mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
549 lines
19 KiB
C
549 lines
19 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, "webrtcice", 0,
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"webrtcice");
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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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/* 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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GstStructure *in, *out, *r_in, *r_out;
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gchar *in_id, *out_id, *r_in_id, *r_out_id;
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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 have_rb = FALSE, sent_rb = FALSE;
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double ts;
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gst_structure_get_double (s, "timestamp", &ts);
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gst_structure_get_uint (source_stats, "ssrc", &ssrc);
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gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
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"sent_rb", G_TYPE_BOOLEAN, &sent_rb, "clock-rate", G_TYPE_INT,
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&clock_rate, NULL);
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in_id = g_strdup_printf ("rtp-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_id = g_strdup_printf ("rtp-remote-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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/* RTCStreamStats */
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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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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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/* RTCReceivedRTPStreamStats */
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if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
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gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
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if (gst_structure_get_uint64 (source_stats, "octets-received", &bytes))
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gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
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if (gst_structure_get_int (source_stats, "packets-lost", &lost))
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gst_structure_set (in, "packets-lost", G_TYPE_INT, lost, NULL);
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if (gst_structure_get_uint (source_stats, "jitter", &jitter))
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gst_structure_set (in, "jitter", G_TYPE_DOUBLE,
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CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
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/*
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RTCReceivedRTPStreamStats
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double fractionLost;
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unsigned long packetsDiscarded;
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unsigned long packetsFailedDecryption;
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unsigned long packetsRepaired;
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unsigned long burstPacketsLost;
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unsigned long burstPacketsDiscarded;
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unsigned long burstLossCount;
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unsigned long burstDiscardCount;
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double burstLossRate;
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double burstDiscardRate;
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double gapLossRate;
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double gapDiscardRate;
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*/
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/* RTCInboundRTPStreamStats */
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gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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r_in = gst_structure_new_empty (r_in_id);
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_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
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/* RTCStreamStats */
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gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
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gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
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gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* RTCReceivedRTPStreamStats */
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if (sent_rb) {
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if (gst_structure_get_uint (source_stats, "sent-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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if (gst_structure_get_int (source_stats, "sent-rb-packetslost", &lost))
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gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
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/* packetsReceived, bytesReceived */
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} else {
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/* default values */
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gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE, 0.0, "packets-lost",
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G_TYPE_INT, 0, NULL);
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}
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/* XXX: RTCReceivedRTPStreamStats
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double fractionLost;
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unsigned long packetsDiscarded;
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unsigned long packetsFailedDecryption;
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unsigned long packetsRepaired;
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unsigned long burstPacketsLost;
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unsigned long burstPacketsDiscarded;
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unsigned long burstLossCount;
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unsigned long burstDiscardCount;
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double burstLossRate;
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double burstDiscardRate;
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double gapLossRate;
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double gapDiscardRate;
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*/
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/* RTCRemoteInboundRTPStreamStats */
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gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
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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 =
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(double) ((rtt & 0xffff0000) >> 16) + ((rtt & 0xffff) / 65536.0);
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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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} else {
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/* default values */
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gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
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}
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/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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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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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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/* RTCSentRTPStreamStats */
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if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
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gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
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if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
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gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
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/* XXX:
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unsigned long packetsDiscardedOnSend;
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unsigned long long bytesDiscardedOnSend;
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*/
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/* RTCOutboundRTPStreamStats */
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gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
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/* XXX:
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DOMHighResTimeStamp lastPacketSentTimestamp;
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double targetBitrate;
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unsigned long framesEncoded;
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double totalEncodeTime;
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double averageRTCPInterval;
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*/
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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, NULL);
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/* XXX: mediaType, trackId, sliCount, qpSum */
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/* RTCSentRTPStreamStats */
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/* if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
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gst_structure_set (r_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 (r_out, "packets-sent", G_TYPE_UINT64, packets, NULL);*/
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/* XXX:
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unsigned long packetsDiscardedOnSend;
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unsigned long long bytesDiscardedOnSend;
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*/
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gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
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gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
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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_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);
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gst_structure_free (in);
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gst_structure_free (out);
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gst_structure_free (r_in);
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gst_structure_free (r_out);
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g_free (in_id);
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g_free (out_id);
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g_free (r_in_id);
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g_free (r_out_id);
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}
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/* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* */
