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func_jitterbuffer: Add audio/video sync support.

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This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.

ASTERISK-28533

Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492
This commit is contained in:
Joshua Colp
2019-09-06 13:18:55 +00:00
parent f9f17f3bfe
commit 6647be69ac
7 changed files with 237 additions and 24 deletions
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34
res/res_rtp_asterisk.c
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@@ -3204,15 +3204,6 @@ static int rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size,
return res;
}
static int rtp_get_rate(struct ast_format *format)
{
/* For those wondering: due to a fluke in RFC publication, G.722 is advertised
* as having a sample rate of 8kHz, while implementations must know that its
* real rate is 16kHz. Seriously.
*/
return (ast_format_cmp(format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) ? 8000 : (int)ast_format_get_sample_rate(format);
}
static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
{
unsigned int interval;
@@ -4096,7 +4087,7 @@ static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, cha
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass.format) / 1000) > rtp->send_duration) {
if (duration > 0 && (measured_samples = duration * ast_rtp_get_rate(rtp->f.subclass.format) / 1000) > rtp->send_duration) {
ast_debug(2, "Adjusting final end duration from %d to %u\n", rtp->send_duration, measured_samples);
rtp->send_duration = measured_samples;
}
@@ -4349,7 +4340,7 @@ static int ast_rtcp_generate_report(struct ast_rtp_instance *instance, unsigned
report_block->lost_count.fraction = (fraction_lost & 0xff);
report_block->lost_count.packets = (lost_packets & 0xffffff);
report_block->highest_seq_no = (rtp->cycles | (rtp->lastrxseqno & 0xffff));
report_block->ia_jitter = (unsigned int)(rtp->rxjitter * rtp_get_rate(rtp->f.subclass.format));
report_block->ia_jitter = (unsigned int)(rtp->rxjitter * ast_rtp_get_rate(rtp->f.subclass.format));
report_block->lsr = rtp->rtcp->themrxlsr;
/* If we haven't received an SR report, DLSR should be 0 */
if (!ast_tvzero(rtp->rtcp->rxlsr)) {
@@ -4431,7 +4422,7 @@ static int ast_rtcp_calculate_sr_rr_statistics(struct ast_rtp_instance *instance
ast_verbose(" Fraction lost: %d\n", report_block->lost_count.fraction);
ast_verbose(" Cumulative loss: %u\n", report_block->lost_count.packets);
ast_verbose(" Highest seq no: %u\n", report_block->highest_seq_no);
ast_verbose(" IA jitter: %.4f\n", (double)report_block->ia_jitter / rtp_get_rate(rtp->f.subclass.format));
ast_verbose(" IA jitter: %.4f\n", (double)report_block->ia_jitter / ast_rtp_get_rate(rtp->f.subclass.format));
ast_verbose(" Their last SR: %u\n", report_block->lsr);
ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(report_block->dlsr / 65536.0));
}
@@ -4684,7 +4675,7 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
int pred, mark = 0;
unsigned int ms = calc_txstamp(rtp, &frame->delivery);
struct ast_sockaddr remote_address = { {0,} };
int rate = rtp_get_rate(frame->subclass.format) / 1000;
int rate = ast_rtp_get_rate(frame->subclass.format) / 1000;
unsigned int seqno;
#ifdef TEST_FRAMEWORK
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
@@ -5204,7 +5195,7 @@ 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.format);
int rate = ast_rtp_get_rate(rtp->f.subclass.format);
double normdev_rxjitter_current;
if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
@@ -5359,7 +5350,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
rtp->dtmf_duration = new_duration;
rtp->resp = resp;
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, ast_rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
AST_LIST_INSERT_TAIL(frames, f, frame_list);
@@ -5390,7 +5381,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
if (rtp->resp && rtp->resp != resp) {
/* Another digit already began. End it */
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, ast_rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
AST_LIST_INSERT_TAIL(frames, f, frame_list);
@@ -5487,10 +5478,10 @@ static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, u
}
} else if ((rtp->resp == resp) && !power) {
f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
f->samples = rtp->dtmfsamples * (rtp_get_rate(rtp->lastrxformat) / 1000);
f->samples = rtp->dtmfsamples * (ast_rtp_get_rate(rtp->lastrxformat) / 1000);
rtp->resp = 0;
} else if (rtp->resp == resp) {
rtp->dtmfsamples += 20 * (rtp_get_rate(rtp->lastrxformat) / 1000);
rtp->dtmfsamples += 20 * (ast_rtp_get_rate(rtp->lastrxformat) / 1000);
}
rtp->dtmf_timeout = 0;
@@ -6229,6 +6220,7 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, s
transport_rtp->f.delivery.tv_sec = 0;
transport_rtp->f.delivery.tv_usec = 0;
transport_rtp->f.src = "RTP";
transport_rtp->f.stream_num = rtp->stream_num;
f = &transport_rtp->f;
break;
case AST_RTP_RTCP_RTPFB:
@@ -7104,7 +7096,7 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
if (rtp->resp) {
struct ast_frame *f;
f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0);
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, ast_rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
@@ -7188,7 +7180,7 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
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 / (rtp_get_rate(rtp->f.subclass.format) / 1000);
rtp->f.ts = timestamp / (ast_rtp_get_rate(rtp->f.subclass.format) / 1000);
rtp->f.len = rtp->f.samples / ((ast_format_get_sample_rate(rtp->f.subclass.format) / 1000));
} else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_VIDEO) {
/* Video -- samples is # of samples vs. 90000 */
@@ -7196,7 +7188,7 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
rtp->lastividtimestamp = timestamp;
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass.format) / 1000);
rtp->f.ts = timestamp / (ast_rtp_get_rate(rtp->f.subclass.format) / 1000);
rtp->f.samples = timestamp - rtp->lastividtimestamp;
rtp->lastividtimestamp = timestamp;
rtp->f.delivery.tv_sec = 0;