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gstreamer/ext/webrtc/gstwebrtcstats.c

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webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
/* 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)) {
webrtc/stats: rename debug category not to be ice related
2018-09-10 13:27:08 +00:00
GST_DEBUG_CATEGORY_INIT (gst_webrtc_stats_debug, "webrtcstats", 0,
"webrtcstats");
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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;
}
webrtcstats: Avoid copy of GstStructure Instead transfer the ownership to the new structure Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:59:58 +00:00
static void
_gst_structure_take_structure (GstStructure * s, const char *fieldname,
GstStructure ** value_s)
{
GValue v = G_VALUE_INIT;
g_value_init (&v, GST_TYPE_STRUCTURE);
g_value_take_boxed (&v, *value_s);
gst_structure_take_value (s, fieldname, &v);
*value_s = NULL;
}
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
#define CLOCK_RATE_VALUE_TO_SECONDS(v,r) ((double) v / (double) clock_rate)
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
#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))
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
/* https://www.w3.org/TR/webrtc-stats/#remoteinboundrtpstats-dict* */
static gboolean
_get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc,
TransportStream * stream, const GstStructure * source_stats,
guint ssrc, guint clock_rate, const gchar * codec_id,
const gchar * transport_id, GstStructure * s)
{
gboolean have_rb = FALSE, internal = 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, "internal", G_TYPE_BOOLEAN, &internal,
"have-rb", G_TYPE_BOOLEAN, &have_rb, NULL);
/* This isn't what we're looking for */
if (internal == TRUE || 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);
/* To be added: kind */
/* RTCReceivedRtpStreamStats */
if (gst_structure_get_int (source_stats, "rb-packetslost", &lost))
gst_structure_set (r_in, "packets-lost", G_TYPE_INT, 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;
}
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
/* 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,
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
TransportStream * stream, const GstStructure * source_stats,
const gchar * codec_id, const gchar * transport_id, GstStructure * s)
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
{
guint ssrc, fir, pli, nack, jitter;
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
int clock_rate;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
guint64 packets, bytes;
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gboolean internal;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
double ts;
gst_structure_get_double (s, "timestamp", &ts);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
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) {
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
GstStructure *out;
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gchar *out_id, *r_in_id;
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
out = gst_structure_new_empty (out_id);
_set_base_stats (out, GST_WEBRTC_STATS_OUTBOUND_RTP, ts, out_id);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
/* 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);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* To be added: kind */
/* RTCSentRtpStreamStats */
if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
gst_structure_set (out, "bytes-sent", G_TYPE_UINT64, bytes, NULL);
if (gst_structure_get_uint64 (source_stats, "packets-sent", &packets))
gst_structure_set (out, "packets-sent", G_TYPE_UINT64, packets, NULL);
/* RTCOutboundRTPStreamStats */
webrtcstats: PLI/FIR/NACK direction are the opposite of the media Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
2020-12-29 18:29:05 +00:00
if (gst_structure_get_uint (source_stats, "recv-fir-count", &fir))
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gst_structure_set (out, "fir-count", G_TYPE_UINT, fir, NULL);
webrtcstats: PLI/FIR/NACK direction are the opposite of the media Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
2020-12-29 18:29:05 +00:00
if (gst_structure_get_uint (source_stats, "recv-pli-count", &pli))
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gst_structure_set (out, "pli-count", G_TYPE_UINT, pli, NULL);
webrtcstats: PLI/FIR/NACK direction are the opposite of the media Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
2020-12-29 18:29:05 +00:00
if (gst_structure_get_uint (source_stats, "recv-nack-count", &nack))
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gst_structure_set (out, "nack-count", G_TYPE_UINT, nack, NULL);
/* XXX: mediaType, trackId, sliCount, qpSum */
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
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);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* 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;
*/
webrtcstats: Also return the raw rtpsource stats for more information Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 22:21:19 +00:00
/* 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);
webrtcstats: Avoid copy of GstStructure Instead transfer the ownership to the new structure Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:59:58 +00:00
_gst_structure_take_structure (s, out_id, &out);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
g_free (out_id);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
} else {
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
GstStructure *in, *r_out;
gchar *r_out_id, *in_id;
webrtcstats: RTCP computed RTT is only available at sender The receiver doesn't have the information to compute it. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-09 23:38:15 +00:00
gboolean have_sr = FALSE;
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
GstStructure *jb_stats = NULL;
guint i;
guint64 jb_lost, duplicates, late, rtx_success;
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
webrtcstats: RTCP computed RTT is only available at sender The receiver doesn't have the information to compute it. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-09 23:38:15 +00:00
gst_structure_get (source_stats, "have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
for (i = 0; i < stream->remote_ssrcmap->len; i++) {
webrtc: Make ssrc map into separate data structures They now contain a weak reference and that could be freed later causing strange crashes as GWeakRef are not movable. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-20 22:32:44 +00:00
