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

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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.
This commit is contained in:
Wim Taymans 2007-08-29 21:43:08 +00:00
parent 32621485d5
commit 14e218c083
2 changed files with 9 additions and 0 deletions
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7
ChangeLog
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@ -1,3 +1,10 @@
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:

2
gst/rtsp/gstrtspsrc.c
View file

@ -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