[asterisk-commits] branch kpfleming/vldtmf r8960 -
/team/kpfleming/vldtmf/rtp.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Jan 31 14:21:08 MST 2006
Author: kpfleming
Date: Mon Jan 30 21:40:05 2006
New Revision: 8960
URL: http://svn.digium.com/view/asterisk?rev=8960&view=rev
Log:
typo fix
use proper address for RFC2833 event structure
Modified:
team/kpfleming/vldtmf/rtp.c
Modified: team/kpfleming/vldtmf/rtp.c
URL: http://svn.digium.com/view/asterisk/team/kpfleming/vldtmf/rtp.c?rev=8960&r1=8959&r2=8960&view=diff
==============================================================================
--- team/kpfleming/vldtmf/rtp.c (original)
+++ team/kpfleming/vldtmf/rtp.c Mon Jan 30 21:40:05 2006
@@ -23,7 +23,7 @@
*
* \author Mark Spencer <markster at digium.com>
*
- * \note RTP is deffined in RFC 3550.
+ * \note RTP is defined in RFC 3550.
*/
#include <stdio.h>
@@ -572,7 +572,7 @@
/* This is special in-band data that's not one of our codecs */
/* It's special -- rfc2833 process it */
if (rtp_debug_test_addr(&sin)) {
- struct rfc2833_event *event = (struct rfc2833_event *) buffer;
+ struct rfc2833_event *event = (struct rfc2833_event *) header;
ast_verbose("Got rfc2833 RTP packet from %s:%d (type %d, seq %d, ts %d, len %d, mark %d, event %08x, end %d, duration %d) \n",
ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr),
@@ -605,12 +605,11 @@
return f ? f : &null_frame;
}
- rtp->f.subclass = rtpPT.code;
+ rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO)
rtp->f.frametype = AST_FRAME_VOICE;
else
rtp->f.frametype = AST_FRAME_VIDEO;
- rtp->lastrxformat = rtp->f.subclass;
if (!rtp->lastrxts)
rtp->lastrxts = timestamp;
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