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

Change-Id: I52a9bde7155d146056dbc8bf8c450d3534b67847
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>

diff --git a/ext/webrtc/gstwebrtcstats.c b/ext/webrtc/gstwebrtcstats.c
index 403b51e..09f2688 100644 (file)
--- a/ext/webrtc/gstwebrtcstats.c
+++ b/ext/webrtc/gstwebrtcstats.c
@@ -110,18 +110,19 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
     r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
 
-    gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
-        NULL);
+    gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
 
     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 */
+    /* RTCRtpStreamStats */
     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 */
+    /* To be added: kind */
+
+    /* RTCReceivedRtpStreamStats */
 
     if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
       gst_structure_set (r_in, "packets-received", G_TYPE_UINT64, packets,
@@ -132,20 +133,32 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
       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;
-*/
+    /* RTCReceivedRtpStreamStats:
+
+       To be added:
+
+       double             fractionLost;
+       unsigned long      packetsDiscarded;
+       unsigned long      packetsRepaired;
+       unsigned long      burstPacketsLost;
+       unsigned long      burstPacketsDiscarded;
+       unsigned long      burstLossCount;
+       unsigned long      burstDiscardCount;
+       double             burstLossRate;
+       double             burstDiscardRate;
+       double             gapLossRate;
+       double             gapDiscardRate;
+
+       Can't be implemented frame re-assembly happens after rtpbin:
+
+       unsigned long        framesDropped;
+       unsigned long        partialFramesLost;
+       unsigned long        fullFramesLost;
+     */
+
+
+    /* RTCRemoteInboundRTPStreamStats */
+
     if (have_rb) {
       guint32 rtt;
       if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
@@ -155,8 +168,15 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
       }
     }
 
-    /* RTCRemoteInboundRTPStreamStats */
-    /* XXX: framesDecoded, lastPacketReceivedTimestamp */
+    /* RTCRemoteInboundRTPStreamStats:
+
+       To be added:
+
+       DOMString            localId;
+       double               totalRoundTripTime;
+       unsigned long long   reportsReceived;
+       unsigned long long   roundTripTimeMeasurements;
+     */
 
     out = gst_structure_new_empty (out_id);
     _set_base_stats (out, GST_WEBRTC_STATS_OUTBOUND_RTP, ts, out_id);
@@ -165,6 +185,17 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     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);
+    /* To be added: kind */
+
+
+    /* 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);
+
+    /* RTCOutboundRTPStreamStats */
+
     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))
@@ -173,25 +204,55 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
       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;
-*/
+
+
+    /*  RTCOutboundRTPStreamStats:
+
+       To be added:
+
+       unsigned long        sliCount;
+       unsigned long        rtxSsrc;
+       DOMString            mediaSourceId;
+       DOMString            senderId;
+       DOMString            remoteId;
+       DOMString            rid;
+       DOMHighResTimeStamp  lastPacketSentTimestamp;
+       unsigned long long   headerBytesSent;
+       unsigned long        packetsDiscardedOnSend;
+       unsigned long long   bytesDiscardedOnSend;
+       unsigned long        fecPacketsSent;
+       unsigned long long   retransmittedPacketsSent;
+       unsigned long long   retransmittedBytesSent;
+       double               averageRtcpInterval;
+       record<USVString, unsigned long long> perDscpPacketsSent;
+
+       Not relevant because webrtcbin doesn't encode:
+
+       double               targetBitrate;
+       unsigned long long   totalEncodedBytesTarget;
+       unsigned long        frameWidth;
+       unsigned long        frameHeight;
+       unsigned long        frameBitDepth;
+       double               framesPerSecond;
+       unsigned long        framesSent;
+       unsigned long        hugeFramesSent;
+       unsigned long        framesEncoded;
+       unsigned long        keyFramesEncoded;
+       unsigned long        framesDiscardedOnSend;
+       unsigned long long   qpSum;
+       unsigned long long   totalSamplesSent;
+       unsigned long long   samplesEncodedWithSilk;
+       unsigned long long   samplesEncodedWithCelt;
+       boolean              voiceActivityFlag;
+       double               totalEncodeTime;
+       double               totalPacketSendDelay;
+       RTCQualityLimitationReason                 qualityLimitationReason;
+       record<DOMString, double> qualityLimitationDurations;
+       unsigned long        qualityLimitationResolutionChanges;
+       DOMString            encoderImplementation;
+     */
+
     gst_structure_set (s, out_id, GST_TYPE_STRUCTURE, out, NULL);
     gst_structure_set (s, r_in_id, GST_TYPE_STRUCTURE, r_in, NULL);
 
@@ -213,47 +274,108 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     in = gst_structure_new_empty (in_id);
     _set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
 
-    /* RTCStreamStats */
+    /* RTCRtpStreamStats */
     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 */
+    /* To be added: kind */
+
+
+    /* RTCReceivedRtpStreamStats */
 
-    /* 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;
-    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;
-*/
 
