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rtpjitterbuffer: Allow earlier reference-timestamp-meta

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Allow reference-timestamp-meta to be added earlier if an RTCP sender
report is sent before the first RTP packet.

Fixes #2843

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5084>
This commit is contained in:
Charlie Blevins 2023-07-20 17:04:22 -04:00 committed by GStreamer Marge Bot
parent 6e5ca29e8f
commit 05cffc19dd
2 changed files with 137 additions and 0 deletions
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6
subprojects/gst-plugins-good/gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -4835,6 +4835,12 @@ out:
priv->last_known_ext_rtptime = ext_rtptime;
priv->last_known_ntpnstime = ntpnstime;
if (G_UNLIKELY (priv->last_ssrc == -1)) {
GST_DEBUG_OBJECT (jitterbuffer, "SSRC changed from %u to %u",
priv->last_ssrc, ssrc);
priv->last_ssrc = ssrc;
}
if (priv->last_ntpnstime != GST_CLOCK_TIME_NONE
&& ntpnstime - priv->last_ntpnstime < priv->sync_interval * GST_MSECOND) {
gst_buffer_replace (&priv->last_sr, NULL);

131
subprojects/gst-plugins-good/tests/check/elements/rtpjitterbuffer.c
View file

@ -29,6 +29,7 @@
#include <gst/check/gsttestclock.h>
#include <gst/check/gstharness.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
@ -58,6 +59,11 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_STATIC_CAPS ("application/x-rtp, "
"clock-rate = (int) [ 1, 2147483647 ]")
);
static GstStaticPadTemplate rtcpsrctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtcp")
);
static void
buffer_dropped (G_GNUC_UNUSED gpointer data, GstMiniObject * obj)
@ -709,6 +715,71 @@ construct_deterministic_initial_state (GstHarness * h, gint latency_ms)
return next_seqnum;
}
static GstBuffer *
setup_rtcp_sender_report (GstElement * jitterbuffer,
guint64 ntp_time_seconds, guint32 rtp_time)
{
GstRTCPBuffer rtcp_buf = GST_RTCP_BUFFER_INIT;
GstRTCPPacket packet;
GstBuffer *srep_buf;
srep_buf = gst_rtcp_buffer_new (1000);
if (gst_rtcp_buffer_map (srep_buf, GST_MAP_READWRITE, &rtcp_buf)) {
if (gst_rtcp_buffer_add_packet (&rtcp_buf, GST_RTCP_TYPE_SR, &packet)) {
gst_rtcp_packet_sr_set_sender_info (&packet, TEST_BUF_SSRC, /* SSRC */
/* ntp_time_seconds is the test time in seconds since Jan 1 1900.
Here it is converted to NTP format */
(guint64) ntp_time_seconds << 32, /* NTP timestamp */
rtp_time, /* RTP timestamp */
1, /* sender's packet count */
100); /* sender's octet count */
}
gst_rtcp_buffer_unmap (&rtcp_buf);
}
return srep_buf;
}
static GstPad *
setup_rtcp_pads (GstElement * jitterbuffer)
{
GstPad *rtcp_fxsrc_pad;
GstPad *rtcp_sink_pad;
GstPadTemplate *pad_tmp;
GstCaps *rtcp_caps;
pad_tmp = gst_static_pad_template_get (&rtcpsrctemplate);
rtcp_fxsrc_pad = gst_pad_new_from_template (pad_tmp, "src");
fail_if (rtcp_fxsrc_pad == NULL, "Could not create a srcpad");
rtcp_sink_pad = gst_element_request_pad_simple (jitterbuffer, "sink_rtcp");
fail_if (rtcp_sink_pad == NULL, "Could not get sink pad from %s",
GST_ELEMENT_NAME (jitterbuffer));
fail_unless (gst_pad_link (rtcp_fxsrc_pad, rtcp_sink_pad) == GST_PAD_LINK_OK,
"Could not link source and %s sink pads",
GST_ELEMENT_NAME (jitterbuffer));
gst_pad_set_active (rtcp_sink_pad, TRUE);
gst_pad_set_active (rtcp_fxsrc_pad, TRUE);
rtcp_caps = gst_caps_new_simple ("application/x-rtcp",
"clock-rate", G_TYPE_INT, TEST_BUF_CLOCK_RATE, NULL);
gst_check_setup_events_with_stream_id (rtcp_fxsrc_pad, jitterbuffer,
rtcp_caps, GST_FORMAT_TIME, "/test/jitbuf/rtcp");
gst_object_unref (pad_tmp);
gst_caps_unref (rtcp_caps);
gst_object_unref (rtcp_sink_pad);
return rtcp_fxsrc_pad;
}
GST_START_TEST (test_lost_event)
{
GstHarness *h = gst_harness_new ("rtpjitterbuffer");
@ -3498,6 +3569,65 @@ GST_START_TEST (test_gap_using_rtx_does_not_stall)
GST_END_TEST;
GST_START_TEST (test_early_rtcp_sr_allows_meta)
{
GstElement *jitterbuffer;
GstPad *rtcp_fxsrc_pad;
GstBuffer *srep_buf;
GstBuffer *rtp_buffer;
GstReferenceTimestampMeta *meta;
GstCaps *ntp_caps;
// No buffers since we want to control them later
jitterbuffer = setup_jitterbuffer (0);
g_object_set (G_OBJECT (jitterbuffer),
"add-reference-timestamp-meta", TRUE, NULL);
fail_unless (start_jitterbuffer (jitterbuffer)
== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
srep_buf = setup_rtcp_sender_report (jitterbuffer, 3899471400, 1000);
rtcp_fxsrc_pad = setup_rtcp_pads (jitterbuffer);
/* rtcp sr is first */
gst_pad_push (rtcp_fxsrc_pad, srep_buf);
/* create rtp buf, with matching rtp timestamp */
rtp_buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
if (rtp_buffer) {
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
if (gst_rtp_buffer_map (rtp_buffer, GST_MAP_WRITE, &rtp)) {
gst_rtp_buffer_set_ssrc (&rtp, TEST_BUF_SSRC);
/* first rtp buffer, but second buffer overall, arrives 1 clock unit
after rtcp sr */
gst_rtp_buffer_set_timestamp (&rtp, 1001);
gst_rtp_buffer_unmap (&rtp);
}
}
/* RTP buf is second */
gst_pad_push (mysrcpad, rtp_buffer);
ntp_caps = gst_caps_new_empty_simple ("timestamp/x-ntp");
meta = gst_buffer_get_reference_timestamp_meta (rtp_buffer, ntp_caps);
/* result should match the test time plus one clock unit. One
clock unit is 125000 nanoseconds */
fail_unless (meta->timestamp == (3899471400 * GST_SECOND + 125000));
/* cleanup */
cleanup_jitterbuffer (jitterbuffer);
gst_object_unref (rtcp_fxsrc_pad);
gst_caps_unref (ntp_caps);
}
GST_END_TEST;
static Suite *
rtpjitterbuffer_suite (void)
{
@ -3577,6 +3707,7 @@ rtpjitterbuffer_suite (void)
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);
tcase_add_test (tc_chain, test_early_rtcp_sr_allows_meta);
return s;