICAST    = (1 << 0),
  GST_RTSP_PROTO_UDP_MULTICAST  = (1 << 1),
  GST_RTSP_PROTO_TCP            = (1 << 2),
} GstRTSPProto;

typedef struct _GstRTSPStream GstRTSPStream;

struct _GstRTSPStream {
  gint          id;

  GstRTSPSrc   *parent;

  /* pad we expose or NULL when it does not have an actual pad */
  GstPad       *srcpad;
  GstFlowReturn last_ret;

  /* for interleaved mode */
  gint          rtpchannel;
  gint          rtcpchannel;
  GstCaps      *caps;

  /* our udp sources for RTP */
  GstElement   *rtpsrc;
  GstElement   *rtcpsrc;

  /* our udp sink back to the server */
  GstElement   *rtcpsink;

  /* the RTP decoder */
  GstElement   *rtpdec;
  GstPad       *rtpdecrtp;
  GstPad       *rtpdecrtcp;

  /* state */
  gint          pt;
  gboolean      container;
  gchar        *setup_url;
  guint32       ssrc; 
  guint32       seqbase;
};

struct _GstRTSPSrc {
  GstBin           parent;

  /* task and mutex for interleaved mode */
  gboolean         interleaved;
  GstTask         *task;
  GStaticRecMutex *stream_rec_lock;
  GstSegment       segment;
  gboolean         running;

  /* cond to signal loop */
  GCond           *loop_cond;
  gint             loop_cmd;

  gint             numstreams;
  GList           *streams;
  GstStructure    *props;

  gchar           *location;
  RTSPUrl         *url;
  gchar           *content_base;
  GstRTSPProto     protocols;
  gboolean         debug;
  guint   	   retry;
  guint64          timeout;

  /* supported methods */
  gint             methods;

  RTSPConnection  *connection;
  RTSPMessage     *request;
  RTSPMessage     *response;

  RTSPExtensionCtx *extension;
};

struct _GstRTSPSrcClass {
  GstBinClass parent_class;
};

GType gst_rtspsrc_get_type(void);

gboolean   gst_rtspsrc_send 		(GstRTSPSrc * src, RTSPMessage * request,
		   			 RTSPMessage * response, RTSPStatusCode * code);


G_END_DECLS

#endif /* __GST_RTSPSRC_H__ */