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rtpjitterbuffer: fixed stall on gap when using rtx

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Co-authored-by: Håvard Graff <havard@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1055>
This commit is contained in:
Tulio Beloqui 2021-08-06 16:25:02 +02:00 committed by GStreamer Marge Bot
parent 620e9323c5
commit 9af6ce974a
2 changed files with 47 additions and 6 deletions
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16
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -2498,7 +2498,8 @@ gst_rtp_jitter_buffer_handle_missing_packets (GstRtpJitterBuffer * jitterbuffer,
guint lost_packets;
GstClockTime lost_duration;
GstClockTimeDiff gap_time;
guint saveable_packets = 0;
guint max_saveable_packets = 0;
GstClockTime max_saveable_duration;
GstClockTime saveable_duration;
/* gap time represents the total duration of all missing packets */
@ -2507,17 +2508,20 @@ gst_rtp_jitter_buffer_handle_missing_packets (GstRtpJitterBuffer * jitterbuffer,
/* based on the estimated packet duration, we
can figure out how many packets we could possibly save */
if (est_pkt_duration)
saveable_packets = offset / est_pkt_duration;
max_saveable_packets = offset / est_pkt_duration;
/* and say that the amount of lost packet is the sequence-number
gap minus these saveable packets, but at least 1 */
lost_packets = MAX (1, (gint) gap - (gint) saveable_packets);
lost_packets = MAX (1, (gint) gap - (gint) max_saveable_packets);
/* now we know how many packets we can actually save */
saveable_packets = gap - lost_packets;
/* now we know how many packets we can possibly save */
max_saveable_packets = gap - lost_packets;
/* we convert that to time */
saveable_duration = saveable_packets * est_pkt_duration;
max_saveable_duration = max_saveable_packets * est_pkt_duration;
/* determine the actual amount of time we can save */
saveable_duration = MIN (max_saveable_duration, gap_time);
/* and we now have the duration we need to fill */
lost_duration = GST_CLOCK_DIFF (saveable_duration, gap_time);

37
tests/check/elements/rtpjitterbuffer.c
View file

@ -3257,6 +3257,21 @@ check_for_stall (GstHarness * h, BufferArrayCtx * bufs, guint num_bufs)
buffer_array_push (h, array, base_seqnum, base_rtptime);
g_array_unref (array);
{
gint64 start_time = g_get_monotonic_time ();
gint64 timeout_s = 30;
while (gst_harness_buffers_in_queue (h) <= in_queue) {
gint64 duration_s =
(g_get_monotonic_time () - start_time) / G_USEC_PER_SEC;
if (duration_s > timeout_s)
break;
g_usleep (G_USEC_PER_SEC / 100);
}
}
/* we expect at least some of those buffers to come through */
return gst_harness_buffers_in_queue (h) > in_queue;
}
@ -3330,6 +3345,27 @@ GST_START_TEST (test_reset_using_rtx_packets_does_not_stall)
GST_END_TEST;
GST_START_TEST (test_gap_using_rtx_does_not_stall)
{
GstHarness *h = gst_harness_new ("rtpjitterbuffer");
BufferArrayCtx bufs[] = {
/* *INDENT-OFF* */
{ 201, -1440, FALSE, 185591 },
{ 265, 1, FALSE, 0 },
/* *INDENT-ON* */
};
g_object_set (h->element, "do-lost", TRUE,
"do-retransmission", TRUE,
"rtx-next-seqnum", FALSE, "rtx-delay-reorder", 0, NULL);
fail_unless (check_for_stall (h, bufs, G_N_ELEMENTS (bufs)));
gst_harness_teardown (h);
}
GST_END_TEST;
static Suite *
rtpjitterbuffer_suite (void)
{
@ -3405,6 +3441,7 @@ rtpjitterbuffer_suite (void)
tcase_add_test (tc_chain, test_reset_timers_does_not_stall);
tcase_add_test (tc_chain, test_multiple_lost_do_not_stall);
tcase_add_test (tc_chain, test_reset_using_rtx_packets_does_not_stall);
tcase_add_test (tc_chain, test_gap_using_rtx_does_not_stall);
return s;