/* GStreamer * Copyright (C) 2017 Matthew Waters * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif /* for GValueArray... */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #include "gstwebrtcstats.h" #include "gstwebrtcbin.h" #include "transportstream.h" #include "transportreceivebin.h" #include "utils.h" #include "webrtctransceiver.h" #define GST_CAT_DEFAULT gst_webrtc_stats_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); static void _init_debug (void) { static gsize _init = 0; if (g_once_init_enter (&_init)) { GST_DEBUG_CATEGORY_INIT (gst_webrtc_stats_debug, "webrtcstats", 0, "webrtcstats"); g_once_init_leave (&_init, 1); } } static double monotonic_time_as_double_milliseconds (void) { return g_get_monotonic_time () / 1000.0; } static void _set_base_stats (GstStructure * s, GstWebRTCStatsType type, double ts, const char *id) { gchar *name = _enum_value_to_string (GST_TYPE_WEBRTC_STATS_TYPE, type); g_return_if_fail (name != NULL); gst_structure_set_name (s, name); gst_structure_set (s, "type", GST_TYPE_WEBRTC_STATS_TYPE, type, "timestamp", G_TYPE_DOUBLE, ts, "id", G_TYPE_STRING, id, NULL); g_free (name); } static GstStructure * _get_peer_connection_stats (GstWebRTCBin * webrtc) { GstStructure *s = gst_structure_new_empty ("unused"); /* FIXME: datachannel */ gst_structure_set (s, "data-channels-opened", G_TYPE_UINT, 0, "data-channels-closed", G_TYPE_UINT, 0, "data-channels-requested", G_TYPE_UINT, 0, "data-channels-accepted", G_TYPE_UINT, 0, NULL); return s; } #define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate) #define FIXED_16_16_TO_DOUBLE(v) ((double) ((v & 0xffff0000) >> 16) + ((v & 0xffff) / 65536.0)) #define FIXED_32_32_TO_DOUBLE(v) ((double) ((v & G_GUINT64_CONSTANT (0xffffffff00000000)) >> 32) + ((v & G_GUINT64_CONSTANT (0xffffffff)) / 4294967296.0)) /* https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict* https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */ static void _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc, TransportStream * stream, const GstStructure * source_stats, const gchar * codec_id, const gchar * transport_id, GstStructure * s) { guint ssrc, fir, pli, nack, jitter; int lost, clock_rate; guint64 packets, bytes; gboolean internal; double ts; gst_structure_get_double (s, "timestamp", &ts); gst_structure_get (source_stats, "ssrc", G_TYPE_UINT, &ssrc, "clock-rate", G_TYPE_INT, &clock_rate, "internal", G_TYPE_BOOLEAN, &internal, NULL); if (internal) { GstStructure *r_in, *out; gchar *out_id, *r_in_id; gboolean have_rb = FALSE; 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); r_in = gst_structure_new_empty (r_in_id); _set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id); /* 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); /* To be added: kind */ /* RTCReceivedRtpStreamStats: To be added: unsigned long long packetsReceived; long long packetsLost; double jitter; 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)) { /* 16.16 fixed point to double */ double val = FIXED_16_16_TO_DOUBLE (rtt); gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL); } } /* 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); /* 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); /* 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)) 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 */ gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL); /* 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 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 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); gst_structure_free (out); gst_structure_free (r_in); g_free (out_id); g_free (r_in_id); } else { GstStructure *in, *r_out; gchar *r_out_id, *in_id; gboolean have_sr = FALSE; GstStructure *jb_stats = NULL; guint i; guint64 jb_lost, duplicates, late, rtx_success; gst_structure_get (source_stats, "have-sr", G_TYPE_BOOLEAN, &have_sr, NULL); for (i = 0; i < stream->remote_ssrcmap->len; i++) { SsrcMapItem *item = &g_array_index (stream->remote_ssrcmap, SsrcMapItem, i); if (item->ssrc == ssrc) { GObject *jb = g_weak_ref_get (&item->rtpjitterbuffer); if (jb) { g_object_get (jb, "stats", &jb_stats, NULL); g_object_unref (jb); } break; } } if (jb_stats) gst_structure_get (jb_stats, "num-lost", G_TYPE_UINT64, &jb_lost, "num-duplicates", G_TYPE_UINT64, &duplicates, "num-late", G_TYPE_UINT64, &late, "rtx-success-count", G_TYPE_UINT64, &rtx_success, NULL); in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc); r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc); in = gst_structure_new_empty (in_id); _set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id); /* 