diff options
| author | David Vossel <dvossel@digium.com> | 2009-07-08 23:19:09 +0000 |
|---|---|---|
| committer | David Vossel <dvossel@digium.com> | 2009-07-08 23:19:09 +0000 |
| commit | ba2a8457b85dbfd2cf85a3177fa23fbaf9942c9b (patch) | |
| tree | 8536e6e448e2c0309565c638cb5b35c41dd1f7b7 /channels | |
| parent | a6380004519dd3239b3ad86dee48ea73628f625b (diff) | |
Merged revisions 205471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines
Fixes 8khz assumptions
Many calculations assume 8khz is the codec rate. This
is not always the case. This patch only addresses chan_iax.c
and res_rtp_asterisk.c, but I am sure there are other areas
that make this assumption as well.
Review: https://reviewboard.asterisk.org/r/306/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_iax2.c | 15 |
1 files changed, 8 insertions, 7 deletions
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c index ead0f8f39..ed5db44b4 100644 --- a/channels/chan_iax2.c +++ b/channels/chan_iax2.c @@ -3225,7 +3225,7 @@ static void __get_from_jb(const void *p) /* create an interpolation frame */ af.frametype = AST_FRAME_VOICE; af.subclass = pvt->voiceformat; - af.samples = frame.ms * 8; + af.samples = frame.ms * (ast_format_rate(pvt->voiceformat) / 1000); af.src = "IAX2 JB interpolation"; af.delivery = ast_tvadd(pvt->rxcore, ast_samp2tv(next, 1000)); af.offset = AST_FRIENDLY_OFFSET; @@ -3297,7 +3297,7 @@ static int schedule_delivery(struct iax_frame *fr, int updatehistory, int fromtr if(fr->af.frametype == AST_FRAME_VOICE) { type = JB_TYPE_VOICE; - len = ast_codec_get_samples(&fr->af) / 8; + len = ast_codec_get_samples(&fr->af) / (ast_format_rate(fr->af.subclass) / 1000); } else if(fr->af.frametype == AST_FRAME_CNG) { type = JB_TYPE_SILENCE; } @@ -4683,6 +4683,7 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str int voice = 0; int genuine = 0; int adjust; + int rate = ast_format_rate(f->subclass) / 1000; struct timeval *delivery = NULL; @@ -4750,7 +4751,7 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str p->offset = ast_tvadd(p->offset, ast_samp2tv(adjust, 10000)); if (!p->nextpred) { - p->nextpred = ms; /*f->samples / 8;*/ + p->nextpred = ms; /*f->samples / rate;*/ if (p->nextpred <= p->lastsent) p->nextpred = p->lastsent + 3; } @@ -4769,11 +4770,11 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str ast_debug(1, "predicted timestamp skew (%u) > max (%u), using real ts instead.\n", abs(ms - p->nextpred), MAX_TIMESTAMP_SKEW); - if (f->samples >= 8) /* check to make sure we dont core dump */ + if (f->samples >= rate) /* check to make sure we dont core dump */ { - int diff = ms % (f->samples / 8); + int diff = ms % (f->samples / rate); if (diff) - ms += f->samples/8 - diff; + ms += f->samples/rate - diff; } p->nextpred = ms; @@ -4805,7 +4806,7 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str } p->lastsent = ms; if (voice) - p->nextpred = p->nextpred + f->samples / 8; + p->nextpred = p->nextpred + f->samples / rate; return ms; } |