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Fix a bug that I just noticed in the RTP code. The calculation for setting the

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len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@105932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-03-05 01:52:18 +00:00
parent be005c60d6
commit d564404d73
3 changed files with 15 additions and 12 deletions
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2
main/rtp.c
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@@ -1313,7 +1313,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
rtp->f.ts = timestamp / 8;
rtp->f.len = rtp->f.samples / 8;
rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
} else {
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)