webrtc/stats: redo considering internal sources

Internal sources seem to be rtp streams we are sending whereas
non-internal sources are the rtp streams we are receiving. Redo the
statistics with that in mind.
This commit is contained in:
Matthew Waters 2019-08-12 19:42:12 +10:00
parent 08b53ca456
commit 523f4e4b50

View file

@ -82,6 +82,8 @@ _get_peer_connection_stats (GstWebRTCBin * webrtc)
}
#define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate)
#define FIXED_16_16_TO_DOUBLE(v) ((double) ((v & 0xffff0000) >> 16) + ((v & 0xffff) / 65536.0))
#define FIXED_32_32_TO_DOUBLE(v) ((double) ((v & G_GUINT64_CONSTANT (0xffffffff00000000)) >> 32) + ((v & G_GUINT64_CONSTANT (0xffffffff)) / 4294967296.0))
/* https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*
https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */
@ -90,50 +92,138 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
const GstStructure * source_stats, const gchar * codec_id,
const gchar * transport_id, GstStructure * s)
{
GstStructure *in, *out, *r_in, *r_out;
gchar *in_id, *out_id, *r_in_id, *r_out_id;
guint ssrc, fir, pli, nack, jitter;
int lost, clock_rate;
guint64 packets, bytes;
gboolean have_rb = FALSE, sent_rb = FALSE;
gboolean internal;
double ts;
gst_structure_get_double (s, "timestamp", &ts);
gst_structure_get_uint (source_stats, "ssrc", &ssrc);
gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
"sent_rb", G_TYPE_BOOLEAN, &sent_rb, "clock-rate", G_TYPE_INT,
&clock_rate, NULL);
gst_structure_get (source_stats, "ssrc", G_TYPE_UINT, &ssrc, "clock-rate",
G_TYPE_INT, &clock_rate, "internal", G_TYPE_BOOLEAN, &internal, NULL);
in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
if (internal) {
GstStructure *r_in, *out;
gchar *out_id, *r_in_id;
in = gst_structure_new_empty (in_id);
_set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
/* RTCStreamStats */
gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
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 */
r_in = gst_structure_new_empty (r_in_id);
_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
/* RTCReceivedRTPStreamStats */
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_uint64 (source_stats, "octets-received", &bytes))
gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_int (source_stats, "packets-lost", &lost))
gst_structure_set (in, "packets-lost", G_TYPE_INT, 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);
/* RTCStreamStats */
gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
gst_structure_set (r_in, "packets-received", G_TYPE_UINT64, packets,
NULL);
if (gst_structure_get_int (source_stats, "packets-lost", &lost))
gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
if (gst_structure_get_uint (source_stats, "jitter", &jitter))
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
/* XXX: RTCReceivedRTPStreamStats
double fractionLost;
unsigned long packetsDiscarded;
unsigned long packetsFailedDecryption;
unsigned long packetsRepaired;
unsigned long burstPacketsLost;
unsigned long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
*/
/* RTCRemoteInboundRTPStreamStats */
/* XXX: framesDecoded, lastPacketReceivedTimestamp */
out = gst_structure_new_empty (out_id);
_set_base_stats (out, GST_WEBRTC_STATS_OUTBOUND_RTP, ts, out_id);
/* RTCStreamStats */
gst_structure_set (out, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (out, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (out, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (gst_structure_get_uint (source_stats, "sent-fir-count", &fir))
gst_structure_set (out, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "sent-pli-count", &pli))
gst_structure_set (out, "pli-count", G_TYPE_UINT, pli, NULL);
if (gst_structure_get_uint (source_stats, "sent-nack-count", &nack))
gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCSentRTPStreamStats */
if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
/* XXX:
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
*/
/* RTCOutboundRTPStreamStats */
gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
/* XXX:
DOMHighResTimeStamp lastPacketSentTimestamp;
double targetBitrate;
unsigned long framesEncoded;
double totalEncodeTime;
double averageRTCPInterval;
*/
gst_structure_set (s, out_id, GST_TYPE_STRUCTURE, out, NULL);
gst_structure_set (s, r_in_id, GST_TYPE_STRUCTURE, r_in, NULL);
gst_structure_free (out);
gst_structure_free (r_in);
g_free (out_id);
g_free (r_in_id);
} else {
GstStructure *in, *r_out;
gchar *r_out_id, *in_id;
gboolean have_rb = FALSE, have_sr = FALSE;
gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
"have-sr", G_TYPE_BOOLEAN, &have_sr, 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);
in = gst_structure_new_empty (in_id);
_set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
/* RTCStreamStats */
gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
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 */
/* RTCReceivedRTPStreamStats */
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_uint64 (source_stats, "octets-received", &bytes))
gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_int (source_stats, "packets-lost", &lost))
gst_structure_set (in, "packets-lost", G_TYPE_INT, 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);
/*
RTCReceivedRTPStreamStats
double fractionLost;
@ -150,132 +240,65 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
double gapDiscardRate;
*/
/* RTCInboundRTPStreamStats */
gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
/* XXX: framesDecoded, lastPacketReceivedTimestamp */
/* RTCInboundRTPStreamStats */
gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
/* XXX: framesDecoded, lastPacketReceivedTimestamp */
