[svn-commits] dvossel: branch 1.6.1 r205596 - in /branches/1.6.1: ./ channels/ include/aste...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jul 9 10:47:28 CDT 2009
Author: dvossel
Date: Thu Jul 9 10:47:25 2009
New Revision: 205596
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=205596
Log:
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/
........
................
Modified:
branches/1.6.1/ (props changed)
branches/1.6.1/channels/chan_iax2.c
branches/1.6.1/include/asterisk/frame.h
branches/1.6.1/main/rtp.c
Propchange: branches/1.6.1/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.1/channels/chan_iax2.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.1/channels/chan_iax2.c?view=diff&rev=205596&r1=205595&r2=205596
==============================================================================
--- branches/1.6.1/channels/chan_iax2.c (original)
+++ branches/1.6.1/channels/chan_iax2.c Thu Jul 9 10:47:25 2009
@@ -3078,7 +3078,7 @@
/* 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;
@@ -3150,7 +3150,7 @@
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;
}
@@ -4494,6 +4494,7 @@
int voice = 0;
int genuine = 0;
int adjust;
+ int rate = ast_format_rate(f->subclass) / 1000;
struct timeval *delivery = NULL;
@@ -4561,7 +4562,7 @@
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;
}
@@ -4580,11 +4581,11 @@
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;
@@ -4616,7 +4617,7 @@
}
p->lastsent = ms;
if (voice)
- p->nextpred = p->nextpred + f->samples / 8;
+ p->nextpred = p->nextpred + f->samples / rate;
return ms;
}
Modified: branches/1.6.1/include/asterisk/frame.h
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.1/include/asterisk/frame.h?view=diff&rev=205596&r1=205595&r2=205596
==============================================================================
--- branches/1.6.1/include/asterisk/frame.h (original)
+++ branches/1.6.1/include/asterisk/frame.h Thu Jul 9 10:47:25 2009
@@ -149,7 +149,7 @@
int subclass;
/*! Length of data */
int datalen;
- /*! Number of 8khz samples in this frame */
+ /*! Number of samples in this frame */
int samples;
/*! Was the data malloc'd? i.e. should we free it when we discard the frame? */
int mallocd;
Modified: branches/1.6.1/main/rtp.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.1/main/rtp.c?view=diff&rev=205596&r1=205595&r2=205596
==============================================================================
--- branches/1.6.1/main/rtp.c (original)
+++ branches/1.6.1/main/rtp.c Thu Jul 9 10:47:25 2009
@@ -732,6 +732,11 @@
return -1;
}
+static int rtp_get_rate(int subclass)
+{
+ return (subclass == AST_FORMAT_G722) ? 8000 : ast_format_rate(subclass);
+}
+
unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
{
unsigned int interval;
@@ -990,10 +995,10 @@
}
} else if ((rtp->resp == resp) && !power) {
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->samples = rtp->dtmfsamples * 8;
+ f->samples = rtp->dtmfsamples * (rtp_get_rate(f->subclass) / 1000);
rtp->resp = 0;
} else if (rtp->resp == resp)
- rtp->dtmfsamples += 20 * 8;
+ rtp->dtmfsamples += 20 * (rtp_get_rate(f->subclass) / 1000);
rtp->dtmf_timeout = dtmftimeout;
return f;
}
@@ -1074,7 +1079,7 @@
if ((rtp->lastevent != seqno) && rtp->resp) {
rtp->dtmf_duration = new_duration;
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
}
@@ -1084,7 +1089,7 @@
if (rtp->resp && rtp->resp != resp) {
/* Another digit already began. End it */
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
}
@@ -1459,15 +1464,16 @@
double d;
double dtv;
double prog;
-
double normdev_rxjitter_current;
+ int rate = rtp_get_rate(rtp->f.subclass);
+
if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
gettimeofday(&rtp->rxcore, NULL);
rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
/* map timestamp to a real time */
rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
- rtp->rxcore.tv_sec -= timestamp / 8000;
- rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
+ rtp->rxcore.tv_sec -= timestamp / rate;
+ rtp->rxcore.tv_usec -= (timestamp % rate) * 125;
/* Round to 0.1ms for nice, pretty timestamps */
rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
if (rtp->rxcore.tv_usec < 0) {
@@ -1479,13 +1485,13 @@
gettimeofday(&now,NULL);
/* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
- when->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
- when->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
+ when->tv_sec = rtp->rxcore.tv_sec + timestamp / rate;
+ when->tv_usec = rtp->rxcore.tv_usec + (timestamp % rate) * 125;
if (when->tv_usec >= 1000000) {
when->tv_usec -= 1000000;
when->tv_sec += 1;
}
- prog = (double)((timestamp-rtp->seedrxts)/8000.);
+ prog = (double)((timestamp-rtp->seedrxts)/(float)(rate));
dtv = (double)rtp->drxcore + (double)(prog);
current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
transit = current_time - dtv;
@@ -1797,7 +1803,7 @@
if (rtp->resp) {
struct ast_frame *f;
f = send_dtmf(rtp, AST_FRAME_DTMF_END);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
return f;
@@ -1874,7 +1880,7 @@
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
/* 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.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000);
rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
} else if (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
/* Video -- samples is # of samples vs. 90000 */
@@ -3564,6 +3570,11 @@
unsigned int ms;
int pred;
int mark = 0;
+ int rate = rtp_get_rate(f->subclass) / 1000;
+
+ if (f->subclass == AST_FORMAT_G722) {
+ f->samples /= 2;
+ }
if (rtp->sending_digit) {
return 0;
@@ -3575,7 +3586,7 @@
pred = rtp->lastts + f->samples;
/* Re-calculate last TS */
- rtp->lastts = rtp->lastts + ms * 8;
+ rtp->lastts = rtp->lastts + ms * rate;
if (ast_tvzero(f->delivery)) {
/* If this isn't an absolute delivery time, Check if it is close to our prediction,
and if so, go with our prediction */
@@ -3630,7 +3641,7 @@
rtp->lastdigitts = rtp->lastts;
if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
- rtp->lastts = f->ts * 8;
+ rtp->lastts = f->ts * rate;
/* Get a pointer to the header */
rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
@@ -3801,11 +3812,6 @@
}
while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
- if (f->subclass == AST_FORMAT_G722) {
- /* G.722 is silllllllllllllly */
- f->samples /= 2;
- }
-
ast_rtp_raw_write(rtp, f, codec);
}
} else {
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