gstreamer/gst/rtsp-server/rtsp-stream-transport.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., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

#include <string.h>
#include <stdlib.h>

#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>

#include "rtsp-stream-transport.h"

enum
{
  PROP_0,
  PROP_LAST
};

GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
#define GST_CAT_DEFAULT rtsp_stream_transport_debug

static void gst_rtsp_stream_transport_finalize (GObject * obj);

G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
    G_TYPE_OBJECT);

static void
gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
{
  GObjectClass *gobject_class;

  gobject_class = G_OBJECT_CLASS (klass);

  gobject_class->finalize = gst_rtsp_stream_transport_finalize;

  GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
      0, "GstRTSPStreamTransport");
}

static void
gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
{
}

static void
gst_rtsp_stream_transport_finalize (GObject * obj)
{
  GstRTSPStreamTransport *trans;

  trans = GST_RTSP_STREAM_TRANSPORT (obj);

  /* remove callbacks now */
  gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
  gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);

  if (trans->transport)
    gst_rtsp_transport_free (trans->transport);

#if 0
  if (trans->rtpsource)
    g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
#endif

  G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
}

/**
 * gst_rtsp_stream_transport_new:
 * @stream: a #GstRTSPStream
 *
 * Create a new #GstRTSPStreamTransport that can be used for
 * @stream.
 *
 * Returns: a new #GstRTSPStreamTransport
 */
GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream)
{
  GstRTSPStreamTransport *trans;

  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);

  trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
  trans->stream = stream;

  return trans;
}

/**
 * gst_rtsp_stream_transport_set_callbacks:
 * @trans: a #GstRTSPStreamTransport
 * @send_rtp: (scope notified): a callback called when RTP should be sent
 * @send_rtcp: (scope notified): a callback called when RTCP should be sent
 * @user_data: user data passed to callbacks
 * @notify: called with the user_data when no longer needed.
 *
 * Install callbacks that will be called when data for a stream should be sent
 * to a client. This is usually used when sending RTP/RTCP over TCP.
 */
void
gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
    GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
    gpointer user_data, GDestroyNotify notify)
{
  trans->send_rtp = send_rtp;
  trans->send_rtcp = send_rtcp;
  if (trans->notify)
    trans->notify (trans->user_data);
  trans->user_data = user_data;
  trans->notify = notify;
}

/**
 * gst_rtsp_stream_transport_set_keepalive:
 * @trans: a #GstRTSPStreamTransport
 * @keep_alive: a callback called when the receiver is active
 * @user_data: user data passed to callback
 * @notify: called with the user_data when no longer needed.
 *
 * Install callbacks that will be called when RTCP packets are received from the
 * receiver of @trans.
 */
void
gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
    GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
{
  trans->keep_alive = keep_alive;
  if (trans->ka_notify)
    trans->ka_notify (trans->ka_user_data);
  trans->ka_user_data = user_data;
  trans->ka_notify = notify;
}


/**
 * gst_rtsp_stream_transport_set_transport:
 * @trans: a #GstRTSPStreamTransport
 * @ct: a client #GstRTSPTransport
 *
 * Set @ct as the client transport and create and return a matching server
 * transport. This function takes ownership of the passed @ct.
 *
 * Returns: a server transport or NULL if something went wrong. Use
 * gst_rtsp_transport_free () after usage.
 */
GstRTSPTransport *
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
    GstRTSPTransport * ct)
{
  GstRTSPTransport *st;

  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
  g_return_val_if_fail (ct != NULL, NULL);

  /* prepare the server transport */
  gst_rtsp_transport_new (&st);

  st->trans = ct->trans;
  st->profile = ct->profile;
  st->lower_transport = ct->lower_transport;

  switch (st->lower_transport) {
    case GST_RTSP_LOWER_TRANS_UDP:
      st->client_port = ct->client_port;
      st->server_port = trans->stream->server_port;
      break;
    case GST_RTSP_LOWER_TRANS_UDP_MCAST:
      ct->port = st->port = trans->stream->server_port;
      st->destination = g_strdup (ct->destination);
      st->ttl = ct->ttl;
      break;
    case GST_RTSP_LOWER_TRANS_TCP:
      st->interleaved = ct->interleaved;
    default:
      break;
  }

  if (trans->stream->session)
    g_object_get (trans->stream->session, "internal-ssrc", &st->ssrc, NULL);

  /* keep track of the transports in the stream. */
  if (trans->transport)
    gst_rtsp_transport_free (trans->transport);
  trans->transport = ct;

  return st;
}