Revert "Reland "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.""
This reverts commit 2843bbc96dc3deee0406bac5ec9440cf3385ef2e.
Reason for revert: Even more references to unimplemented metrics remaining...
Original change's description:
> Reland "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs."
>
> This is a reland of commit 626f87d90501fd8d7a4ea071686cd8befd0d430c
>
> Original change's description:
> > [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
> >
> > In preparation for the spec moving closer to PR, let's not have
> > placeholder metrics not implemented.
> >
> > Bug: webrtc:14167
> > Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37205}
>
> Bug: webrtc:14167
> Change-Id: Ifdc37e7a48fea516c727c06d2f510780386cb204
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265805
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37215}
Bug: webrtc:14167
Change-Id: I959d61512d5896224302a70aadbac6f75afc819e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265810
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37217}
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 6984169..7f93c1c 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -176,7 +176,7 @@
RTCStatsMember<std::string> label;
RTCStatsMember<std::string> protocol;
RTCStatsMember<int32_t> data_channel_identifier;
- // Enum type RTCDataChannelState.
+ // TODO(hbos): Support enum types? "RTCStatsMember<RTCDataChannelState>"?
RTCStatsMember<std::string> state;
RTCStatsMember<uint32_t> messages_sent;
RTCStatsMember<uint64_t> bytes_sent;
@@ -185,6 +185,7 @@
};
// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
+// TODO(hbos): Tracking bug https://bugs.webrtc.org/7062
class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -197,16 +198,17 @@
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> local_candidate_id;
RTCStatsMember<std::string> remote_candidate_id;
- // Enum type RTCStatsIceCandidatePairState.
+ // TODO(hbos): Support enum types?
+ // "RTCStatsMember<RTCStatsIceCandidatePairState>"?
RTCStatsMember<std::string> state;
// Obsolete: priority
RTCStatsMember<uint64_t> priority;
RTCStatsMember<bool> nominated;
- // `writable` does not exist in the spec and old comments suggest it used to
- // exist but was incorrectly implemented.
- // TODO(https://crbug.com/webrtc/14171): Standardize and/or modify
- // implementation.
+ // TODO(hbos): Collect this the way the spec describes it. We have a value for
+ // it but it is not spec-compliant. https://bugs.webrtc.org/7062
RTCStatsMember<bool> writable;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
+ RTCStatsMember<bool> readable;
RTCStatsMember<uint64_t> packets_sent;
RTCStatsMember<uint64_t> packets_received;
RTCStatsMember<uint64_t> bytes_sent;
@@ -214,17 +216,35 @@
RTCStatsMember<double> total_round_trip_time;
RTCStatsMember<double> current_round_trip_time;
RTCStatsMember<double> available_outgoing_bitrate;
+ // TODO(hbos): Populate this value. It is wired up and collected the same way
+ // "VideoBwe.googAvailableReceiveBandwidth" is, but that value is always
+ // undefined. https://bugs.webrtc.org/7062
RTCStatsMember<double> available_incoming_bitrate;
RTCStatsMember<uint64_t> requests_received;
RTCStatsMember<uint64_t> requests_sent;
RTCStatsMember<uint64_t> responses_received;
RTCStatsMember<uint64_t> responses_sent;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
+ RTCStatsMember<uint64_t> retransmissions_received;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
+ RTCStatsMember<uint64_t> retransmissions_sent;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
+ RTCStatsMember<uint64_t> consent_requests_received;
RTCStatsMember<uint64_t> consent_requests_sent;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
+ RTCStatsMember<uint64_t> consent_responses_received;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
+ RTCStatsMember<uint64_t> consent_responses_sent;
RTCStatsMember<uint64_t> packets_discarded_on_send;
RTCStatsMember<uint64_t> bytes_discarded_on_send;
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
+// TODO(hbos): `RTCStatsCollector` only collects candidates that are part of
+// ice candidate pairs, but there could be candidates not paired with anything.
+// crbug.com/632723
+// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
+// them in the new PeerConnection::GetStats.
class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -241,7 +261,7 @@
RTCStatsMember<int32_t> port;
RTCStatsMember<std::string> protocol;
RTCStatsMember<std::string> relay_protocol;
- // Enum type RTCIceCandidateType.
+ // TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
RTCStatsMember<std::string> candidate_type;
RTCStatsMember<int32_t> priority;
RTCStatsMember<std::string> url;
@@ -280,8 +300,8 @@
const char* type() const override;
};
-// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats
-// TODO(https://crbug.com/webrtc/14172): Deprecate and remove.
