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I thought I was going to be able to leave 1.4 alone, but that was not the case.

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I ran into some problems with G.722 in 1.4, so I have merged in all of the fixes
in this area that I have made in trunk/1.6.0, and things are happy again.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-04-22 21:14:55 +00:00
parent e4ed5a76f1
commit b3a211bdc3
4 changed files with 60 additions and 18 deletions
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19
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 / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
rtp->f.len = rtp->f.samples / (ast_format_rate(rtp->f.subclass) / 1000);
} else {
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)
@@ -2795,17 +2795,28 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
ast_smoother_feed(rtp->smoother, _f);
}
while((f = ast_smoother_read(rtp->smoother)) && (f->data))
while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
if (f->subclass == AST_FORMAT_G722) {
/* G.722 is silllllllllllllly */
f->samples /= 2;
}
ast_rtp_raw_write(rtp, f, codec);
}
} else {
/* Don't buffer outgoing frames; send them one-per-packet: */
/* Don't buffer outgoing frames; send them one-per-packet: */
if (_f->offset < hdrlen) {
f = ast_frdup(_f);
} else {
f = _f;
}
if (f->data)
if (f->data) {
if (f->subclass == AST_FORMAT_G722) {
/* G.722 is silllllllllllllly */
f->samples /= 2;
}
ast_rtp_raw_write(rtp, f, codec);
}
if (f != _f)
ast_frfree(f);
}