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webrtcstats: Extract statistics from the rtpjitterbuffer

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And expose them as standardised webrtc statistics

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
This commit is contained in:
Olivier Crête 2020-10-09 20:27:40 -04:00 committed by GStreamer Merge Bot
parent fc0f6db856
commit ba0dfa52d2
1 changed files with 39 additions and 7 deletions
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46
ext/webrtc/gstwebrtcstats.c
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@ -89,8 +89,8 @@ _get_peer_connection_stats (GstWebRTCBin * webrtc)
https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */
static void
_get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
const GstStructure * source_stats, const gchar * codec_id,
const gchar * transport_id, GstStructure * s)
TransportStream * stream, const GstStructure * source_stats,
const gchar * codec_id, const gchar * transport_id, GstStructure * s)
{
guint ssrc, fir, pli, nack, jitter;
int lost, clock_rate;
@ -265,9 +265,33 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
GstStructure *in, *r_out;
gchar *r_out_id, *in_id;
gboolean have_sr = FALSE;
GstStructure *jb_stats = NULL;
guint i;
guint64 jb_lost, duplicates, late, rtx_success;
gst_structure_get (source_stats, "have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
for (i = 0; i < stream->remote_ssrcmap->len; i++) {
SsrcMapItem *item =
&g_array_index (stream->remote_ssrcmap, SsrcMapItem, i);
if (item->ssrc == ssrc) {
GObject *jb = g_weak_ref_get (&item->rtpjitterbuffer);
if (jb) {
g_object_get (jb, "stats", &jb_stats, NULL);
g_object_unref (jb);
}
break;
}
}
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);
@ -285,18 +309,21 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
if (gst_structure_get_int (source_stats, "packets-lost", &lost))
gst_structure_set (in, "packets-lost", G_TYPE_INT, lost, NULL);
if (jb_stats)
gst_structure_set (in, "packets-lost", G_TYPE_UINT64, jb_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);
/*
RTCReceivedRtpStreamStats
To be added:
unsigned long long packetsRepaired;
unsigned long long burstPacketsLost;
unsigned long long burstPacketsDiscarded;
unsigned long burstLossCount;
@ -325,7 +352,9 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
if (jb_stats)
gst_structure_set (in, "packets-duplicated", G_TYPE_UINT64, duplicates,
NULL);
/* RTCInboundRtpStreamStats:
@ -461,6 +490,8 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
reportsSent
*/
gst_structure_free (jb_stats);
gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
gst_structure_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);
@ -642,7 +673,8 @@ _get_stats_from_transport_channel (GstWebRTCBin * webrtc,
if (ssrc && stats_ssrc && ssrc != stats_ssrc)
continue;
_get_stats_from_rtp_source_stats (webrtc, stats, codec_id, transport_id, s);
_get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
transport_id, s);
}
g_object_unref (rtp_session);