From 97e083be4bc51f35a7f6bb34ce9e0df08011d515 Mon Sep 17 00:00:00 2001 From: dvossel 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 ++++++++------- include/asterisk/frame.h | 2 +- main/rtp.c | 44 +++++++++++++++++++++++++------------------- 3 files changed, 34 insertions(+), 27 deletions(-) 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; } diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h index b69c5a1d0..0593f2281 100644 --- a/include/asterisk/frame.h +++ b/include/asterisk/frame.h @@ -149,7 +149,7 @@ 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; diff --git a/main/rtp.c b/main/rtp.c index ac99c96a7..a19f2b952 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -726,6 +726,11 @@ 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; @@ -984,10 +989,10 @@ static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char * } } 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; } @@ -1068,7 +1073,7 @@ 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; } @@ -1078,7 +1083,7 @@ 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; } @@ -1453,15 +1458,16 @@ static void calc_rxstamp(struct timeval *when, struct ast_rtp *rtp, unsigned int 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) { @@ -1473,13 +1479,13 @@ static void calc_rxstamp(struct timeval *when, struct ast_rtp *rtp, unsigned int 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; @@ -1791,7 +1797,7 @@ 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; @@ -1868,7 +1874,7 @@ 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 if (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) { /* Video -- samples is # of samples vs. 90000 */ @@ -3605,6 +3611,11 @@ 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) { + f->samples /= 2; + } if (rtp->sending_digit) { return 0; @@ -3616,7 +3627,7 @@ 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 */ @@ -3671,7 +3682,7 @@ 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.ptr - hdrlen); @@ -3853,11 +3864,6 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f) } 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 { -- cgit v1.2.3