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); /* To be added: kind */ /* RTCReceivedRtpStreamStats */ if (gst_structure_get_uint64 (source_stats, "packets-received", &packets)) gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL); if (jb_stats) gst_structure_set (in, "packets-lost", G_TYPE_UINT64, jb_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); if (jb_stats) gst_structure_set (in, "packets-discarded", G_TYPE_UINT64, late, "packets-repaired", G_TYPE_UINT64, rtx_success, NULL); /* RTCReceivedRtpStreamStats To be added: 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); 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); if (jb_stats) gst_structure_set (in, "packets-duplicated", G_TYPE_UINT64, duplicates, NULL); /* 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 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); /* 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 */ /* 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); } /* 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 qualityLimitationDurations; unsigned long qualityLimitationResolutionChanges; record perDscpPacketsSent; DOMString encoderImplementation; */ /* RTCRemoteOutboundRtpStreamStats */ if (have_sr) { guint64 ntptime; if (gst_structure_get_uint64 (source_stats, "sr-ntptime", &ntptime)) { /* 16.16 fixed point to double */ double val = FIXED_32_32_TO_DOUBLE (ntptime); gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, val, NULL); } } else { /* default values */ gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, 0.0, NULL); } gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL); /* To be added: reportsSent */ gst_structure_free (jb_stats); gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL); gst_structure_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL); gst_structure_free (in); gst_structure_free (r_out); g_free (in_id); g_free (r_out_id); } } /* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* */ static gchar * _get_stats_from_ice_transport (GstWebRTCBin * webrtc, GstWebRTCICETransport * transport, GstStructure * s) { GstStructure *stats; gchar *id; double ts; gst_structure_get_double (s, "timestamp", &ts); id = g_strdup_printf ("ice-candidate-pair_%s", GST_OBJECT_NAME (transport)); stats = gst_structure_new_empty (id); _set_base_stats (stats, GST_WEBRTC_STATS_TRANSPORT, ts, id); /* XXX: RTCIceCandidatePairStats DOMString transportId; DOMString localCandidateId; DOMString remoteCandidateId; RTCStatsIceCandidatePairState state; unsigned long long priority; boolean nominated; unsigned long packetsSent; unsigned long packetsReceived; unsigned long long bytesSent; unsigned long long bytesReceived; DOMHighResTimeStamp lastPacketSentTimestamp; DOMHighResTimeStamp lastPacketReceivedTimestamp; DOMHighResTimeStamp firstRequestTimestamp; DOMHighResTimeStamp lastRequestTimestamp; DOMHighResTimeStamp lastResponseTimestamp; double totalRoundTripTime; double currentRoundTripTime; double availableOutgoingBitrate; double availableIncomingBitrate; unsigned long circuitBreakerTriggerCount; unsigned long long requestsReceived; unsigned long long requestsSent; unsigned long long responsesReceived; unsigned long long responsesSent; unsigned long long retransmissionsReceived; unsigned long long retransmissionsSent; unsigned long long consentRequestsSent; DOMHighResTimeStamp consentExpiredTimestamp; */ /* XXX: RTCIceCandidateStats DOMString transportId; boolean isRemote; RTCNetworkType networkType; DOMString ip; long port; DOMString protocol; RTCIceCandidateType candidateType; long priority; DOMString url; DOMString relayProtocol; boolean deleted = false; }; */ gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL); gst_structure_free (stats); return id; } /* https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats */ static gchar * _get_stats_from_dtls_transport (GstWebRTCBin * webrtc, GstWebRTCDTLSTransport * transport, GstStructure * s) { GstStructure *stats; gchar *id; double ts; gchar *ice_id; gst_structure_get_double (s, "timestamp", &ts); id = g_strdup_printf ("transport-stats_%s", GST_OBJECT_NAME (transport)); stats = gst_structure_new_empty (id); _set_base_stats (stats, GST_WEBRTC_STATS_TRANSPORT, ts, id); /* XXX: RTCTransportStats unsigned long packetsSent; unsigned long packetsReceived; unsigned long long bytesSent; unsigned long long bytesReceived; DOMString rtcpTransportStatsId; RTCIceRole iceRole; RTCDtlsTransportState dtlsState; DOMString selectedCandidatePairId; DOMString localCertificateId; DOMString remoteCertificateId; */ /* XXX: RTCCertificateStats DOMString fingerprint; DOMString fingerprintAlgorithm; DOMString