SsrcMapItem *item = g_ptr_array_index (stream->remote_ssrcmap, i);
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
if (item->ssrc == ssrc) {
GObject *jb = g_weak_ref_get (&item->rtpjitterbuffer);
if (jb) {
g_object_get (jb, "stats", &jb_stats, NULL);
g_object_unref (jb);
}
break;
}
}
if (jb_stats)
gst_structure_get (jb_stats, "num-lost", G_TYPE_UINT64, &jb_lost,
"num-duplicates", G_TYPE_UINT64, &duplicates, "num-late",
G_TYPE_UINT64, &late, "rtx-success-count", G_TYPE_UINT64,
&rtx_success, NULL);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
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);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* RTCRtpStreamStats */
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
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);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* To be added: kind */
/* RTCReceivedRtpStreamStats */
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
gst_structure_set (in, "packets-received", G_TYPE_UINT64, packets, NULL);
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
if (jb_stats)
gst_structure_set (in, "packets-lost", G_TYPE_UINT64, jb_lost, NULL);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
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);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
if (jb_stats)
gst_structure_set (in, "packets-discarded", G_TYPE_UINT64, late,
"packets-repaired", G_TYPE_UINT64, rtx_success, NULL);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/*
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 */
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gst_structure_set (in, "remote-id", G_TYPE_STRING, r_out_id, NULL);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
if (gst_structure_get_uint64 (source_stats, "octets-received", &bytes))
gst_structure_set (in, "bytes-received", G_TYPE_UINT64, bytes, NULL);
webrtcstats: PLI/FIR/NACK direction are the opposite of the media Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
2020-12-29 18:29:05 +00:00
if (gst_structure_get_uint (source_stats, "sent-fir-count", &fir))
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
gst_structure_set (in, "fir-count", G_TYPE_UINT, fir, NULL);
webrtcstats: PLI/FIR/NACK direction are the opposite of the media Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
2020-12-29 18:29:05 +00:00
if (gst_structure_get_uint (source_stats, "sent-pli-count", &pli))
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
gst_structure_set (in, "pli-count", G_TYPE_UINT, pli, NULL);
webrtcstats: PLI/FIR/NACK direction are the opposite of the media Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
2020-12-29 18:29:05 +00:00
if (gst_structure_get_uint (source_stats, "sent-nack-count", &nack))
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
gst_structure_set (in, "nack-count", G_TYPE_UINT, nack, NULL);
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
if (jb_stats)
gst_structure_set (in, "packets-duplicated", G_TYPE_UINT64, duplicates,
NULL);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* 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;
*/
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
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);
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* XXX: mediaType, trackId */
/* RTCSentRtpStreamStats */
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
if (have_sr) {
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
guint sr_bytes, sr_packets;
if (gst_structure_get_uint (source_stats, "sr-octet-count", &sr_bytes))
gst_structure_set (r_out, "bytes-sent", G_TYPE_UINT, sr_bytes, NULL);
if (gst_structure_get_uint (source_stats, "sr-packet-count", &sr_packets))
gst_structure_set (r_out, "packets-sent", G_TYPE_UINT, sr_packets,
NULL);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
}
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* 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 */
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
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);
}
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
gst_structure_set (r_out, "local-id", G_TYPE_STRING, in_id, NULL);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtcstats: Document all RTP missing fields according to the latest spec 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>
2020-10-09 23:59:18 +00:00
/* To be added:
reportsSent
*/
webrtcstats: Also return the raw rtpsource stats for more information Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 22:21:19 +00:00
/* Store the raw stats from GStreamer into the structure for advanced
* information.
*/
_gst_structure_take_structure (in, "gst-rtpjitterbuffer-stats", &jb_stats);
gst_structure_set (in, "gst-rtpsource-stats", GST_TYPE_STRUCTURE,
source_stats, NULL);
webrtcstats: Extract statistics from the rtpjitterbuffer And expose them as standardised webrtc statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:27:40 +00:00
webrtcstats: Avoid copy of GstStructure Instead transfer the ownership to the new structure Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-10 00:59:58 +00:00
_gst_structure_take_structure (s, in_id, &in);
_gst_structure_take_structure (s, r_out_id, &r_out);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
g_free (in_id);
g_free (r_out_id);
}
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
}
/* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* */
static gchar *
_get_stats_from_ice_transport (GstWebRTCBin * webrtc,
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
GstWebRTCICETransport * transport, const GstStructure * twcc_stats,
GstStructure * s)
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
{
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;
};
*/
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
/* 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);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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,
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
GstWebRTCDTLSTransport * transport, const GstStructure * twcc_stats,
GstStructure * s)
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
{
GstStructure *stats;
gchar *id;
double ts;
webrtcbin: Fixed memory leak in gstwebrtcstats The function _get_stats_from_ice_transport returns a string which must be freed by the caller. However, _get_stats_from_dtls_transport was ignoring the return value from this function, resulting in a leak. Ran this with valgrind. Before this fix there was a leak of 40 bytes each time this was called. After there was no leak.