-    /* RTCInboundRTPStreamStats */
+    /*
+       RTCReceivedRtpStreamStats
+
+       To be added:
+
+       unsigned long long   packetsRepaired;
+       unsigned long long   burstPacketsLost;
+       unsigned long long   burstPacketsDiscarded;
+       unsigned long        burstLossCount;
+       unsigned long        burstDiscardCount;
+       double               burstLossRate;
+       double               burstDiscardRate;
+       double               gapLossRate;
+       double               gapDiscardRate;
+
+       Not relevant because webrtcbin doesn't decode:
+
+       unsigned long        framesDropped;
+       unsigned long        partialFramesLost;
+       unsigned long        fullFramesLost;
+     */
+
+    /* RTCInboundRtpStreamStats */
     gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
-    /* XXX: framesDecoded, lastPacketReceivedTimestamp */
+
+    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_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 */
+
+    /* RTCInboundRtpStreamStats:
+
+       To be added:
+
+       required DOMString   receiverId;
+       double               averageRtcpInterval;
+       unsigned long long   headerBytesReceived;
+       unsigned long long   fecPacketsReceived;
+       unsigned long long   fecPacketsDiscarded;
+       unsigned long long   bytesReceived;
+       unsigned long long   packetsFailedDecryption;
+       record<USVString, unsigned long long> perDscpPacketsReceived;
+       unsigned long        nackCount;
+       unsigned long        firCount;
+       unsigned long        pliCount;
+       unsigned long        sliCount;
+       double               jitterBufferDelay;
+
+       Not relevant because webrtcbin doesn't decode or depayload:
+       unsigned long        framesDecoded;
+       unsigned long        keyFramesDecoded;
+       unsigned long        frameWidth;
+       unsigned long        frameHeight;
+       unsigned long        frameBitDepth;
+       double               framesPerSecond;
+       unsigned long long   qpSum;
+       double               totalDecodeTime;
+       double               totalInterFrameDelay;
+       double               totalSquaredInterFrameDelay;
+       boolean              voiceActivityFlag;
+       DOMHighResTimeStamp  lastPacketReceivedTimestamp;
+       double               totalProcessingDelay;
+       DOMHighResTimeStamp  estimatedPlayoutTimestamp;
+       unsigned long long   jitterBufferEmittedCount;
+       unsigned long long   totalSamplesReceived;
+       unsigned long long   totalSamplesDecoded;
+       unsigned long long   samplesDecodedWithSilk;
+       unsigned long long   samplesDecodedWithCelt;
+       unsigned long long   concealedSamples;
+       unsigned long long   silentConcealedSamples;
+       unsigned long long   concealmentEvents;
+       unsigned long long   insertedSamplesForDeceleration;
+       unsigned long long   removedSamplesForAcceleration;
+       double               audioLevel;
+       double               totalAudioEnergy;
+       double               totalSamplesDuration;
+       unsigned long        framesReceived;
+       DOMString            decoderImplementation;
+     */
 
     r_out = gst_structure_new_empty (r_out_id);
     _set_base_stats (r_out, GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, ts, r_out_id);
@@ -262,19 +384,64 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     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 */
+    /* XXX: mediaType, trackId */
+
+    /* RTCSentRtpStreamStats */
 
-/* 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;
-*/
+
+    /* RTCSentRtpStreamStats:
+
+       To be added:
+
+       unsigned long        rtxSsrc;
+       DOMString            mediaSourceId;
+       DOMString            senderId;
+       DOMString            remoteId;
+       DOMString            rid;
+       DOMHighResTimeStamp  lastPacketSentTimestamp;
+       unsigned long long   headerBytesSent;
+       unsigned long        packetsDiscardedOnSend;
+       unsigned long long   bytesDiscardedOnSend;
+       unsigned long        fecPacketsSent;
+       unsigned long long   retransmittedPacketsSent;
+       unsigned long long   retransmittedBytesSent;
+       double               averageRtcpInterval;
+       unsigned long        sliCount;
+
+       Can't be implemented because we don't decode:
+
+       double               targetBitrate;
+       unsigned long long   totalEncodedBytesTarget;
+       unsigned long        frameWidth;
+       unsigned long        frameHeight;
+       unsigned long        frameBitDepth;
+       double               framesPerSecond;
+       unsigned long        framesSent;
+       unsigned long        hugeFramesSent;
+       unsigned long        framesEncoded;
+       unsigned long        keyFramesEncoded;
+       unsigned long        framesDiscardedOnSend;
+       unsigned long long   qpSum;
+       unsigned long long   totalSamplesSent;
+       unsigned long long   samplesEncodedWithSilk;
+       unsigned long long   samplesEncodedWithCelt;
+       boolean              voiceActivityFlag;
+       double               totalEncodeTime;
+       double               totalPacketSendDelay;
+       RTCQualityLimitationReason                 qualityLimitationReason;
+       record<DOMString, double> qualityLimitationDurations;
+       unsigned long        qualityLimitationResolutionChanges;
+       record<USVString, unsigned long long> perDscpPacketsSent;
+       DOMString            encoderImplementation;
+     */
+
+    /* RTCRemoteOutboundRtpStreamStats */
 
     if (have_sr) {
       guint64 ntptime;
@@ -290,6 +457,10 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
 
     gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
 
+    /* To be added:
+       reportsSent
+     */
+
     gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
     gst_structure_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);