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webrtcbin: Always populate rtp-inbound stats fields

Browse source 
Even if there's no jitterbuffer yet for an incoming stream,
make sure to populate the mandatory statistics with 0 entries.

Fixes problems with the unit test failing sometimes for the
unit test introduced in MR !7338

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7387>
This commit is contained in:
Jan Schmidt 2024-08-20 18:57:50 +10:00 committed by GStreamer Marge Bot
parent 671281d860
commit 055b5af99e
1 changed files with 33 additions and 19 deletions
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52
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
View file

@ -338,12 +338,6 @@ _get_stats_from_rtp_source_stats (TransportStream * stream,
}
}
if (jb_stats)
gst_structure_get (jb_stats, "num-lost", G_TYPE_UINT64, &jb_lost,
"num-duplicates", G_TYPE_UINT64, &duplicates, "num-late",
G_TYPE_UINT64, &late, "rtx-success-count", G_TYPE_UINT64,
&rtx_success, NULL);
in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
@ -359,20 +353,39 @@ _get_stats_from_rtp_source_stats (TransportStream * stream,
/* RTCReceivedRtpStreamStats */
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets)) {
gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
} else {
gst_structure_set (in, "packets-received", G_TYPE_UINT64,
G_GUINT64_CONSTANT (0), NULL);
}
if (jb_stats) {
gst_structure_get (jb_stats, "num-lost", G_TYPE_UINT64, &jb_lost,
"num-duplicates", G_TYPE_UINT64, &duplicates, "num-late",
G_TYPE_UINT64, &late, "rtx-success-count", G_TYPE_UINT64,
&rtx_success, NULL);
gint64 packets_lost = jb_lost > G_MAXINT64 ?
G_MAXINT64 : (gint64) jb_lost;
gst_structure_set (in, "packets-lost", G_TYPE_INT64, packets_lost, NULL);
}
if (gst_structure_get_uint (source_stats, "jitter", &jitter))
gst_structure_set (in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
if (jb_stats)
gst_structure_set (in, "packets-discarded", G_TYPE_UINT64, late,
"packets-repaired", G_TYPE_UINT64, rtx_success, NULL);
} else {
gst_structure_set (in, "packets-lost", G_TYPE_INT64,
G_GINT64_CONSTANT (0), NULL);
gst_structure_set (in, "packets-discarded", G_TYPE_UINT64,
G_GUINT64_CONSTANT (0), "packets-repaired", G_TYPE_UINT64,
G_GUINT64_CONSTANT (0), NULL);
}
if (gst_structure_get_uint (source_stats, "jitter", &jitter)) {
gst_structure_set (in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
} else {
gst_structure_set (in, "jitter", G_TYPE_DOUBLE, 0.0, NULL);
}
/*
RTCReceivedRtpStreamStats
@ -468,19 +481,20 @@ _get_stats_from_rtp_source_stats (TransportStream * stream,
gst_structure_set (r_out, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_out, "transport-id", G_TYPE_STRING, transport_id,
NULL);
if (kind)
gst_structure_set (r_out, "kind", G_TYPE_STRING, kind, NULL);
/* XXX: mediaType, trackId */
/* RTCSentRtpStreamStats */
guint sr_bytes = 0, sr_packets = 0;
if (have_sr) {
guint sr_bytes, sr_packets;
if (gst_structure_get_uint (source_stats, "sr-octet-count", &sr_bytes))
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT, sr_bytes, NULL);
if (gst_structure_get_uint (source_stats, "sr-packet-count", &sr_packets))
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT, sr_packets,
NULL);
gst_structure_get_uint (source_stats, "sr-octet-count", &sr_bytes);
gst_structure_get_uint (source_stats, "sr-packet-count", &sr_packets);
}
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT, sr_bytes, NULL);
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT, sr_packets, NULL);
/* RTCSentRtpStreamStats: