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gstreamer/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
Jan Schmidt 6b94f22bd6 webrtcbin: Retrieve RR stats from internal sources 
Check and generate remote reception statistics from the info stored on
internal sources, as they are stored there  when running against newer rtpbin
since MR !7424

This fixes cases where statistics are incomplete when
peers send RR reports from a single remote ssrc, which GStreamer does
when bundling is enabled and other RTP stacks may too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7425>
2024-10-12 04:12:22 +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"
#include <stdlib.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)
{
const 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);
}
static GstStructure *
_get_peer_connection_stats (GstWebRTCBin * webrtc)
{
guint opened;
guint closed;
GstStructure *s = gst_structure_new_empty ("peer-connection-stats");
gst_webrtc_bin_get_peer_connection_stats (webrtc, &opened, &closed);
gst_structure_set (s, "data-channels-opened", G_TYPE_UINT, opened,
"data-channels-closed", G_TYPE_UINT, closed, "data-channels-requested",
G_TYPE_UINT, 0, "data-channels-accepted", G_TYPE_UINT, 0, NULL);
return s;
}
static void
_gst_structure_take_structure (GstStructure * s, const char *fieldname,
GstStructure ** value_s)
{
GValue v = G_VALUE_INIT;
g_return_if_fail (GST_IS_STRUCTURE (*value_s));
g_value_init (&v, GST_TYPE_STRUCTURE);
g_value_take_boxed (&v, *value_s);
gst_structure_take_value (s, fieldname, &v);
*value_s = NULL;
}
#define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate)
#define FIXED_16_16_TO_DOUBLE(v) ((double) ((v & 0xffff0000) >> 16) + ((v & 0xffff) / 65536.0))
#define FIXED_32_32_TO_DOUBLE(v) ((double) ((v & G_GUINT64_CONSTANT (0xffffffff00000000)) >> 32) + ((v & G_GUINT64_CONSTANT (0xffffffff)) / 4294967296.0))
/* https://www.w3.org/TR/webrtc-stats/#remoteinboundrtpstats-dict* */
static gboolean
_get_stats_from_remote_rtp_source_stats (TransportStream * stream,
const GstStructure * source_stats, guint ssrc, guint clock_rate,
const gchar * codec_id, const gchar * kind, const gchar * transport_id,
GstStructure * s)
{
gboolean have_rb = FALSE;
int lost;
GstStructure *r_in;
gchar *r_in_id, *out_id;
guint32 rtt;
guint fraction_lost, jitter;
double ts;
gst_structure_get_double (s, "timestamp", &ts);
gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
/* This isn't what we're looking for */
if (have_rb == FALSE)
return FALSE;
r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
r_in = gst_structure_new_empty (r_in_id);
_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
/* RTCRtpStreamStats */
gst_structure_set (r_in, "local-id", G_TYPE_STRING, out_id, NULL);
gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (kind)
gst_structure_set (r_in, "kind", G_TYPE_STRING, kind, NULL);
/* RTCReceivedRtpStreamStats */
if (gst_structure_get_int (source_stats, "rb-packetslost", &lost))
gst_structure_set (r_in, "packets-lost", G_TYPE_INT64, (gint64) lost, NULL);
if (clock_rate && gst_structure_get_uint (source_stats, "rb-jitter", &jitter))
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
/* RTCReceivedRtpStreamStats:
unsigned long long packetsReceived;
unsigned long packetsDiscarded;
unsigned long packetsRepaired;
unsigned long burstPacketsLost;
unsigned long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
Can't be implemented frame re-assembly happens after rtpbin:
unsigned long framesDropped;
unsigned long partialFramesLost;
unsigned long fullFramesLost;
*/
/* RTCRemoteInboundRTPStreamStats */
if (gst_structure_get_uint (source_stats, "rb-fractionlost", &fraction_lost))
gst_structure_set (r_in, "fraction-lost", G_TYPE_DOUBLE,
(double) fraction_lost / 256.0, NULL);
if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
/* 16.16 fixed point to double */
double val = FIXED_16_16_TO_DOUBLE (rtt);
gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
}
/* RTCRemoteInboundRTPStreamStats:
To be added:
DOMString localId;
double totalRoundTripTime;
unsigned long long reportsReceived;
unsigned long long roundTripTimeMeasurements;
*/
gst_structure_set (r_in, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
source_stats, NULL);
_gst_structure_take_structure (s, r_in_id, &r_in);
g_free (r_in_id);
g_free (out_id);
return TRUE;
}
/* 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 (TransportStream * stream,
const GstStructure * source_stats, const gchar * codec_id,
const gchar * kind, const gchar * transport_id, GstStructure * s)
{
guint ssrc, fir, pli, nack, jitter;
int clock_rate;
guint64 packets, bytes;
gboolean internal;
double ts;
gst_structure_get_double (s, "timestamp", &ts);
gst_structure_get (source_stats, "ssrc", G_TYPE_UINT, &ssrc, "clock-rate",
G_TYPE_INT, &clock_rate, "internal", G_TYPE_BOOLEAN, &internal, NULL);
if (internal) {
GstStructure *out;
gchar *out_id, *r_in_id;
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
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 (kind)
gst_structure_set (out, "kind", G_TYPE_STRING, kind, NULL);
/* RTCSentRtpStreamStats */
if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
/* RTCOutboundRTPStreamStats */
if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
gst_structure_set (out, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
gst_structure_set (out, "pli-count", G_TYPE_UINT, pli, NULL);
if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
if (gst_structure_has_field (s, r_in_id))
gst_structure_set (out, "remote-id", G_TYPE_STRING, r_in_id, NULL);
g_free (r_in_id);
/* RTCOutboundRTPStreamStats:
To be added:
unsigned long sliCount;
unsigned long rtxSsrc;
DOMString mediaSourceId;
DOMString senderId;
DOMString remoteId;
DOMString rid;
DOMHighResTimeStamp lastPacketSentTimestamp;
unsigned long long headerBytesSent;
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
unsigned long fecPacketsSent;
unsigned long long retransmittedPacketsSent;
unsigned long long retransmittedBytesSent;
double averageRtcpInterval;
record<USVString, unsigned long long> perDscpPacketsSent;
Not relevant because webrtcbin doesn't encode:
double targetBitrate;
unsigned long long totalEncodedBytesTarget;
unsigned long frameWidth;
unsigned long frameHeight;
unsigned long frameBitDepth;
double framesPerSecond;
unsigned long framesSent;
unsigned long hugeFramesSent;
unsigned long framesEncoded;
unsigned long keyFramesEncoded;
unsigned long framesDiscardedOnSend;
unsigned long long qpSum;
unsigned long long totalSamplesSent;
unsigned long long samplesEncodedWithSilk;
unsigned long long samplesEncodedWithCelt;
boolean voiceActivityFlag;
double totalEncodeTime;
double totalPacketSendDelay;
RTCQualityLimitationReason qualityLimitationReason;
record<DOMString, double> qualityLimitationDurations;
unsigned long qualityLimitationResolutionChanges;
DOMString encoderImplementation;
*/
/* Store the raw stats from GStreamer into the structure for advanced
* information.
