use existing sample size in frame instead of recalculating it (issue #5390)
authorKevin P. Fleming <kpfleming@digium.com>
Fri, 14 Oct 2005 00:03:41 +0000 (00:03 +0000)
committerKevin P. Fleming <kpfleming@digium.com>
Fri, 14 Oct 2005 00:03:41 +0000 (00:03 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6781 65c4cc65-6c06-0410-ace0-fbb531ad65f3

rtp.c

diff --git a/rtp.c b/rtp.c
index 78cdfb0..468445e 100755 (executable)
--- a/rtp.c
+++ b/rtp.c
@@ -1231,7 +1231,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
        ms = calc_txstamp(rtp, &f->delivery);
        /* Default prediction */
        if (f->subclass < AST_FORMAT_MAX_AUDIO) {
-                pred = rtp->lastts + ast_codec_get_samples(f);
+                pred = rtp->lastts + f->samples;
 
                /* Re-calculate last TS */
                rtp->lastts = rtp->lastts + ms * 8;