mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
If, say, a rtx-packet arrives really late, this can have a dramatic effect on the jitterbuffer clock-skew logic, having it being reset and losing track of the current dts-to-pts calculations, directly affecting the packets that arrive later. This is demonstrated in the test, where a RTX packet is pushed in really late, and without this patch the last packet will have its PTS affected by this, where as a late RTX packet should be redundant information, and not affect anything.
This commit is contained in:
parent
b9c3e354ee
commit
ec5fa49631
2 changed files with 83 additions and 4 deletions
|
@ -2302,6 +2302,12 @@ get_rtx_delay (GstRtpJitterBufferPrivate * priv)
|
|||
GstClockTime delay;
|
||||
|
||||
if (priv->rtx_delay == -1) {
|
||||
/* the maximum delay for any RTX-packet is given by the latency, since
|
||||
anything after that is considered lost. For various calulcations,
|
||||
(given large avg_jitter and/or packet_spacing), the resuling delay
|
||||
could exceed the configured latency, ending up issuing an RTX-request
|
||||
that would never arrive in time. To help this we cap the delay
|
||||
for any RTX with the last possible time it could still arrive in time. */
|
||||
GstClockTime delay_max = (priv->latency_ns > priv->avg_rtx_rtt) ?
|
||||
priv->latency_ns - priv->avg_rtx_rtt : priv->latency_ns;
|
||||
|
||||
|
@ -3074,10 +3080,13 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
|
|||
|
||||
/* calculate a pts based on rtptime and arrival time (dts) */
|
||||
/* If we estimated the DTS, don't consider it in the clock skew calculations */
|
||||
pts =
|
||||
rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
|
||||
rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
|
||||
|
||||
if (gap >= 0) {
|
||||
pts =
|
||||
rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
|
||||
rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
|
||||
}
|
||||
/* else gap < 0 then we will drop the buffer anyway, so we don't need to
|
||||
calculate it's pts */
|
||||
if (G_LIKELY (gap == 0)) {
|
||||
/* packet is expected */
|
||||
calculate_packet_spacing (jitterbuffer, rtptime, pts);
|
||||
|
|
|
@ -2224,6 +2224,74 @@ GST_START_TEST (test_rtx_large_packet_spacing_and_large_rtt)
|
|||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_rtx_large_packet_spacing_does_not_reset_jitterbuffer)
|
||||
{
|
||||
gint latency_ms = 20;
|
||||
gint frame_dur_ms = 50;
|
||||
gint rtx_rtt_ms = 5;
|
||||
gint i, seq;
|
||||
GstBuffer *buffer;
|
||||
GstClockTime now, lost_packet_time;
|
||||
GstClockTime frame_dur = frame_dur_ms * GST_MSECOND;
|
||||
GstHarness *h = gst_harness_new ("rtpjitterbuffer");
|
||||
|
||||
gst_harness_set_src_caps (h, generate_caps ());
|
||||
g_object_set (h->element,
|
||||
"do-lost", TRUE, "latency", latency_ms, "do-retransmission", TRUE, NULL);
|
||||
|
||||
/* Pushing 2 frames @frame_dur_ms ms apart from each other to initialize
|
||||
* packet_spacing and avg jitter */
|
||||
for (seq = 0, now = 0; seq < 2; ++seq, now += frame_dur) {
|
||||
gst_harness_set_time (h, now);
|
||||
gst_harness_push (h, generate_test_buffer_full (now, seq,
|
||||
AS_TEST_BUF_RTP_TIME (now)));
|
||||
if (seq == 0)
|
||||
gst_harness_crank_single_clock_wait (h);
|
||||
buffer = gst_harness_pull (h);
|
||||
fail_unless_equals_int64 (now, GST_BUFFER_PTS (buffer));
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
/* drop GstEventStreamStart & GstEventCaps & GstEventSegment */
|
||||
for (i = 0; i < 3; i++)
|
||||
gst_event_unref (gst_harness_pull_event (h));
|
||||
/* drop reconfigure event */
|
||||
gst_event_unref (gst_harness_pull_upstream_event (h));
|
||||
|
||||
/* Waiting for the RTX timer of packet #2 to timeout */
|
||||
lost_packet_time = now;
|
||||
gst_harness_crank_single_clock_wait (h);
|
||||
fail_unless_equals_int64 (now + latency_ms * GST_MSECOND,
|
||||
gst_clock_get_time (GST_ELEMENT_CLOCK (h->element)));
|
||||
verify_rtx_event (h, seq, now, latency_ms, frame_dur);
|
||||
verify_lost_event (h, seq, now, frame_dur);
|
||||
now += latency_ms * GST_MSECOND;
|
||||
|
||||
/* Pushing packet #2 as RTX */
|
||||
now += rtx_rtt_ms * GST_MSECOND;
|
||||
gst_harness_set_time (h, now);
|
||||
buffer =
|
||||
generate_test_buffer_full (now, seq,
|
||||
AS_TEST_BUF_RTP_TIME (lost_packet_time));
|
||||
GST_BUFFER_FLAG_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION);
|
||||
fail_unless_equals_int (GST_FLOW_OK, gst_harness_push (h, buffer));
|
||||
fail_unless_equals_int (0, gst_harness_buffers_in_queue (h));
|
||||
|
||||
/* Packet #3 should have PTS not affected by clock skew logic */
|
||||
seq += 1;
|
||||
now = seq * frame_dur;
|
||||
gst_harness_set_time (h, now);
|
||||
gst_harness_push (h, generate_test_buffer_full (now, seq,
|
||||
AS_TEST_BUF_RTP_TIME (now)));
|
||||
buffer = gst_harness_pull (h);
|
||||
fail_unless_equals_int64 (now, GST_BUFFER_PTS (buffer));
|
||||
gst_buffer_unref (buffer);
|
||||
|
||||
gst_harness_teardown (h);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_deadline_ts_offset)
|
||||
{
|
||||
GstHarness *h = gst_harness_new ("rtpjitterbuffer");
|
||||
|
@ -2584,6 +2652,8 @@ rtpjitterbuffer_suite (void)
|
|||
tcase_add_test (tc_chain, test_rtx_timer_reuse);
|
||||
tcase_add_test (tc_chain, test_rtx_large_packet_spacing_and_small_rtt);
|
||||
tcase_add_test (tc_chain, test_rtx_large_packet_spacing_and_large_rtt);
|
||||
tcase_add_test (tc_chain,
|
||||
test_rtx_large_packet_spacing_does_not_reset_jitterbuffer);
|
||||
|
||||
tcase_add_test (tc_chain, test_deadline_ts_offset);
|
||||
tcase_add_test (tc_chain, test_big_gap_seqnum);
|
||||
|
|
Loading…
Reference in a new issue