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res_rtp_asterisk: Address jittery DTMF events in RTP streams

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(closes issue ASTERISK-21170)
Reported by: NITESH BANSAL
Patches:
    dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
........

Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jonathan Rose
2013-10-23 17:37:15 +00:00
parent 4e5291c1db
commit 4241f7b6ea
2 changed files with 37 additions and 20 deletions
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4
main/translate.c
View File

@@ -540,6 +540,10 @@ struct ast_frame *ast_translate(struct ast_trans_pvt *path, struct ast_frame *f,
/* Predict next outgoing timestamp from samples in this
frame. */
path->nextout = ast_tvadd(path->nextout, ast_samp2tv(out->samples, ast_format_rate(&out->subclass.format)));
if (f->samples != out->samples && ast_test_flag(out, AST_FRFLAG_HAS_TIMING_INFO)) {
ast_debug(4, "Sample size different %u vs %u\n", f->samples, out->samples);
ast_clear_flag(out, AST_FRFLAG_HAS_TIMING_INFO);
}
} else {
out->delivery = ast_tv(0, 0);
ast_set2_flag(out, has_timing_info, AST_FRFLAG_HAS_TIMING_INFO);

53
res/res_rtp_asterisk.c
View File

@@ -96,6 +96,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#define RTCP_PT_APP 204
#define RTP_MTU 1200
#define DTMF_SAMPLE_RATE_MS 8 /*!< DTMF samples per millisecond */
#define DEFAULT_DTMF_TIMEOUT (150 * (8000 / 1000)) /*!< samples */
@@ -1725,6 +1726,35 @@ static void rtp_add_candidates_to_ice(struct ast_rtp_instance *instance, struct
}
#endif
/*!
* \internal
* \brief Calculates the elapsed time from issue of the first tx packet in an
* rtp session and a specified time
*
* \param rtp pointer to the rtp struct with the transmitted rtp packet
* \param delivery time of delivery - if NULL or zero value, will be ast_tvnow()
*
* \return time elapsed in milliseconds
*/
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
{
struct timeval t;
long ms;
if (ast_tvzero(rtp->txcore)) {
rtp->txcore = ast_tvnow();
rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
}
t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
ms = 0;
}
rtp->txcore = t;
return (unsigned int) ms;
}
static int ast_rtp_new(struct ast_rtp_instance *instance,
struct ast_sched_context *sched, struct ast_sockaddr *addr,
void *data)
@@ -1958,6 +1988,7 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
rtp->send_duration = 160;
rtp->lastts += calc_txstamp(rtp, NULL) * DTMF_SAMPLE_RATE_MS;
rtp->lastdigitts = rtp->lastts + rtp->send_duration;
/* Create the actual packet that we will be sending */
@@ -2042,6 +2073,7 @@ static int ast_rtp_dtmf_continuation(struct ast_rtp_instance *instance)
/* And now we increment some values for the next time we swing by */
rtp->seqno++;
rtp->send_duration += 160;
rtp->lastts += calc_txstamp(rtp, NULL) * DTMF_SAMPLE_RATE_MS;
return 0;
}
@@ -2121,7 +2153,7 @@ static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, cha
res = 0;
/* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
rtp->lastts += rtp->send_duration;
rtp->lastts += calc_txstamp(rtp, NULL) * DTMF_SAMPLE_RATE_MS;
cleanup:
rtp->sending_digit = 0;
rtp->send_digit = 0;
@@ -2171,25 +2203,6 @@ static void ast_rtp_change_source(struct ast_rtp_instance *instance)
return;
}
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
{
struct timeval t;
long ms;
if (ast_tvzero(rtp->txcore)) {
rtp->txcore = ast_tvnow();
rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
}
t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
ms = 0;
}
rtp->txcore = t;
return (unsigned int) ms;
}
static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
{
unsigned int sec, usec, frac;