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webrtcstats: Document all RTP missing fields according to the latest spec
Just document all the missing fields and document which ones will never be implemented because they depend on the codec or depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
This commit is contained in:
Olivier Crête
GStreamer Merge Bot
parent
895ea210c2
commit
d9d7814182
1 changed files with 244 additions and 73 deletions
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@ -110,18 +110,19 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
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r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
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gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
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NULL);
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gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
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r_in = gst_structure_new_empty (r_in_id);
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_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
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/* RTCStreamStats */
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/* RTCRtpStreamStats */
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gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
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gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, 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_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* To be added: kind */
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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 (r_in, "packets-received", G_TYPE_UINT64, packets,
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@ -132,20 +133,32 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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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/* RTCReceivedRtpStreamStats:
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To be added:
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double fractionLost;
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unsigned long packetsDiscarded;
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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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Can't be implemented frame re-assembly happens after rtpbin:
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unsigned long framesDropped;
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unsigned long partialFramesLost;
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unsigned long fullFramesLost;
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*/
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/* RTCRemoteInboundRTPStreamStats */
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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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@ -155,8 +168,15 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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}
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}
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/* RTCRemoteInboundRTPStreamStats */
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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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/* RTCRemoteInboundRTPStreamStats:
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To be added:
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DOMString localId;
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double totalRoundTripTime;
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unsigned long long reportsReceived;
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unsigned long long roundTripTimeMeasurements;
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*/
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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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@ -165,6 +185,17 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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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/* To be added: kind */
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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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/* RTCOutboundRTPStreamStats */
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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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@ -173,25 +204,55 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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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/* RTCOutboundRTPStreamStats:
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To be added:
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unsigned long sliCount;
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unsigned long rtxSsrc;
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DOMString mediaSourceId;
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DOMString senderId;
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DOMString remoteId;
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DOMString rid;
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DOMHighResTimeStamp lastPacketSentTimestamp;
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unsigned long long headerBytesSent;
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unsigned long packetsDiscardedOnSend;
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unsigned long long bytesDiscardedOnSend;
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unsigned long fecPacketsSent;
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unsigned long long retransmittedPacketsSent;
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unsigned long long retransmittedBytesSent;
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double averageRtcpInterval;
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record<USVString, unsigned long long> perDscpPacketsSent;
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Not relevant because webrtcbin doesn't encode:
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double targetBitrate;
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unsigned long long totalEncodedBytesTarget;
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unsigned long frameWidth;
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unsigned long frameHeight;
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unsigned long frameBitDepth;
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double framesPerSecond;
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unsigned long framesSent;
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unsigned long hugeFramesSent;
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unsigned long framesEncoded;
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unsigned long keyFramesEncoded;
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unsigned long framesDiscardedOnSend;
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unsigned long long qpSum;
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unsigned long long totalSamplesSent;
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unsigned long long samplesEncodedWithSilk;
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unsigned long long samplesEncodedWithCelt;
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boolean voiceActivityFlag;
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double totalEncodeTime;
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double totalPacketSendDelay;
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RTCQualityLimitationReason qualityLimitationReason;
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record<DOMString, double> qualityLimitationDurations;
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unsigned long qualityLimitationResolutionChanges;
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DOMString encoderImplementation;
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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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@ -213,10 +274,51 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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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/* RTCRtpStreamStats */
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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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/* To be added: kind */
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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_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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RTCReceivedRtpStreamStats
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To be added:
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unsigned long long packetsRepaired;
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unsigned long long burstPacketsLost;
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unsigned long 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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Not relevant because webrtcbin doesn't decode:
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unsigned long framesDropped;
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unsigned long partialFramesLost;
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unsigned long fullFramesLost;
