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gstreamer/ext/webrtc/gstwebrtcstats.c
Jan Alexander Steffens (heftig) dc0e95acab webrtcbin: Filter transport stream stats by ssrc 
Since the addition of BUNDLE support, the pads and the transceivers
share a single transport stream. When getting stats from the stream,
filter by the ssrc of the current pad to avoid merging the stats for
different pads.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/889
2019-03-12 01:40:59 +00:00

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/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* 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)
/* 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,
const GstStructure * source_stats, const gchar * codec_id,
const gchar * transport_id, GstStructure * s)
{
GstStructure *in, *out, *r_in, *r_out;
gchar *in_id, *out_id, *r_in_id, *r_out_id;
guint ssrc, fir, pli, nack, jitter;
int lost, clock_rate;
guint64 packets, bytes;
gboolean have_rb = FALSE, sent_rb = FALSE;
double ts;
gst_structure_get_double (s, "timestamp", &ts);
gst_structure_get_uint (source_stats, "ssrc", &ssrc);
gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
"sent_rb", G_TYPE_BOOLEAN, &sent_rb, "clock-rate", G_TYPE_INT,
&clock_rate, NULL);
in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
r_in_id = g_strdup_printf ("rtp-remote-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);
/* RTCStreamStats */
gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCReceivedRTPStreamStats */
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
if (gst_structure_get_uint64 (source_stats, "octets-received", &bytes))
gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_int (source_stats, "packets-lost", &lost))
gst_structure_set (in, "packets-lost", G_TYPE_INT, lost, NULL);
if (gst_structure_get_uint (source_stats, "jitter", &jitter))
gst_structure_set (in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
/*
RTCReceivedRTPStreamStats
double fractionLost;
unsigned long packetsDiscarded;
unsigned long packetsFailedDecryption;
unsigned long packetsRepaired;
unsigned long burstPacketsLost;
unsigned long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
*/
/* RTCInboundRTPStreamStats */
gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
/* XXX: framesDecoded, lastPacketReceivedTimestamp */
r_in = gst_structure_new_empty (r_in_id);
_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
/* RTCStreamStats */
gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCReceivedRTPStreamStats */
if (sent_rb) {
if (gst_structure_get_uint (source_stats, "sent-rb-jitter", &jitter))
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
if (gst_structure_get_int (source_stats, "sent-rb-packetslost", &lost))
gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
/* packetsReceived, bytesReceived */
} else {
/* default values */
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE, 0.0, "packets-lost",
G_TYPE_INT, 0, NULL);
}
/* XXX: RTCReceivedRTPStreamStats
double fractionLost;
unsigned long packetsDiscarded;
unsigned long packetsFailedDecryption;
unsigned long packetsRepaired;
unsigned long burstPacketsLost;
unsigned long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
*/
/* RTCRemoteInboundRTPStreamStats */
gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
if (have_rb) {
guint32 rtt;
if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
/* 16.16 fixed point to double */
double val =
(double) ((rtt & 0xffff0000) >> 16) + ((rtt & 0xffff) / 65536.0);
gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
}
} else {
/* default values */
gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
}
/* XXX: framesDecoded, lastPacketReceivedTimestamp */
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);
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 */
/* RTCSentRTPStreamStats */
if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
/* XXX:
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
*/
/* RTCOutboundRTPStreamStats */
gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
/* XXX:
DOMHighResTimeStamp lastPacketSentTimestamp;
double targetBitrate;
unsigned long framesEncoded;
double totalEncodeTime;
double averageRTCPInterval;
*/
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, sliCount, qpSum */
/* RTCSentRTPStreamStats */
/* if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT64, packets, NULL);*/
/* XXX:
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
*/
gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
gst_structure_set (s, in_id, GST_TYPE_STRUCTURE, in, NULL);
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_set (s, r_out_id, GST_TYPE_STRUCTURE, r_out, NULL);
gst_structure_free (in);
gst_structure_free (out);
gst_structure_free (r_in);
gst_structure_free (r_out);
g_free (in_id);
g_free (out_id);
g_free (r_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;
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);
_get_stats_from_ice_transport (webrtc, transport->transport, s);
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)
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);
gboolean internal;
guint stats_ssrc = 0;
stats = gst_value_get_structure (val);
/* skip internal or foreign sources */
gst_structure_get (stats,
"internal", G_TYPE_BOOLEAN, &internal,
"ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
if (internal || (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;
}
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