*/
gst_structure_set (out, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
source_stats, NULL);
_gst_structure_take_structure (s, out_id, &out);
g_free (out_id);
} else {
GstStructure *in, *r_out;
gchar *r_out_id, *in_id;
gboolean have_sr = FALSE;
GstStructure *jb_stats = NULL;
guint i;
guint64 jb_lost, duplicates, late, rtx_success;
gst_structure_get (source_stats, "have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
for (i = 0; i < stream->ssrcmap->len; i++) {
SsrcMapItem *item = g_ptr_array_index (stream->ssrcmap, i);
if (item->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
&& item->ssrc == ssrc) {
GObject *jb = g_weak_ref_get (&item->rtpjitterbuffer);
if (jb) {
g_object_get (jb, "stats", &jb_stats, NULL);
g_object_unref (jb);
}
break;
}
}
in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
in = gst_structure_new_empty (in_id);
_set_base_stats (in, GST_WEBRTC_STATS_INBOUND_RTP, ts, in_id);
/* RTCRtpStreamStats */
gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
if (kind)
gst_structure_set (in, "kind", G_TYPE_STRING, kind, NULL);
/* RTCReceivedRtpStreamStats */
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets)) {
gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
} else {
gst_structure_set (in, "packets-received", G_TYPE_UINT64,
G_GUINT64_CONSTANT (0), NULL);
}
if (jb_stats) {
gst_structure_get (jb_stats, "num-lost", G_TYPE_UINT64, &jb_lost,
"num-duplicates", G_TYPE_UINT64, &duplicates, "num-late",
G_TYPE_UINT64, &late, "rtx-success-count", G_TYPE_UINT64,
&rtx_success, NULL);
gint64 packets_lost = jb_lost > G_MAXINT64 ?
G_MAXINT64 : (gint64) jb_lost;
gst_structure_set (in, "packets-lost", G_TYPE_INT64, packets_lost, NULL);
gst_structure_set (in, "packets-discarded", G_TYPE_UINT64, late,
"packets-repaired", G_TYPE_UINT64, rtx_success, NULL);
} else {
gst_structure_set (in, "packets-lost", G_TYPE_INT64,
G_GINT64_CONSTANT (0), NULL);
gst_structure_set (in, "packets-discarded", G_TYPE_UINT64,
G_GUINT64_CONSTANT (0), "packets-repaired", G_TYPE_UINT64,
G_GUINT64_CONSTANT (0), 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);
} else {
gst_structure_set (in, "jitter", G_TYPE_DOUBLE, 0.0, NULL);
}
/*
RTCReceivedRtpStreamStats
To be added:
unsigned long long burstPacketsLost;
unsigned long long burstPacketsDiscarded;
unsigned long burstLossCount;
unsigned long burstDiscardCount;
double burstLossRate;
double burstDiscardRate;
double gapLossRate;
double gapDiscardRate;
Not relevant because webrtcbin doesn't decode:
unsigned long framesDropped;
unsigned long partialFramesLost;
unsigned long fullFramesLost;
*/
/* RTCInboundRtpStreamStats */
gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
if (gst_structure_get_uint64 (source_stats, "octets-received", &bytes))
gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint (source_stats, "sent-fir-count", &fir))
gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
if (gst_structure_get_uint (source_stats, "sent-pli-count", &pli))
gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
if (gst_structure_get_uint (source_stats, "sent-nack-count", &nack))
gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
if (jb_stats)
gst_structure_set (in, "packets-duplicated", G_TYPE_UINT64, duplicates,
NULL);
/* RTCInboundRtpStreamStats:
To be added:
required DOMString receiverId;
double averageRtcpInterval;
unsigned long long headerBytesReceived;
unsigned long long fecPacketsReceived;
unsigned long long fecPacketsDiscarded;
unsigned long long bytesReceived;
unsigned long long packetsFailedDecryption;
record<USVString, unsigned long long> perDscpPacketsReceived;
unsigned long nackCount;
unsigned long firCount;
unsigned long pliCount;
unsigned long sliCount;
double jitterBufferDelay;