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*/
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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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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_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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@ -225,35 +327,55 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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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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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/* 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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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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To be added:
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required DOMString receiverId;
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double averageRtcpInterval;
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unsigned long long headerBytesReceived;
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unsigned long long fecPacketsReceived;
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unsigned long long fecPacketsDiscarded;
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unsigned long long bytesReceived;
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unsigned long long packetsFailedDecryption;
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record<USVString, unsigned long long> perDscpPacketsReceived;
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unsigned long nackCount;
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unsigned long firCount;
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unsigned long pliCount;
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unsigned long sliCount;
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double jitterBufferDelay;
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Not relevant because webrtcbin doesn't decode or depayload:
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unsigned long framesDecoded;
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unsigned long keyFramesDecoded;
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unsigned long frameWidth;
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unsigned long frameHeight;
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unsigned long frameBitDepth;
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double framesPerSecond;
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unsigned long long qpSum;
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double totalDecodeTime;
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double totalInterFrameDelay;
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double totalSquaredInterFrameDelay;
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boolean voiceActivityFlag;
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DOMHighResTimeStamp lastPacketReceivedTimestamp;
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double totalProcessingDelay;
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DOMHighResTimeStamp estimatedPlayoutTimestamp;
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unsigned long long jitterBufferEmittedCount;
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unsigned long long totalSamplesReceived;
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unsigned long long totalSamplesDecoded;
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unsigned long long samplesDecodedWithSilk;
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unsigned long long samplesDecodedWithCelt;
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unsigned long long concealedSamples;
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unsigned long long silentConcealedSamples;
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unsigned long long concealmentEvents;
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unsigned long long insertedSamplesForDeceleration;
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unsigned long long removedSamplesForAcceleration;
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double audioLevel;
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double totalAudioEnergy;
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double totalSamplesDuration;
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unsigned long framesReceived;
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DOMString decoderImplementation;
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*/
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r_out = gst_structure_new_empty (r_out_id);
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_set_base_stats (r_out, GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, ts, r_out_id);
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@ -262,19 +384,64 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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_out, "transport-id", G_TYPE_STRING, transport_id,
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NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* XXX: mediaType, trackId */
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/* RTCSentRtpStreamStats */
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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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/* RTCSentRtpStreamStats:
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To be added:
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unsigned long rtxSsrc;
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DOMString mediaSourceId;
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DOMString senderId;
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DOMString remoteId;
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DOMString rid;
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DOMHighResTimeStamp lastPacketSentTimestamp;
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unsigned long long headerBytesSent;
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unsigned long packetsDiscardedOnSend;
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unsigned long long bytesDiscardedOnSend;
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unsigned long fecPacketsSent;
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unsigned long long retransmittedPacketsSent;
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unsigned long long retransmittedBytesSent;
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double averageRtcpInterval;
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unsigned long sliCount;
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Can't be implemented because we don't decode:
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double targetBitrate;
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unsigned long long totalEncodedBytesTarget;
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unsigned long frameWidth;
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unsigned long frameHeight;
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unsigned long frameBitDepth;
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double framesPerSecond;
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unsigned long framesSent;
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unsigned long hugeFramesSent;
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unsigned long framesEncoded;
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unsigned long keyFramesEncoded;
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unsigned long framesDiscardedOnSend;
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unsigned long long qpSum;
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unsigned long long totalSamplesSent;
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unsigned long long samplesEncodedWithSilk;
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unsigned long long samplesEncodedWithCelt;
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boolean voiceActivityFlag;
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double totalEncodeTime;
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double totalPacketSendDelay;
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RTCQualityLimitationReason qualityLimitationReason;
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record<DOMString, double> qualityLimitationDurations;
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unsigned long qualityLimitationResolutionChanges;
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record<USVString, unsigned long long> perDscpPacketsSent;
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DOMString encoderImplementation;
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*/
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/* RTCRemoteOutboundRtpStreamStats */
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if (have_sr) {
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guint64 ntptime;
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@ -290,6 +457,10 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
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/* To be added:
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reportsSent
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*/
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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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