mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-17 03:35:21 +00:00
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
parent
08b53ca456
commit
523f4e4b50
1 changed files with 183 additions and 163 deletions
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@ -82,6 +82,8 @@ _get_peer_connection_stats (GstWebRTCBin * webrtc)
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}
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}
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#define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate)
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#define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate)
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#define FIXED_16_16_TO_DOUBLE(v) ((double) ((v & 0xffff0000) >> 16) + ((v & 0xffff) / 65536.0))
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#define FIXED_32_32_TO_DOUBLE(v) ((double) ((v & G_GUINT64_CONSTANT (0xffffffff00000000)) >> 32) + ((v & G_GUINT64_CONSTANT (0xffffffff)) / 4294967296.0))
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/* https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*
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/* https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*
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https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */
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https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */
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@ -90,50 +92,138 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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const GstStructure * source_stats, const gchar * codec_id,
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const GstStructure * source_stats, const gchar * codec_id,
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const gchar * transport_id, GstStructure * s)
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const gchar * transport_id, GstStructure * s)
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{
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{
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GstStructure *in, *out, *r_in, *r_out;
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gchar *in_id, *out_id, *r_in_id, *r_out_id;
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guint ssrc, fir, pli, nack, jitter;
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guint ssrc, fir, pli, nack, jitter;
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int lost, clock_rate;
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int lost, clock_rate;
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guint64 packets, bytes;
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guint64 packets, bytes;
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gboolean have_rb = FALSE, sent_rb = FALSE;
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gboolean internal;
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double ts;
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double ts;
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gst_structure_get_double (s, "timestamp", &ts);
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gst_structure_get_double (s, "timestamp", &ts);
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gst_structure_get_uint (source_stats, "ssrc", &ssrc);
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gst_structure_get (source_stats, "ssrc", G_TYPE_UINT, &ssrc, "clock-rate",
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gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
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G_TYPE_INT, &clock_rate, "internal", G_TYPE_BOOLEAN, &internal, NULL);
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"sent_rb", G_TYPE_BOOLEAN, &sent_rb, "clock-rate", G_TYPE_INT,
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&clock_rate, NULL);
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in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
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if (internal) {
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out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
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GstStructure *r_in, *out;
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r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
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gchar *out_id, *r_in_id;
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r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
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in = gst_structure_new_empty (in_id);
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out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
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_set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
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r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
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/* RTCStreamStats */
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r_in = gst_structure_new_empty (r_in_id);
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gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
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_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
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gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
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if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
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gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
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if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
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gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
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if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
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gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* RTCReceivedRTPStreamStats */
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/* RTCStreamStats */
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if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
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gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
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gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
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gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
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if (gst_structure_get_uint64 (source_stats, "octets-received", &bytes))
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gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
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gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
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if (gst_structure_get_int (source_stats, "packets-lost", &lost))
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/* XXX: mediaType, trackId, sliCount, qpSum */
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gst_structure_set (in, "packets-lost", G_TYPE_INT, lost, NULL);
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if (gst_structure_get_uint (source_stats, "jitter", &jitter))
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if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
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gst_structure_set (in, "jitter", G_TYPE_DOUBLE,
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gst_structure_set (r_in, "packets-received", G_TYPE_UINT64, packets,
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CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
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NULL);
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if (gst_structure_get_int (source_stats, "packets-lost", &lost))
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gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
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if (gst_structure_get_uint (source_stats, "jitter", &jitter))
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gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
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CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
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/* XXX: RTCReceivedRTPStreamStats
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double fractionLost;
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unsigned long packetsDiscarded;
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unsigned long packetsFailedDecryption;
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unsigned long packetsRepaired;
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unsigned long burstPacketsLost;
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unsigned long burstPacketsDiscarded;
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unsigned long burstLossCount;
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unsigned long burstDiscardCount;
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double burstLossRate;
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double burstDiscardRate;
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double gapLossRate;
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double gapDiscardRate;
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*/
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/* RTCRemoteInboundRTPStreamStats */
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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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out = gst_structure_new_empty (out_id);
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_set_base_stats (out, GST_WEBRTC_STATS_OUTBOUND_RTP, ts, out_id);
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/* RTCStreamStats */
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gst_structure_set (out, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (out, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (out, "transport-id", G_TYPE_STRING, transport_id, NULL);
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if (gst_structure_get_uint (source_stats, "sent-fir-count", &fir))
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gst_structure_set (out, "fir-count", G_TYPE_UINT, fir, NULL);
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if (gst_structure_get_uint (source_stats, "sent-pli-count", &pli))
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gst_structure_set (out, "pli-count", G_TYPE_UINT, pli, NULL);
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if (gst_structure_get_uint (source_stats, "sent-nack-count", &nack))
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gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* RTCSentRTPStreamStats */
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if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
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gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
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if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
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gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
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/* XXX:
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unsigned long packetsDiscardedOnSend;
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unsigned long long bytesDiscardedOnSend;
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*/
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/* RTCOutboundRTPStreamStats */
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gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
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/* XXX:
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DOMHighResTimeStamp lastPacketSentTimestamp;
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double targetBitrate;
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unsigned long framesEncoded;
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double totalEncodeTime;
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double averageRTCPInterval;
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*/
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gst_structure_set (s, out_id, GST_TYPE_STRUCTURE, out, NULL);
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gst_structure_set (s, r_in_id, GST_TYPE_STRUCTURE, r_in, NULL);
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gst_structure_free (out);
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gst_structure_free (r_in);
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g_free (out_id);
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g_free (r_in_id);
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} else {
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GstStructure *in, *r_out;
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gchar *r_out_id, *in_id;
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gboolean have_rb = FALSE, have_sr = FALSE;
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gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
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"have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
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in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
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r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
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in = gst_structure_new_empty (in_id);
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_set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
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/* RTCStreamStats */
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gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
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if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
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gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
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if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
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gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
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if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
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gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* RTCReceivedRTPStreamStats */
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if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
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gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
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if (gst_structure_get_uint64 (source_stats, "octets-received", &bytes))
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gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
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if (gst_structure_get_int (source_stats, "packets-lost", &lost))
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gst_structure_set (in, "packets-lost", G_TYPE_INT, lost, NULL);
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if (gst_structure_get_uint (source_stats, "jitter", &jitter))
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gst_structure_set (in, "jitter", G_TYPE_DOUBLE,
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CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
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/*
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/*
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RTCReceivedRTPStreamStats
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RTCReceivedRTPStreamStats
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double fractionLost;
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double fractionLost;
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@ -150,132 +240,65 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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double gapDiscardRate;
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double gapDiscardRate;
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*/
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*/
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/* RTCInboundRTPStreamStats */
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/* RTCInboundRTPStreamStats */
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gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
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gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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r_in = gst_structure_new_empty (r_in_id);
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r_out = gst_structure_new_empty (r_out_id);
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_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
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_set_base_stats (r_out, GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, ts, r_out_id);
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/* RTCStreamStats */
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/* RTCStreamStats */
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gst_structure_set (r_out, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (r_out, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (r_out, "transport-id", G_TYPE_STRING, transport_id,
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gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
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NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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if (have_rb) {
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guint32 rtt;
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/* RTCReceivedRTPStreamStats */
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if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
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if (sent_rb) {
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/* 16.16 fixed point to double */
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if (gst_structure_get_uint (source_stats, "sent-rb-jitter", &jitter))
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double val = FIXED_16_16_TO_DOUBLE (rtt);
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gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
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gst_structure_set (r_out, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
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CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
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}
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if (gst_structure_get_int (source_stats, "sent-rb-packetslost", &lost))
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} else {
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gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
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/* default values */
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/* packetsReceived, bytesReceived */
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gst_structure_set (r_out, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
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} else {
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/* default values */
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gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE, 0.0, "packets-lost",
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G_TYPE_INT, 0, NULL);
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}
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/* XXX: RTCReceivedRTPStreamStats
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double fractionLost;
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unsigned long packetsDiscarded;
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unsigned long packetsFailedDecryption;
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unsigned long packetsRepaired;
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unsigned long burstPacketsLost;
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unsigned long burstPacketsDiscarded;
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unsigned long burstLossCount;
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unsigned long burstDiscardCount;
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double burstLossRate;
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double burstDiscardRate;
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double gapLossRate;
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double gapDiscardRate;
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*/
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/* RTCRemoteInboundRTPStreamStats */
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gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
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if (have_rb) {
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guint32 rtt;
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if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
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/* 16.16 fixed point to double */
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double val =
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(double) ((rtt & 0xffff0000) >> 16) + ((rtt & 0xffff) / 65536.0);
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gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
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}
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}
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} else {
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* default values */
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gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
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/* RTCSentRTPStreamStats */
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if (have_sr) {
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if (gst_structure_get_uint64 (source_stats, "sr-octet-count", &bytes))
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gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
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if (gst_structure_get_uint64 (source_stats, "sr-packet-count", &packets))
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gst_structure_set (r_out, "packets-sent", G_TYPE_UINT64, packets, NULL);
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}
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/* XXX:
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unsigned long packetsDiscardedOnSend;
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unsigned long long bytesDiscardedOnSend;
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*/
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if (have_sr) {
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guint64 ntptime;
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if (gst_structure_get_uint64 (source_stats, "sr-ntptime", &ntptime)) {
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/* 16.16 fixed point to double */
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double val = FIXED_32_32_TO_DOUBLE (ntptime);
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gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, val, NULL);
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}
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} else {
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/* default values */
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gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, 0.0, NULL);
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}
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gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
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gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
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gst_structure_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);
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gst_structure_free (in);
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gst_structure_free (r_out);
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g_free (in_id);
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g_free (r_out_id);
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}
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}
|
||||||
/* 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* */
|
/* 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++) {
|
for (i = 0; i < source_stats->n_values; i++) {
|
||||||
const GstStructure *stats;
|
const GstStructure *stats;
|
||||||
const GValue *val = g_value_array_get_nth (source_stats, i);
|
const GValue *val = g_value_array_get_nth (source_stats, i);
|
||||||
gboolean internal;
|
|
||||||
guint stats_ssrc = 0;
|
guint stats_ssrc = 0;
|
||||||
|
|
||||||
stats = gst_value_get_structure (val);
|
stats = gst_value_get_structure (val);
|
||||||
|
|
||||||
/* skip internal or foreign sources */
|
/* skip foreign sources */
|
||||||
gst_structure_get (stats,
|
gst_structure_get (stats, "ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
|
||||||
"internal", G_TYPE_BOOLEAN, &internal,
|
if (ssrc && stats_ssrc && ssrc != stats_ssrc)
|
||||||
"ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
|
|
||||||
if (internal || (ssrc && stats_ssrc && ssrc != stats_ssrc))
|
|
||||||
continue;
|
continue;
|
||||||
|
|
||||||
_get_stats_from_rtp_source_stats (webrtc, stats, codec_id, transport_id, s);
|
_get_stats_from_rtp_source_stats (webrtc, stats, codec_id, transport_id, s);
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue