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Fix wrap around for rtp (bug #5595)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-11-11 04:07:03 +00:00
parent c6e7b2a333
commit 4606d74a66
1 changed files with 4 additions and 2 deletions
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6
rtp.c
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@@ -1098,6 +1098,8 @@ static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
/* Use previous txcore if available */
t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
ms = ast_tvdiff_ms(t, rtp->txcore);
if (ms < 0)
ms = 0;
/* Use what we just got for next time */
rtp->txcore = t;
return (unsigned int) ms;
@@ -1226,14 +1228,14 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
char iabuf[INET_ADDRSTRLEN];
int hdrlen = 12;
int res;
int ms;
unsigned int ms;
int pred;
int mark = 0;
ms = calc_txstamp(rtp, &f->delivery);
/* Default prediction */
if (f->subclass < AST_FORMAT_MAX_AUDIO) {
pred = rtp->lastts + f->samples;
pred = rtp->lastts + f->samples;
/* Re-calculate last TS */
rtp->lastts = rtp->lastts + ms * 8;