From 97e083be4bc51f35a7f6bb34ce9e0df08011d515 Mon Sep 17 00:00:00 2001
From: dvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>
Date: Thu, 9 Jul 2009 15:37:05 +0000
Subject: Merged revisions 205479 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/trunk

................
  r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines

  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: http://svn.digium.com/svn/asterisk/branches/1.6.2@205595 f38db490-d61c-443f-a65b-d21fe96a405b
---
 channels/chan_iax2.c | 15 ++++++++-------
 1 file changed, 8 insertions(+), 7 deletions(-)

(limited to 'channels/chan_iax2.c')

diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 82b2e6122..81dddcd5a 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -3181,7 +3181,7 @@ 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;
@@ -3253,7 +3253,7 @@ 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;
 	}
@@ -4601,6 +4601,7 @@ 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;
 
 
@@ -4668,7 +4669,7 @@ 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;
 				}
@@ -4687,11 +4688,11 @@ 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;
@@ -4723,7 +4724,7 @@ 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;
 }
 
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