r_in = gst_structure_new_empty (r_in_id);
_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
/* RTCStreamStats */
gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCReceivedRTPStreamStats */
if (sent_rb) {
if (gst_structure_get_uint (source_stats, "sent-rb-jitter", &jitter))
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
if (gst_structure_get_int (source_stats, "sent-rb-packetslost", &lost))
gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
/* packetsReceived, bytesReceived */
} else {
/* default values */
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE, 0.0, "packets-lost",
G_TYPE_INT, 0, NULL);
}
/* XXX: RTCReceivedRTPStreamStats
double fractionLost;
unsigned long packetsDiscarded;
unsigned long packetsFailedDecryption;
unsigned long packetsRepaired;
unsigned long burstPacketsLost;
unsigned long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
*/
/* RTCRemoteInboundRTPStreamStats */
gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
if (have_rb) {
guint32 rtt;
if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
/* 16.16 fixed point to double */
double val =
(double) ((rtt & 0xffff0000) >> 16) + ((rtt & 0xffff) / 65536.0);
gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
r_out = gst_structure_new_empty (r_out_id);
_set_base_stats (r_out, GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, ts, r_out_id);
/* RTCStreamStats */
gst_structure_set (r_out, "ssrc", G_TYPE_UINT, ssrc, NULL);
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 (have_rb) {
guint32 rtt;
if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
/* 16.16 fixed point to double */
double val = FIXED_16_16_TO_DOUBLE (rtt);
gst_structure_set (r_out, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
}
} else {
/* default values */
gst_structure_set (r_out, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
}
} else {
/* default values */
gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCSentRTPStreamStats */
if (have_sr) {
if (gst_structure_get_uint64 (source_stats, "sr-octet-count", &bytes))
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "sr-packet-count", &packets))
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT64, packets, NULL);
}
/* XXX:
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
*/
if (have_sr) {
guint64 ntptime;
if (gst_structure_get_uint64 (source_stats, "sr-ntptime", &ntptime)) {
/* 16.16 fixed point to double */
double val = FIXED_32_32_TO_DOUBLE (ntptime);
gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, val, NULL);
}
} else {
/* default values */
gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, 0.0, NULL);
}
gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
gst_structure_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);
gst_structure_free (in);
gst_structure_free (r_out);
g_free (in_id);
g_free (r_out_id);
}
/* XXX: framesDecoded, lastPacketReceivedTimestamp */
out = gst_structure_new_empty (out_id);
_set_base_stats (out, GST_WEBRTC_STATS_OUTBOUND_RTP, ts, out_id);
/* RTCStreamStats */
gst_structure_set (out, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (out, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (out, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (gst_structure_get_uint (source_stats, "sent-fir-count", &fir))
gst_structure_set (out, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "sent-pli-count", &pli))
gst_structure_set (out, "pli-count", G_TYPE_UINT, pli, NULL);
if (gst_structure_get_uint (source_stats, "sent-nack-count", &nack))
gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCSentRTPStreamStats */
if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
/* XXX:
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
*/
/* RTCOutboundRTPStreamStats */
gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
/* XXX:
DOMHighResTimeStamp lastPacketSentTimestamp;
double targetBitrate;
unsigned long framesEncoded;
double totalEncodeTime;
double averageRTCPInterval;
*/
r_out = gst_structure_new_empty (r_out_id);
_set_base_stats (r_out, GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, ts, r_out_id);
/* RTCStreamStats */
gst_structure_set (r_out, "ssrc", G_TYPE_UINT, ssrc, NULL);
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);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCSentRTPStreamStats */
/* if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT64, packets, NULL);*/
/* XXX:
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
*/
gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
gst_structure_set (s, out_id, GST_TYPE_STRUCTURE, out, NULL);
gst_structure_set (s, r_in_id, GST_TYPE_STRUCTURE, r_in, NULL);
gst_structure_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);
gst_structure_free (in);
gst_structure_free (out);
gst_structure_free (r_in);
gst_structure_free (r_out);
g_free (in_id);
g_free (out_id);
g_free (r_in_id);
g_free (r_out_id);
}
/* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* */
@ -441,16 +464,13 @@ _get_stats_from_transport_channel (GstWebRTCBin * webrtc,
for (i = 0; i < source_stats->n_values; i++) {
const GstStructure *stats;
const GValue *val = g_value_array_get_nth (source_stats, i);
gboolean internal;
guint stats_ssrc = 0;
stats = gst_value_get_structure (val);
/* skip internal or foreign sources */
gst_structure_get (stats,
"internal", G_TYPE_BOOLEAN, &internal,
"ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
if (internal || (ssrc && stats_ssrc && ssrc != stats_ssrc))
/* skip foreign sources */
gst_structure_get (stats, "ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
if (ssrc && stats_ssrc && ssrc != stats_ssrc)
continue;
_get_stats_from_rtp_source_stats (webrtc, stats, codec_id, transport_id, s);