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/branches/1.4@205471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
David Vossel 17 years ago
parent 2e330f772c
commit beaf6217b3

@ -2768,7 +2768,7 @@ 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;
@ -2841,7 +2841,7 @@ 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;
}
@ -4072,6 +4072,7 @@ 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;
@ -4139,7 +4140,7 @@ 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;
}
@ -4158,11 +4159,11 @@ static unsigned int calc_timestamp(struct chan_iax2_pvt *p, unsigned int ts, str
ast_log(LOG_DEBUG, "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;
@ -4194,7 +4195,7 @@ 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;
}

@ -150,7 +150,7 @@ struct ast_frame {
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;

@ -524,6 +524,11 @@ int ast_rtcp_fd(struct ast_rtp *rtp)
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;
@ -771,7 +776,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
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;
}
@ -781,7 +786,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
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;
}
@ -1047,14 +1052,15 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
double d;
double dtv;
double prog;
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) {
@ -1066,13 +1072,13 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
gettimeofday(&now,NULL);
/* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
tv->tv_sec = rtp->rxcore.tv_sec + timestamp / rate;
tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % rate) * 125;
if (tv->tv_usec >= 1000000) {
tv->tv_usec -= 1000000;
tv->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;
@ -1340,7 +1346,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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;
@ -1362,7 +1368,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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) / 1000);
} else {
/* Video -- samples is # of samples vs. 90000 */
@ -2678,6 +2684,12 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
unsigned int ms;
int pred;
int mark = 0;
int rate = rtp_get_rate(f->subclass) / 1000;
if (f->subclass == AST_FORMAT_G722) {
/* G722 is silllllllllllllllllly */
f->samples /= 2;
}
if (rtp->sending_digit) {
return 0;
@ -2689,7 +2701,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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 */
@ -2732,7 +2744,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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 - hdrlen);
@ -2898,11 +2910,6 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
}
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 {
@ -2913,10 +2920,6 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
f = _f;
}
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)

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