Not relevant because webrtcbin doesn't decode or depayload:
unsigned long framesDecoded;
unsigned long keyFramesDecoded;
unsigned long frameWidth;
unsigned long frameHeight;
unsigned long frameBitDepth;
double framesPerSecond;
unsigned long long qpSum;
double totalDecodeTime;
double totalInterFrameDelay;
double totalSquaredInterFrameDelay;
boolean voiceActivityFlag;
DOMHighResTimeStamp lastPacketReceivedTimestamp;
double totalProcessingDelay;
DOMHighResTimeStamp estimatedPlayoutTimestamp;
unsigned long long jitterBufferEmittedCount;
unsigned long long totalSamplesReceived;
unsigned long long totalSamplesDecoded;
unsigned long long samplesDecodedWithSilk;
unsigned long long samplesDecodedWithCelt;
unsigned long long concealedSamples;
unsigned long long silentConcealedSamples;
unsigned long long concealmentEvents;
unsigned long long insertedSamplesForDeceleration;
unsigned long long removedSamplesForAcceleration;
double audioLevel;
double totalAudioEnergy;
double totalSamplesDuration;
unsigned long framesReceived;
DOMString decoderImplementation;
*/
r_out = gst_structure_new_empty (r_out_id);
_set_base_stats (r_out, GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP, ts, r_out_id);
/* RTCStreamStats */
gst_structure_set (r_out, "ssrc", G_TYPE_UINT, ssrc, NULL);
gst_structure_set (r_out, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_out, "transport-id", G_TYPE_STRING, transport_id,
NULL);
if (kind)
gst_structure_set (r_out, "kind", G_TYPE_STRING, kind, NULL);
/* XXX: mediaType, trackId */
/* RTCSentRtpStreamStats */
guint sr_bytes = 0, sr_packets = 0;
if (have_sr) {
gst_structure_get_uint (source_stats, "sr-octet-count", &sr_bytes);
gst_structure_get_uint (source_stats, "sr-packet-count", &sr_packets);
}
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT, sr_bytes, NULL);
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT, sr_packets, NULL);
/* RTCSentRtpStreamStats:
To be added:
unsigned long rtxSsrc;
DOMString mediaSourceId;
DOMString senderId;
DOMString remoteId;
DOMString rid;
DOMHighResTimeStamp lastPacketSentTimestamp;
unsigned long long headerBytesSent;
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
unsigned long fecPacketsSent;
unsigned long long retransmittedPacketsSent;
unsigned long long retransmittedBytesSent;
double averageRtcpInterval;
unsigned long sliCount;
Can't be implemented because we don't decode:
double targetBitrate;
unsigned long long totalEncodedBytesTarget;
unsigned long frameWidth;
unsigned long frameHeight;
unsigned long frameBitDepth;
double framesPerSecond;
unsigned long framesSent;
unsigned long hugeFramesSent;
unsigned long framesEncoded;
unsigned long keyFramesEncoded;
unsigned long framesDiscardedOnSend;
unsigned long long qpSum;
unsigned long long totalSamplesSent;
unsigned long long samplesEncodedWithSilk;
unsigned long long samplesEncodedWithCelt;
boolean voiceActivityFlag;
double totalEncodeTime;
double totalPacketSendDelay;
RTCQualityLimitationReason qualityLimitationReason;
record<DOMString, double> qualityLimitationDurations;
unsigned long qualityLimitationResolutionChanges;
record<USVString, unsigned long long> perDscpPacketsSent;
DOMString encoderImplementation;
*/
/* RTCRemoteOutboundRtpStreamStats */
if (have_sr) {
guint64 ntptime;
if (gst_structure_get_uint64 (source_stats, "sr-ntptime", &ntptime)) {
/* 16.16 fixed point to double */
double val = FIXED_32_32_TO_DOUBLE (ntptime);
gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, val, NULL);
}
} else {
/* default values */
gst_structure_set (r_out, "remote-timestamp", G_TYPE_DOUBLE, 0.0, NULL);
}
gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
/* To be added:
reportsSent
*/
/* Store the raw stats from GStreamer into the structure for advanced
* information.