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static gchar *
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_get_stats_from_ice_transport (GstWebRTCBin * webrtc,
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GstWebRTCICETransport * transport, GstStructure * s)
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{
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GstStructure *stats;
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gchar *id;
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double ts;
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gst_structure_get_double (s, "timestamp", &ts);
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id = g_strdup_printf ("ice-candidate-pair_%s", GST_OBJECT_NAME (transport));
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stats = gst_structure_new_empty (id);
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_set_base_stats (stats, GST_WEBRTC_STATS_TRANSPORT, ts, id);
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/* XXX: RTCIceCandidatePairStats
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DOMString transportId;
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DOMString localCandidateId;
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DOMString remoteCandidateId;
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RTCStatsIceCandidatePairState state;
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unsigned long long priority;
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boolean nominated;
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unsigned long packetsSent;
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unsigned long packetsReceived;
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unsigned long long bytesSent;
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unsigned long long bytesReceived;
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DOMHighResTimeStamp lastPacketSentTimestamp;
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DOMHighResTimeStamp lastPacketReceivedTimestamp;
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DOMHighResTimeStamp firstRequestTimestamp;
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DOMHighResTimeStamp lastRequestTimestamp;
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DOMHighResTimeStamp lastResponseTimestamp;
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double totalRoundTripTime;
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double currentRoundTripTime;
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double availableOutgoingBitrate;
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double availableIncomingBitrate;
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unsigned long circuitBreakerTriggerCount;
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unsigned long long requestsReceived;
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unsigned long long requestsSent;
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unsigned long long responsesReceived;
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unsigned long long responsesSent;
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unsigned long long retransmissionsReceived;
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unsigned long long retransmissionsSent;
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unsigned long long consentRequestsSent;
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DOMHighResTimeStamp consentExpiredTimestamp;
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*/
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/* XXX: RTCIceCandidateStats
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DOMString transportId;
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boolean isRemote;
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RTCNetworkType networkType;
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DOMString ip;
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long port;
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DOMString protocol;
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RTCIceCandidateType candidateType;
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long priority;
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DOMString url;
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DOMString relayProtocol;
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boolean deleted = false;
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};
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*/
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gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
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gst_structure_free (stats);
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return id;
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}
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/* https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats */
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static gchar *
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_get_stats_from_dtls_transport (GstWebRTCBin * webrtc,
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GstWebRTCDTLSTransport * transport, GstStructure * s)
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{
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GstStructure *stats;
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gchar *id;
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double ts;
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gst_structure_get_double (s, "timestamp", &ts);
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id = g_strdup_printf ("transport-stats_%s", GST_OBJECT_NAME (transport));
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stats = gst_structure_new_empty (id);
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_set_base_stats (stats, GST_WEBRTC_STATS_TRANSPORT, ts, id);
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/* XXX: RTCTransportStats
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unsigned long packetsSent;
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unsigned long packetsReceived;
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unsigned long long bytesSent;
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unsigned long long bytesReceived;
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DOMString rtcpTransportStatsId;
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RTCIceRole iceRole;
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RTCDtlsTransportState dtlsState;
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DOMString selectedCandidatePairId;
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DOMString localCertificateId;
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DOMString remoteCertificateId;
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*/
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/* XXX: RTCCertificateStats
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DOMString fingerprint;
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DOMString fingerprintAlgorithm;
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DOMString base64Certificate;
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DOMString issuerCertificateId;
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*/
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/* XXX: RTCIceCandidateStats
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DOMString transportId;
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boolean isRemote;
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DOMString ip;
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long port;
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DOMString protocol;
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RTCIceCandidateType candidateType;
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long priority;
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DOMString url;
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boolean deleted = false;
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*/
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gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
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gst_structure_free (stats);
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_get_stats_from_ice_transport (webrtc, transport->transport, s);
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return id;
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}
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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, GstStructure * s)
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{
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GstWebRTCDTLSTransport *transport;
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GObject *rtp_session;
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GstStructure *rtp_stats;
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GValueArray *source_stats;
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gchar *transport_id;
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double ts;
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int i;
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gst_structure_get_double (s, "timestamp", &ts);
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transport = stream->transport;
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if (!transport)
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transport = stream->transport;
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if (!transport)
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return;
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g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
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stream->session_id, &rtp_session);
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g_object_get (rtp_session, "stats", &rtp_stats, NULL);
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gst_structure_get (rtp_stats, "source-stats", G_TYPE_VALUE_ARRAY,
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&source_stats, NULL);
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GST_DEBUG_OBJECT (webrtc, "retrieving rtp stream stats from transport %"
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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);
|
|
gboolean internal;
|
|
|
|
stats = gst_value_get_structure (val);
|
|
|
|
/* skip internal sources */
|
|
gst_structure_get (stats, "internal", G_TYPE_BOOLEAN, &internal, NULL);
|
|
if (internal)
|
|
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 gchar *
|
|
_get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
|
|
GstStructure * s)
|
|
{
|
|
GstStructure *stats;
|
|
GstCaps *caps;
|
|
gchar *id;
|
|
double ts;
|
|
|
|
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);
|
|
|
|
/* 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);
|
|
|
|
return id;
|
|
}
|
|
|
|
static gboolean
|
|
_get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
|
|
{
|
|
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
|
|
gchar *codec_id;
|
|
|
|
codec_id = _get_codec_stats_from_pad (webrtc, pad, s);
|
|
if (wpad->trans) {
|
|
WebRTCTransceiver *trans;
|
|
trans = WEBRTC_TRANSCEIVER (wpad->trans);
|
|
if (trans->stream)
|
|
_get_stats_from_transport_channel (webrtc, trans->stream, codec_id, s);
|
|
}
|
|
|
|
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;
|
|
}
|