[asterisk-commits] file: branch 13 r423209 - in /branches/13: ./ res/res_rtp_asterisk.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Sep 16 15:35:36 CDT 2014
Author: file
Date: Tue Sep 16 15:35:34 2014
New Revision: 423209
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423209
Log:
res_rtp_asterisk: Fix building when pjproject is not used.
........
Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12
Modified:
branches/13/ (props changed)
branches/13/res/res_rtp_asterisk.c
Propchange: branches/13/
------------------------------------------------------------------------------
Binary property 'branch-12-merged' - no diff available.
Modified: branches/13/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/res/res_rtp_asterisk.c?view=diff&rev=423209&r1=423208&r2=423209
==============================================================================
--- branches/13/res/res_rtp_asterisk.c (original)
+++ branches/13/res/res_rtp_asterisk.c Tue Sep 16 15:35:34 2014
@@ -2476,8 +2476,10 @@
static int ast_rtp_destroy(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+#ifdef HAVE_PJPROJECT
struct timeval wait = ast_tvadd(ast_tvnow(), ast_samp2tv(TURN_STATE_WAIT_TIME, 1000));
struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
+#endif
/* Destroy the smoother that was smoothing out audio if present */
if (rtp->smoother) {
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