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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 02:01:12 +00:00
webrtcstats: Get the remote-inbound stats from the right RTPSource
This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
This commit is contained in:
Olivier Crête
GStreamer Merge Bot
parent
1c1661b54f
commit
1deb034e3d
1 changed files with 131 additions and 83 deletions
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@ -99,6 +99,105 @@ _gst_structure_take_structure (GstStructure * s, const char *fieldname,
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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/#remoteinboundrtpstats-dict* */
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static gboolean
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_get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc,
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TransportStream * stream, const GstStructure * source_stats,
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guint ssrc, guint clock_rate, const gchar * codec_id,
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const gchar * transport_id, GstStructure * s)
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{
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gboolean have_rb = FALSE, internal = FALSE;
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int lost;
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GstStructure *r_in;
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gchar *r_in_id, *out_id;
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guint32 rtt;
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guint fraction_lost, jitter;
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double ts;
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gst_structure_get_double (s, "timestamp", &ts);
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gst_structure_get (source_stats, "internal", G_TYPE_BOOLEAN, &internal,
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"have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
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/* This isn't what we're looking for */
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if (internal == TRUE || have_rb == FALSE)
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return FALSE;
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r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
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out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
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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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/* 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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/* To be added: kind */
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/* RTCReceivedRtpStreamStats */
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if (gst_structure_get_int (source_stats, "rb-packetslost", &lost))
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gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
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if (clock_rate && gst_structure_get_uint (source_stats, "rb-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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/* RTCReceivedRtpStreamStats:
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unsigned long long packetsReceived;
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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 (gst_structure_get_uint (source_stats, "rb-fractionlost", &fraction_lost))
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gst_structure_set (r_in, "fraction-lost", G_TYPE_DOUBLE,
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(double) fraction_lost / 256.0, NULL);
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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 = FIXED_16_16_TO_DOUBLE (rtt);
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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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/* 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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gst_structure_set (r_in, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
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source_stats, NULL);
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_gst_structure_take_structure (s, r_in_id, &r_in);
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g_free (r_in_id);
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g_free (out_id);
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return TRUE;
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}
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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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static void
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@ -107,7 +206,7 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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const gchar * codec_id, const gchar * transport_id, GstStructure * s)
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{
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guint ssrc, fir, pli, nack, jitter;
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int lost, clock_rate;
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int clock_rate;
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guint64 packets, bytes;
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gboolean internal;
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double ts;
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@ -117,71 +216,10 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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G_TYPE_INT, &clock_rate, "internal", G_TYPE_BOOLEAN, &internal, NULL);
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if (internal) {
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GstStructure *r_in, *out;
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GstStructure *out;
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gchar *out_id, *r_in_id;
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gboolean have_rb = FALSE;
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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, 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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/* 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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/* To be added: kind */
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/* RTCReceivedRtpStreamStats:
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To be added:
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unsigned long long packetsReceived;
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long long packetsLost;
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double jitter;
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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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/* 16.16 fixed point to double */
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double val = FIXED_16_16_TO_DOUBLE (rtt);
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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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/* 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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@ -209,8 +247,10 @@ _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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gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
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r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
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if (gst_structure_has_field (s, r_in_id))
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gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
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g_free (r_in_id);
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/* RTCOutboundRTPStreamStats:
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@ -264,12 +304,9 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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gst_structure_set (out, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
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source_stats, NULL);
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_gst_structure_take_structure (s, out_id, &out);
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_gst_structure_take_structure (s, r_in_id, &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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@ -427,10 +464,13 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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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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guint sr_bytes, sr_packets;
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if (gst_structure_get_uint (source_stats, "sr-octet-count", &sr_bytes))
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gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT, sr_bytes, NULL);
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if (gst_structure_get_uint (source_stats, "sr-packet-count", &sr_packets))
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gst_structure_set (r_out, "packets-sent", G_TYPE_UINT, sr_packets,
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NULL);
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}
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/* RTCSentRtpStreamStats:
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@ -642,7 +682,7 @@ _get_stats_from_dtls_transport (GstWebRTCBin * webrtc,
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static void
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_get_stats_from_transport_channel (GstWebRTCBin * webrtc,
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TransportStream * stream, const gchar * codec_id, guint ssrc,
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GstStructure * s)
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guint clock_rate, GstStructure * s)
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{
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GstWebRTCDTLSTransport *transport;
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GObject *rtp_session;
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@ -682,11 +722,14 @@ _get_stats_from_transport_channel (GstWebRTCBin * webrtc,
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/* skip foreign sources */
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gst_structure_get (stats, "ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
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if (ssrc && stats_ssrc && ssrc != stats_ssrc)
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continue;
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_get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
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transport_id, s);
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if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
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ssrc == stats_ssrc)
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_get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
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transport_id, s);
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else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc) &&
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ssrc == stats_ssrc)
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_get_stats_from_remote_rtp_source_stats (webrtc, stream, stats, ssrc,
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clock_rate, codec_id, transport_id, s);
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}
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g_object_unref (rtp_session);
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@ -698,13 +741,14 @@ _get_stats_from_transport_channel (GstWebRTCBin * webrtc,
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/* https://www.w3.org/TR/webrtc-stats/#codec-dict* */
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static void
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_get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
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GstStructure * s, gchar ** out_id, guint * out_ssrc)
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GstStructure * s, gchar ** out_id, guint * out_ssrc, guint * out_clock_rate)
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{
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GstStructure *stats;
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GstCaps *caps;
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gchar *id;
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double ts;
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guint ssrc = 0;
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gint clock_rate = 0;
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gst_structure_get_double (s, "timestamp", &ts);
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@ -715,7 +759,7 @@ _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
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caps = gst_pad_get_current_caps (pad);
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if (caps && gst_caps_is_fixed (caps)) {
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GstStructure *caps_s = gst_caps_get_structure (caps, 0);
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gint pt, clock_rate;
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gint pt;
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if (gst_structure_get_int (caps_s, "payload", &pt))
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gst_structure_set (stats, "payload-type", G_TYPE_UINT, pt, NULL);
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@ -742,6 +786,9 @@ _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
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if (out_ssrc)
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*out_ssrc = ssrc;
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if (out_clock_rate)
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*out_clock_rate = clock_rate;
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}
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static gboolean
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@ -750,9 +797,9 @@ _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
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GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
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TransportStream *stream;
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gchar *codec_id;
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guint ssrc;
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guint ssrc, clock_rate;
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_get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc);
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_get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc, &clock_rate);
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if (!wpad->trans)
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goto out;
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@ -761,7 +808,8 @@ _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
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if (!stream)
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goto out;
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_get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc, s);
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_get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc,
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clock_rate, s);
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out:
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g_free (codec_id);
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