base64Certificate; DOMString issuerCertificateId; */ /* XXX: RTCIceCandidateStats DOMString transportId; boolean isRemote; DOMString ip; long port; DOMString protocol; RTCIceCandidateType candidateType; long priority; DOMString url; boolean deleted = false; */ gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL); gst_structure_free (stats); ice_id = _get_stats_from_ice_transport (webrtc, transport->transport, s); g_free (ice_id); return id; } static void _get_stats_from_transport_channel (GstWebRTCBin * webrtc, TransportStream * stream, const gchar * codec_id, guint ssrc, GstStructure * s) { GstWebRTCDTLSTransport *transport; GObject *rtp_session; GstStructure *rtp_stats; GValueArray *source_stats; gchar *transport_id; double ts; int i; gst_structure_get_double (s, "timestamp", &ts); transport = stream->transport; if (!transport) return; g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session", stream->session_id, &rtp_session); g_object_get (rtp_session, "stats", &rtp_stats, NULL); gst_structure_get (rtp_stats, "source-stats", G_TYPE_VALUE_ARRAY, &source_stats, NULL); GST_DEBUG_OBJECT (webrtc, "retrieving rtp stream stats from transport %" GST_PTR_FORMAT " rtp session %" GST_PTR_FORMAT " with %u rtp sources, " "transport %" GST_PTR_FORMAT, stream, rtp_session, source_stats->n_values, transport); transport_id = _get_stats_from_dtls_transport (webrtc, transport, s); /* construct stats objects */ for (i = 0; i < source_stats->n_values; i++) { const GstStructure *stats; const GValue *val = g_value_array_get_nth (source_stats, i); guint stats_ssrc = 0; stats = gst_value_get_structure (val); /* skip foreign sources */ gst_structure_get (stats, "ssrc", G_TYPE_UINT, &stats_ssrc, NULL); if (ssrc && stats_ssrc && ssrc != stats_ssrc) continue; _get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id, transport_id, s); } g_object_unref (rtp_session); gst_structure_free (rtp_stats); g_value_array_free (source_stats); g_free (transport_id); } /* https://www.w3.org/TR/webrtc-stats/#codec-dict* */ static void _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s, gchar ** out_id, guint * out_ssrc) { GstStructure *stats; GstCaps *caps; gchar *id; double ts; guint ssrc = 0; gst_structure_get_double (s, "timestamp", &ts); stats = gst_structure_new_empty ("unused"); id = g_strdup_printf ("codec-stats-%s", GST_OBJECT_NAME (pad)); _set_base_stats (stats, GST_WEBRTC_STATS_CODEC, ts, id); caps = gst_pad_get_current_caps (pad); if (caps && gst_caps_is_fixed (caps)) { GstStructure *caps_s = gst_caps_get_structure (caps, 0); gint pt, clock_rate; if (gst_structure_get_int (caps_s, "payload", &pt)) gst_structure_set (stats, "payload-type", G_TYPE_UINT, pt, NULL); if (gst_structure_get_int (caps_s, "clock-rate", &clock_rate)) gst_structure_set (stats, "clock-rate", G_TYPE_UINT, clock_rate, NULL); if (gst_structure_get_uint (caps_s, "ssrc", &ssrc)) gst_structure_set (stats, "ssrc", G_TYPE_UINT, ssrc, NULL); /* FIXME: codecType, mimeType, channels, sdpFmtpLine, implementation, transportId */ } if (caps) gst_caps_unref (caps); gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL); gst_structure_free (stats); if (out_id) *out_id = id; else g_free (id); if (out_ssrc) *out_ssrc = ssrc; } static gboolean _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s) { GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad); TransportStream *stream; gchar *codec_id; guint ssrc; _get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc); if (!wpad->trans) goto out; stream = WEBRTC_TRANSCEIVER (wpad->trans)->stream; if (!stream) goto out; _get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc, s); out: g_free (codec_id); return TRUE; } void gst_webrtc_bin_update_stats (GstWebRTCBin * webrtc) { GstStructure *s = gst_structure_new_empty ("application/x-webrtc-stats"); double ts = monotonic_time_as_double_milliseconds (); GstStructure *pc_stats; _init_debug (); gst_structure_set (s, "timestamp", G_TYPE_DOUBLE, ts, NULL); /* FIXME: better unique IDs */ /* FIXME: rate limitting stat updates? */ /* FIXME: all stats need to be kept forever */ GST_DEBUG_OBJECT (webrtc, "updating stats at time %f", ts); if ((pc_stats = _get_peer_connection_stats (webrtc))) { const gchar *id = "peer-connection-stats"; _set_base_stats (pc_stats, GST_WEBRTC_STATS_PEER_CONNECTION, ts, id); gst_structure_set (s, id, GST_TYPE_STRUCTURE, pc_stats, NULL); gst_structure_free (pc_stats); } gst_element_foreach_pad (GST_ELEMENT (webrtc), (GstElementForeachPadFunc) _get_stats_from_pad, s); gst_structure_remove_field (s, "timestamp"); if (webrtc->priv->stats) gst_structure_free (webrtc->priv->stats); webrtc->priv->stats = s; }