2019-08-29 00:26:00 +00:00
gchar *ice_id;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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);
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
ice_id =
_get_stats_from_ice_transport (webrtc, transport->transport, twcc_stats,
s);
webrtcbin: Fixed memory leak in gstwebrtcstats The function _get_stats_from_ice_transport returns a string which must be freed by the caller. However, _get_stats_from_dtls_transport was ignoring the return value from this function, resulting in a leak. Ran this with valgrind. Before this fix there was a leak of 40 bytes each time this was called. After there was no leak.
2019-08-29 00:26:00 +00:00
g_free (ice_id);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
return id;
}
static void
_get_stats_from_transport_channel (GstWebRTCBin * webrtc,
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
2019-03-07 13:13:14 +00:00
TransportStream * stream, const gchar * codec_id, guint ssrc,
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
guint clock_rate, GstStructure * s)
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
{
GstWebRTCDTLSTransport *transport;
GObject *rtp_session;
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
GObject *gst_rtp_session;
GstStructure *rtp_stats, *twcc_stats;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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);
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
g_signal_emit_by_name (webrtc->rtpbin, "get-session",
stream->session_id, &gst_rtp_session);
g_object_get (gst_rtp_session, "twcc-stats", &twcc_stats, NULL);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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);
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
transport_id =
_get_stats_from_dtls_transport (webrtc, transport, twcc_stats, s);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
/* 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);
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
2019-03-07 13:13:14 +00:00
guint stats_ssrc = 0;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
stats = gst_value_get_structure (val);
webrtc/stats: redo considering internal sources Internal sources seem to be rtp streams we are sending whereas non-internal sources are the rtp streams we are receiving. Redo the statistics with that in mind.
2019-08-12 09:42:12 +00:00
/* skip foreign sources */
gst_structure_get (stats, "ssrc", G_TYPE_UINT, &stats_ssrc, NULL);
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
ssrc == stats_ssrc)
_get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
transport_id, s);
else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc) &&
ssrc == stats_ssrc)
_get_stats_from_remote_rtp_source_stats (webrtc, stream, stats, ssrc,
clock_rate, codec_id, transport_id, s);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
}
g_object_unref (rtp_session);
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
g_object_unref (gst_rtp_session);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
gst_structure_free (rtp_stats);
webrtc/stats: provide the twcc stats when available Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2207>
2021-04-29 11:11:25 +00:00
if (twcc_stats)
gst_structure_free (twcc_stats);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
g_value_array_free (source_stats);
g_free (transport_id);
}
/* https://www.w3.org/TR/webrtc-stats/#codec-dict* */
webrtcbin: Syntax cleanup
2019-03-07 13:12:47 +00:00
static void
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
_get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
GstStructure * s, gchar ** out_id, guint * out_ssrc, guint * out_clock_rate)
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
{
GstStructure *stats;
GstCaps *caps;
gchar *id;
double ts;
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
2019-03-07 13:13:14 +00:00
guint ssrc = 0;
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
gint clock_rate = 0;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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);
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
gint pt;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
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);
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
2019-03-07 13:13:14 +00:00
if (gst_structure_get_uint (caps_s, "ssrc", &ssrc))
gst_structure_set (stats, "ssrc", G_TYPE_UINT, ssrc, NULL);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
/* 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);
webrtcbin: Syntax cleanup
2019-03-07 13:12:47 +00:00
if (out_id)
*out_id = id;
else
g_free (id);
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
2019-03-07 13:13:14 +00:00
if (out_ssrc)
*out_ssrc = ssrc;
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
if (out_clock_rate)
*out_clock_rate = clock_rate;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
}
static gboolean
_get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
{
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
webrtcbin: Syntax cleanup
2019-03-07 13:12:47 +00:00
TransportStream *stream;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
gchar *codec_id;
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
guint ssrc, clock_rate;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
_get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc, &clock_rate);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtcbin: Syntax cleanup
2019-03-07 13:12:47 +00:00
if (!wpad->trans)
goto out;
stream = WEBRTC_TRANSCEIVER (wpad->trans)->stream;
if (!stream)
goto out;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
webrtcstats: Get the remote-inbound stats from the right RTPSource This also means that we need to get the clock-rate from the codec instead of from the RTPSource, as the remote one doesn't include a clock rate. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-16 01:23:08 +00:00
_get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc,
clock_rate, s);
webrtcbin: Syntax cleanup
2019-03-07 13:12:47 +00:00
out:
g_free (codec_id);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
return TRUE;
}
webrtcbin: Implement getting stats for a specific pad Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-15 23:36:45 +00:00
GstStructure *
gst_webrtc_bin_create_stats (GstWebRTCBin * webrtc, GstPad * pad)
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
{
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);
}
webrtcbin: Implement getting stats for a specific pad Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-15 23:36:45 +00:00
if (pad)
_get_stats_from_pad (webrtc, pad, s);
else
gst_element_foreach_pad (GST_ELEMENT (webrtc),
(GstElementForeachPadFunc) _get_stats_from_pad, s);
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
gst_structure_remove_field (s, "timestamp");
webrtcbin: Implement getting stats for a specific pad Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-10-15 23:36:45 +00:00
return s;
webrtcbin: an element that handles the transport aspects of webrtc connections 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
2017-01-31 09:56:59 +00:00
}
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