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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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
44
main/rtp.c
44
main/rtp.c
@@ -682,6 +682,11 @@ int ast_rtcp_fd(struct ast_rtp *rtp)
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return -1;
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}
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static int rtp_get_rate(int subclass)
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{
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return (subclass == AST_FORMAT_G722) ? 8000 : ast_format_rate(subclass);
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}
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unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
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{
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unsigned int interval;
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@@ -911,10 +916,10 @@ static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *
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}
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} else if ((rtp->resp == resp) && !power) {
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f = send_dtmf(rtp, AST_FRAME_DTMF_END);
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f->samples = rtp->dtmfsamples * 8;
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f->samples = rtp->dtmfsamples * (rtp_get_rate(f->subclass) / 1000);
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rtp->resp = 0;
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} else if (rtp->resp == resp)
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rtp->dtmfsamples += 20 * 8;
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rtp->dtmfsamples += 20 * (rtp_get_rate(f->subclass) / 1000);
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rtp->dtmf_timeout = dtmftimeout;
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return f;
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}
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@@ -995,7 +1000,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
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if ((rtp->lastevent != seqno) && rtp->resp) {
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rtp->dtmf_duration = new_duration;
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f = send_dtmf(rtp, AST_FRAME_DTMF_END);
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f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
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f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
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rtp->resp = 0;
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rtp->dtmf_duration = rtp->dtmf_timeout = 0;
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}
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@@ -1005,7 +1010,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
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if (rtp->resp && rtp->resp != resp) {
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/* Another digit already began. End it */
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f = send_dtmf(rtp, AST_FRAME_DTMF_END);
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f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
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f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
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rtp->resp = 0;
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rtp->dtmf_duration = rtp->dtmf_timeout = 0;
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}
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@@ -1322,14 +1327,15 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
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double d;
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double dtv;
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double prog;
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int rate = rtp_get_rate(rtp->f.subclass);
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if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
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gettimeofday(&rtp->rxcore, NULL);
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rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
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/* map timestamp to a real time */
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rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
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rtp->rxcore.tv_sec -= timestamp / 8000;
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rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
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rtp->rxcore.tv_sec -= timestamp / rate;
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rtp->rxcore.tv_usec -= (timestamp % rate) * 125;
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/* Round to 0.1ms for nice, pretty timestamps */
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rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
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if (rtp->rxcore.tv_usec < 0) {
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@@ -1341,13 +1347,13 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
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gettimeofday(&now,NULL);
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/* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
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tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
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tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
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tv->tv_sec = rtp->rxcore.tv_sec + timestamp / rate;
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tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % rate) * 125;
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if (tv->tv_usec >= 1000000) {
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tv->tv_usec -= 1000000;
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tv->tv_sec += 1;
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}
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prog = (double)((timestamp-rtp->seedrxts)/8000.);
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prog = (double)((timestamp-rtp->seedrxts)/(float)(rate));
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dtv = (double)rtp->drxcore + (double)(prog);
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current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
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transit = current_time - dtv;
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@@ -1643,7 +1649,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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if (rtp->resp) {
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struct ast_frame *f;
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f = send_dtmf(rtp, AST_FRAME_DTMF_END);
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f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0));
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f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass)), ast_tv(0, 0));
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rtp->resp = 0;
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rtp->dtmf_timeout = rtp->dtmf_duration = 0;
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return f;
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@@ -1665,7 +1671,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
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/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
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ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
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rtp->f.ts = timestamp / 8;
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rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass) / 1000);
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rtp->f.len = rtp->f.samples / ( (ast_format_rate(rtp->f.subclass) == 16000) ? 16 : 8 );
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} else if(rtp->f.subclass & AST_FORMAT_VIDEO_MASK) {
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/* Video -- samples is # of samples vs. 90000 */
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@@ -3067,6 +3073,11 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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unsigned int ms;
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int pred;
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int mark = 0;
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int rate = rtp_get_rate(f->subclass) / 1000;
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if (f->subclass == AST_FORMAT_G722) {
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f->samples /= 2;
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}
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if (rtp->sending_digit) {
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return 0;
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@@ -3078,7 +3089,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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pred = rtp->lastts + f->samples;
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/* Re-calculate last TS */
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rtp->lastts = rtp->lastts + ms * 8;
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rtp->lastts = rtp->lastts + ms * rate;
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if (ast_tvzero(f->delivery)) {
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/* If this isn't an absolute delivery time, Check if it is close to our prediction,
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and if so, go with our prediction */
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@@ -3133,7 +3144,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
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rtp->lastdigitts = rtp->lastts;
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if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
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rtp->lastts = f->ts * 8;
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rtp->lastts = f->ts * rate;
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/* Get a pointer to the header */
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rtpheader = (unsigned char *)(f->data - hdrlen);
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@@ -3297,11 +3308,6 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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}
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while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) {
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if (f->subclass == AST_FORMAT_G722) {
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/* G.722 is silllllllllllllly */
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f->samples /= 2;
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}
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ast_rtp_raw_write(rtp, f, codec);
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}
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} else {
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