*/
if (jb_stats)
_gst_structure_take_structure (in, "gst-rtpjitterbuffer-stats",
&jb_stats);
gst_structure_set (in, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
source_stats, NULL);
_gst_structure_take_structure (s, in_id, &in);
_gst_structure_take_structure (s, r_out_id, &r_out);
g_free (in_id);
g_free (r_out_id);
}
}
/* https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* */
static gchar *
_get_stats_from_ice_candidates (GstWebRTCBin * webrtc,
GstWebRTCICECandidateStats * can, const gchar * transport_id,
const gchar * candidate_tag, GstStructure * s)
{
GstStructure *stats;
GstWebRTCStatsType type;
gchar *id;
double ts;
gst_structure_get_double (s, "timestamp", &ts);
id = g_strdup_printf ("ice-candidate-%s_%u_%s_%u", candidate_tag,
can->stream_id, can->ipaddr, can->port);
stats = gst_structure_new_empty (id);
if (g_str_equal (candidate_tag, "local")) {
type = GST_WEBRTC_STATS_LOCAL_CANDIDATE;
} else if (g_str_equal (candidate_tag, "remote")) {
type = GST_WEBRTC_STATS_REMOTE_CANDIDATE;
} else {
GST_WARNING_OBJECT (webrtc, "Invalid ice candidate tag: %s", candidate_tag);
return NULL;
}
_set_base_stats (stats, type, ts, id);
/* RTCIceCandidateStats
DOMString transportId;
DOMString address;
long port;
DOMString protocol;
RTCIceCandidateType candidateType;
long priority;
DOMString url;
DOMString relayProtocol;
*/
if (transport_id)
gst_structure_set (stats, "transport-id", G_TYPE_STRING, transport_id,
NULL);
gst_structure_set (stats, "address", G_TYPE_STRING, can->ipaddr, NULL);
gst_structure_set (stats, "port", G_TYPE_UINT, can->port, NULL);
gst_structure_set (stats, "candidate-type", G_TYPE_STRING, can->type, NULL);
gst_structure_set (stats, "priority", G_TYPE_UINT, can->prio, NULL);
gst_structure_set (stats, "protocol", G_TYPE_STRING, can->proto, NULL);
if (can->relay_proto)
gst_structure_set (stats, "relay-protocol", G_TYPE_STRING, can->relay_proto,
NULL);
if (can->url)
gst_structure_set (stats, "url", G_TYPE_STRING, can->url, NULL);
gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
gst_structure_free (stats);
return id;
}
/* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* */
static gchar *
_get_stats_from_ice_transport (GstWebRTCBin * webrtc,
GstWebRTCICETransport * transport, GstWebRTCICEStream * stream,
const GstStructure * twcc_stats, const gchar * transport_id,
GstStructure * s)
{
GstStructure *stats;
gchar *id;
gchar *local_cand_id = NULL, *remote_cand_id = NULL;
double ts;
GstWebRTCICECandidateStats *local_cand = NULL, *remote_cand = NULL;
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_CANDIDATE_PAIR, ts, id);
/* RTCIceCandidatePairStats
DOMString transportId;
DOMString localCandidateId;
DOMString remoteCandidateId;
XXX: To be added:
RTCStatsIceCandidatePairState state;
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;
unsigned long packetsDiscardedOnSend;
unsigned long long bytesDiscardedOnSend;
unsigned long long requestBytesSent;
unsigned long long consentRequestBytesSent;
unsigned long long responseBytesSent;
*/
if (gst_webrtc_ice_get_selected_pair (webrtc->priv->ice, stream,
&local_cand, &remote_cand)) {
local_cand_id =
_get_stats_from_ice_candidates (webrtc, local_cand, transport_id,
"local", s);
remote_cand_id =
_get_stats_from_ice_candidates (webrtc, remote_cand, transport_id,
"remote", s);
gst_structure_set (stats, "local-candidate-id", G_TYPE_STRING,
local_cand_id, NULL);
gst_structure_set (stats, "remote-candidate-id", G_TYPE_STRING,
remote_cand_id, NULL);
} else
GST_INFO_OBJECT (webrtc,
"No selected ICE candidate pair was found for transport %s",
GST_OBJECT_NAME (transport));
