mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d9d7814182
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>
760 lines
26 KiB
C
760 lines
26 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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/* for GValueArray... */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include "gstwebrtcstats.h"
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#include "gstwebrtcbin.h"
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#include "transportstream.h"
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#include "transportreceivebin.h"
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#include "utils.h"
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#include "webrtctransceiver.h"
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#define GST_CAT_DEFAULT gst_webrtc_stats_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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static void
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_init_debug (void)
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{
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static gsize _init = 0;
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if (g_once_init_enter (&_init)) {
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_stats_debug, "webrtcstats", 0,
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"webrtcstats");
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g_once_init_leave (&_init, 1);
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}
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}
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static double
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monotonic_time_as_double_milliseconds (void)
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{
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return g_get_monotonic_time () / 1000.0;
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}
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static void
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_set_base_stats (GstStructure * s, GstWebRTCStatsType type, double ts,
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const char *id)
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{
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gchar *name = _enum_value_to_string (GST_TYPE_WEBRTC_STATS_TYPE,
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type);
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g_return_if_fail (name != NULL);
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gst_structure_set_name (s, name);
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gst_structure_set (s, "type", GST_TYPE_WEBRTC_STATS_TYPE, type, "timestamp",
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G_TYPE_DOUBLE, ts, "id", G_TYPE_STRING, id, NULL);
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g_free (name);
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}
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static GstStructure *
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_get_peer_connection_stats (GstWebRTCBin * webrtc)
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{
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GstStructure *s = gst_structure_new_empty ("unused");
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/* FIXME: datachannel */
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gst_structure_set (s, "data-channels-opened", G_TYPE_UINT, 0,
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"data-channels-closed", G_TYPE_UINT, 0, "data-channels-requested",
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G_TYPE_UINT, 0, "data-channels-accepted", G_TYPE_UINT, 0, NULL);
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return s;
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}
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#define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate)
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#define FIXED_16_16_TO_DOUBLE(v) ((double) ((v & 0xffff0000) >> 16) + ((v & 0xffff) / 65536.0))
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#define FIXED_32_32_TO_DOUBLE(v) ((double) ((v & G_GUINT64_CONSTANT (0xffffffff00000000)) >> 32) + ((v & G_GUINT64_CONSTANT (0xffffffff)) / 4294967296.0))
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/* https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*
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https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict* */
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static void
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_get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
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const GstStructure * source_stats, const gchar * codec_id,
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const 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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guint64 packets, bytes;
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gboolean internal;
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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, "ssrc", G_TYPE_UINT, &ssrc, "clock-rate",
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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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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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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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NULL);
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if (gst_structure_get_int (source_stats, "packets-lost", &lost))
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gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
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if (gst_structure_get_uint (source_stats, "jitter", &jitter))
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gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
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CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
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/* 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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/* 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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/* RTCStreamStats */
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gst_structure_set (out, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (out, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (out, "transport-id", G_TYPE_STRING, transport_id, NULL);
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/* 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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gst_structure_set (out, "pli-count", G_TYPE_UINT, pli, NULL);
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if (gst_structure_get_uint (source_stats, "sent-nack-count", &nack))
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gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
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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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gst_structure_free (out);
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gst_structure_free (r_in);
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g_free (out_id);
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g_free (r_in_id);
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} else {
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GstStructure *in, *r_out;
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gchar *r_out_id, *in_id;
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gboolean have_sr = FALSE;
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gst_structure_get (source_stats, "have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
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in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
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r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
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in = gst_structure_new_empty (in_id);
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_set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
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/* 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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gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
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if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
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gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* RTCInboundRtpStreamStats:
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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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/* RTCStreamStats */
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gst_structure_set (r_out, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (r_out, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (r_out, "transport-id", G_TYPE_STRING, transport_id,
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NULL);
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/* XXX: mediaType, trackId */
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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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/* 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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if (gst_structure_get_uint64 (source_stats, "sr-ntptime", &ntptime)) {
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/* 16.16 fixed point to double */
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double val = FIXED_32_32_TO_DOUBLE (ntptime);
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gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, val, NULL);
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}
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} else {
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/* default values */
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gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, 0.0, NULL);
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}
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gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
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|
|
|
/* 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);
|
|
|
|
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, 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;
|
|
}
|