gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP...

gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP packet not wait for preroll.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_dup_printf):
Use new basesink async property to make sparse RTCP packet not wait for
preroll.

ChangeLog

+2007-08-29  Wim Taymans  <wim.taymans@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
+	(gst_rtspsrc_dup_printf):
+	Use new basesink async property to make sparse RTCP packet not wait for
+	preroll.
+
 2007-08-27  Jan Schmidt  <thaytan@mad.scientist.com>
 
 	* gst/audiofx/Makefile.am:

gst/rtsp/gstrtspsrc.c

@@ -1924,6 +1924,8 @@ gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream,
 
   /* no sync needed */
   g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
+  /* no async state changes needed */
+  g_object_set (G_OBJECT (stream->udpsink), "async", FALSE, NULL);
 
   if (stream->udpsrc[1]) {
     /* configure socket, we give it the same UDP socket as the udpsrc for RTCP