/* XXX: these stats are at the rtp session level but there isn't a specific
* stats structure for that. The RTCIceCandidatePairStats is the closest with
* the 'availableIncomingBitrate' and 'availableOutgoingBitrate' fields
*/
if (twcc_stats)
gst_structure_set (stats, "gst-twcc-stats", GST_TYPE_STRUCTURE, twcc_stats,
NULL);
gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
g_free (local_cand_id);
g_free (remote_cand_id);
gst_webrtc_ice_candidate_stats_free (local_cand);
gst_webrtc_ice_candidate_stats_free (remote_cand);
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, GstWebRTCICEStream * stream,
const GstStructure * twcc_stats, 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;
*/
ice_id =
_get_stats_from_ice_transport (webrtc, transport->transport, stream,
twcc_stats, id, s);
if (ice_id) {
gst_structure_set (stats, "selected-candidate-pair-id", G_TYPE_STRING,
ice_id, NULL);
g_free (ice_id);
}
gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
gst_structure_free (stats);
return id;
}
/* https://www.w3.org/TR/webrtc-stats/#codec-dict* */
static gboolean
_get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
GstStructure * s, gchar ** out_id, guint * out_ssrc, guint * out_clock_rate)
{
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
GstStructure *stats;
GstCaps *caps = NULL;
gchar *id;
double ts;
guint ssrc = 0;
gint clock_rate = 0;
gboolean has_caps_ssrc = FALSE;
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);
if (wpad->received_caps)
caps = gst_caps_ref (wpad->received_caps);
else
caps = gst_pad_get_current_caps (pad);
GST_DEBUG_OBJECT (pad, "Pad caps are: %" GST_PTR_FORMAT, caps);
if (caps && gst_caps_is_fixed (caps)) {
GstStructure *caps_s = gst_caps_get_structure (caps, 0);
gint pt;
const gchar *encoding_name, *media, *encoding_params;
GstSDPMedia sdp_media = { 0 };
guint channels = 0;
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);
has_caps_ssrc = TRUE;
}
media = gst_structure_get_string (caps_s, "media");
encoding_name = gst_structure_get_string (caps_s, "encoding-name");
encoding_params = gst_structure_get_string (caps_s, "encoding-params");
if (media || encoding_name) {
gchar *mime_type;
mime_type = g_strdup_printf ("%s/%s", media ? media : "",
encoding_name ? encoding_name : "");
gst_structure_set (stats, "mime-type", G_TYPE_STRING, mime_type, NULL);
g_free (mime_type);
}
if (encoding_params)
channels = atoi (encoding_params);
if (channels)
gst_structure_set (stats, "channels", G_TYPE_UINT, channels, NULL);
if (gst_pad_get_direction (pad) == GST_PAD_SRC)
gst_structure_set (stats, "codec-type", G_TYPE_STRING, "decode", NULL);
else
gst_structure_set (stats, "codec-type", G_TYPE_STRING, "encode", NULL);
gst_sdp_media_init (&sdp_media);
if (gst_sdp_media_set_media_from_caps (caps, &sdp_media) == GST_SDP_OK) {
const gchar *fmtp = gst_sdp_media_get_attribute_val (&sdp_media, "fmtp");
if (fmtp) {
gst_structure_set (stats, "sdp-fmtp-line", G_TYPE_STRING, fmtp, NULL);
}
}
gst_sdp_media_uninit (&sdp_media);
/* FIXME: 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;
if (out_clock_rate)
*out_clock_rate = clock_rate;
return has_caps_ssrc;
}
struct transport_stream_stats
{
TransportStream *stream;
char *transport_id;
char *codec_id;
guint ssrc;
const char *kind;
guint clock_rate;
GValueArray *source_stats;
GstStructure *s; /* Return value stats accumulator */
};
static gboolean
webrtc_stats_get_from_transport_for_one_ssrc (SsrcMapItem * entry,
struct transport_stream_stats *ts_stats)
{
double ts;
int i;
/* We're only interested in the map entry for the ssrc for the
