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rtpjitterbuffer: fix divide-by-zero

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The estimated packet-duration can sometimes end up as zero, and dividing
by that is never a good idea...

The test reproduces the scenario, and the fix is easy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/966>
This commit is contained in:
Havard Graff 2021-04-25 02:16:45 +02:00 committed by Havard Graff
parent 1368b4214b
commit d75c678479
2 changed files with 49 additions and 3 deletions
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6
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -2498,7 +2498,7 @@ gst_rtp_jitter_buffer_handle_missing_packets (GstRtpJitterBuffer * jitterbuffer,
guint lost_packets;
GstClockTime lost_duration;
GstClockTimeDiff gap_time;
guint saveable_packets;
guint saveable_packets = 0;
GstClockTime saveable_duration;
/* gap time represents the total duration of all missing packets */
@ -2506,7 +2506,9 @@ 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 */
saveable_packets = offset / est_pkt_duration;
if (est_pkt_duration)
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);

46
tests/check/elements/rtpjitterbuffer.c
View file

@ -1355,6 +1355,50 @@ GST_START_TEST (test_no_fractional_lost_event_durations)
GST_END_TEST;
GST_START_TEST (test_late_lost_with_same_pts)
{
GstHarness *h = gst_harness_new ("rtpjitterbuffer");
GstClockTime dts, now;
guint latency_ms = 40;
guint16 seqnum;
guint rtp_ts;
g_object_set (h->element, "do-lost", TRUE, NULL);
seqnum = construct_deterministic_initial_state (h, latency_ms);
dts = seqnum * TEST_BUF_DURATION;
rtp_ts = seqnum * TEST_RTP_TS_DURATION;
/* set the time on the clock one buffer-length after the
length of the jitterbuffer */
now = dts + latency_ms * GST_MSECOND + TEST_BUF_DURATION;
gst_test_clock_set_time (GST_TEST_CLOCK (GST_ELEMENT_CLOCK (h->element)),
now);
/* now two buffers arrive, same arrival time (in the past, must
have spent a lot of time from udpsrc to jitterbuffer!),
with the same rtptimestamp (typical of videobuffers),
with a gap in between them */
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (h,
generate_test_buffer_full (dts, seqnum, rtp_ts)));
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (h,
generate_test_buffer_full (dts, seqnum + 2, rtp_ts)));
/* the lost event is generated immediately since we are already
too late to wait for anything */
verify_lost_event (h, seqnum + 1, dts, 0);
gst_buffer_unref (gst_harness_pull (h));
gst_buffer_unref (gst_harness_pull (h));
/* verify that we have pulled out all waiting buffers and events */
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
fail_unless_equals_int (0, gst_harness_events_in_queue (h));
gst_harness_teardown (h);
}
GST_END_TEST;
static void
gst_test_clock_set_time_and_process (GstTestClock * testclock,
@ -3316,7 +3360,7 @@ rtpjitterbuffer_suite (void)
test_loss_equidistant_spacing_with_parameter_packets);
tcase_add_loop_test (tc_chain, test_no_fractional_lost_event_durations, 0,
G_N_ELEMENTS (no_fractional_lost_event_durations_input));
tcase_add_test (tc_chain, test_late_lost_with_same_pts);
tcase_add_test (tc_chain, test_rtx_expected_next);
tcase_add_test (tc_chain, test_rtx_not_bursting_requests);