From 05cffc19ddd50ece70b56e8248468378efd4cd57 Mon Sep 17 00:00:00 2001 From: Charlie Blevins Date: Thu, 20 Jul 2023 17:04:22 -0400 Subject: [PATCH] rtpjitterbuffer: Allow earlier reference-timestamp-meta Allow reference-timestamp-meta to be added earlier if an RTCP sender report is sent before the first RTP packet. Fixes #2843 Part-of: --- .../gst/rtpmanager/gstrtpjitterbuffer.c | 6 + .../tests/check/elements/rtpjitterbuffer.c | 131 ++++++++++++++++++ 2 files changed, 137 insertions(+) diff --git a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpjitterbuffer.c b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpjitterbuffer.c index 063e39b9f7..eb2360d893 100644 --- a/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpjitterbuffer.c @@ -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); diff --git a/subprojects/gst-plugins-good/tests/check/elements/rtpjitterbuffer.c b/subprojects/gst-plugins-good/tests/check/elements/rtpjitterbuffer.c index de8f06af14..8984c35081 100644 --- a/subprojects/gst-plugins-good/tests/check/elements/rtpjitterbuffer.c +++ b/subprojects/gst-plugins-good/tests/check/elements/rtpjitterbuffer.c @@ -29,6 +29,7 @@ #include #include #include +#include /* 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;