* pad under inspection */
if (ts_stats->ssrc != entry->ssrc)
return FALSE; /* Continue iterating */
gst_structure_get_double (ts_stats->s, "timestamp", &ts);
/* construct stats objects */
for (i = 0; i < ts_stats->source_stats->n_values; i++) {
const GValue *val = g_value_array_get_nth (ts_stats->source_stats, i);
const GstStructure *stats = gst_value_get_structure (val);
guint stats_ssrc = 0;
if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
entry->ssrc == stats_ssrc) {
GST_TRACE ("Found source stats for ssrc %u: %" GST_PTR_FORMAT, stats_ssrc,
stats);
_get_stats_from_rtp_source_stats (ts_stats->stream, stats,
ts_stats->codec_id, ts_stats->kind, ts_stats->transport_id,
ts_stats->s);
}
if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc)
&& entry->ssrc == stats_ssrc) {
GST_TRACE ("Found remote source stats for ssrc %u: %" GST_PTR_FORMAT,
stats_ssrc, stats);
_get_stats_from_remote_rtp_source_stats (ts_stats->stream, stats,
entry->ssrc, ts_stats->clock_rate, ts_stats->codec_id, ts_stats->kind,
ts_stats->transport_id, ts_stats->s);
}
}
/* we want to look at all the entries */
return FALSE;
}
static gboolean
_get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
{
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
struct transport_stream_stats ts_stats = { NULL, };
guint clock_rate;
GObject *rtp_session;
GObject *gst_rtp_session;
GstStructure *rtp_stats, *twcc_stats;
GstWebRTCKind kind;
_get_codec_stats_from_pad (webrtc, pad, s, &ts_stats.codec_id,
&ts_stats.ssrc, &clock_rate);
if (!wpad->trans)
goto out;
g_object_get (wpad->trans, "kind", &kind, NULL);
switch (kind) {
case GST_WEBRTC_KIND_AUDIO:
ts_stats.kind = "audio";
break;
case GST_WEBRTC_KIND_VIDEO:
ts_stats.kind = "video";
break;
case GST_WEBRTC_KIND_UNKNOWN:
ts_stats.kind = NULL;
break;
};
ts_stats.stream = WEBRTC_TRANSCEIVER (wpad->trans)->stream;
if (!ts_stats.stream)
goto out;
if (wpad->trans->mline == G_MAXUINT)
goto out;
if (!ts_stats.stream->transport)
goto out;
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
ts_stats.stream->session_id, &rtp_session);
g_object_get (rtp_session, "stats", &rtp_stats, NULL);
g_signal_emit_by_name (webrtc->rtpbin, "get-session",
ts_stats.stream->session_id, &gst_rtp_session);
g_object_get (gst_rtp_session, "twcc-stats", &twcc_stats, NULL);
gst_structure_get (rtp_stats, "source-stats", G_TYPE_VALUE_ARRAY,
&ts_stats.source_stats, NULL);
ts_stats.transport_id =
_get_stats_from_dtls_transport (webrtc, ts_stats.stream->transport,
GST_WEBRTC_ICE_STREAM (ts_stats.stream->stream), twcc_stats, s);
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, ts_stats.stream, rtp_session,
ts_stats.source_stats->n_values, ts_stats.stream->transport);
ts_stats.s = s;
ts_stats.clock_rate = clock_rate;
transport_stream_find_ssrc_map_item (ts_stats.stream, &ts_stats,
(FindSsrcMapFunc) webrtc_stats_get_from_transport_for_one_ssrc);
g_clear_object (&rtp_session);
g_clear_object (&gst_rtp_session);
gst_clear_structure (&rtp_stats);
gst_clear_structure (&twcc_stats);
g_value_array_free (ts_stats.source_stats);
ts_stats.source_stats = NULL;
g_clear_pointer (&ts_stats.transport_id, g_free);
out:
g_clear_pointer (&ts_stats.codec_id, g_free);
return TRUE;
}
GstStructure *
gst_webrtc_bin_create_stats (GstWebRTCBin * webrtc, GstPad * pad)
{
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);
}
if (pad)
_get_stats_from_pad (webrtc, pad, s);
else
gst_element_foreach_pad (GST_ELEMENT (webrtc),
(GstElementForeachPadFunc) _get_stats_from_pad, s);
gst_structure_remove_field (s, "timestamp");
return s;
}
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