+// https://w3c.github.io/webrtc-stats/#msstats-dict*
+// TODO(hbos): Tracking bug crbug.com/660827
class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -295,8 +315,8 @@
RTCStatsMember<std::vector<std::string>> track_ids;
};
-// TODO(https://crbug.com/webrtc/14175): Deprecate and remove in favor of
-// RTCMediaSourceStats/RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats.
+// https://w3c.github.io/webrtc-stats/#mststats-dict*
+// TODO(hbos): Tracking bug crbug.com/659137
class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -314,20 +334,29 @@
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<bool> remote_source;
RTCStatsMember<bool> ended;
- // TODO(https://crbug.com/webrtc/14173): Remove this obsolete metric.
+ // TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks.
+ // crbug.com/659137
RTCStatsMember<bool> detached;
- // Enum type RTCMediaStreamTrackKind.
+ // See `RTCMediaStreamTrackKind` for valid values.
RTCStatsMember<std::string> kind;
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
// Video-only members
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
+ // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
+ RTCStatsMember<double> frames_per_second;
RTCStatsMember<uint32_t> frames_sent;
RTCStatsMember<uint32_t> huge_frames_sent;
RTCStatsMember<uint32_t> frames_received;
RTCStatsMember<uint32_t> frames_decoded;
RTCStatsMember<uint32_t> frames_dropped;
+ // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
+ RTCStatsMember<uint32_t> frames_corrupted;
+ // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
+ RTCStatsMember<uint32_t> partial_frames_lost;
+ // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
+ RTCStatsMember<uint32_t> full_frames_lost;
// Audio-only members
RTCStatsMember<double> audio_level; // Receive-only
RTCStatsMember<double> total_audio_energy; // Receive-only
@@ -341,7 +370,7 @@
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
// Non-standard audio-only member
- // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcaudioreceiverstats-jitterbufferflushes
+ // TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
@@ -351,15 +380,14 @@
// delay, in seconds, at the time that the sample was emitted from the jitter
// buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
// Currently it is implemented only for audio.
- // TODO(https://crbug.com/webrtc/14176): This should be moved to
- // RTCInboundRtpStreamStats and it should be implemented for video as well.
+ // TODO(titovartem) implement for video streams when will be requested.
RTCNonStandardStatsMember<double> jitter_buffer_target_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
- // https://github.com/w3c/webrtc-provisional-stats/issues/17
+ // https://github.com/henbos/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// Non-standard video-only members.
- // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcvideoreceiverstats
+ // https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
RTCNonStandardStatsMember<uint32_t> freeze_count;
RTCNonStandardStatsMember<uint32_t> pause_count;
RTCNonStandardStatsMember<double> total_freezes_duration;
@@ -383,6 +411,7 @@
};
// https://w3c.github.io/webrtc-stats/#streamstats-dict*
+// TODO(hbos): Tracking bug crbug.com/657854
class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -413,6 +442,13 @@
RTCReceivedRtpStreamStats(const RTCReceivedRtpStreamStats& other);
~RTCReceivedRtpStreamStats() override;
+ // TODO(hbos) The following fields need to be added and migrated
+ // both from RTCInboundRtpStreamStats and RTCRemoteInboundRtpStreamStats:
+ // packetsReceived, packetsRepaired, burstPacketsLost,
+ // burstPacketDiscarded, burstLossCount, burstDiscardCount, burstLossRate,
+ // burstDiscardRate, gapLossRate, gapDiscardRate, framesDropped,
+ // partialFramesLost, fullFramesLost
+ // crbug.com/webrtc/12532
RTCStatsMember<double> jitter;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<uint64_t> packets_discarded;
@@ -439,6 +475,8 @@
};
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
+// TODO(hbos): Support the remote case |is_remote = true|.
+// https://bugs.webrtc.org/7065
class RTC_EXPORT RTCInboundRTPStreamStats final
: public RTCReceivedRtpStreamStats {
public:
@@ -449,8 +487,6 @@
RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
~RTCInboundRTPStreamStats() override;
- // TODO(https://crbug.com/webrtc/14174): Implement trackIdentifier and kind.
-
RTCStatsMember<std::string> remote_id;
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> fec_packets_received;
@@ -469,28 +505,48 @@
RTCStatsMember<double> audio_level;
RTCStatsMember<double> total_audio_energy;
RTCStatsMember<double> total_samples_duration;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<double> round_trip_time;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<uint32_t> packets_repaired;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<uint32_t> burst_packets_lost;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<uint32_t> burst_packets_discarded;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<uint32_t> burst_loss_count;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<uint32_t> burst_discard_count;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<double> burst_loss_rate;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<double> burst_discard_rate;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<double> gap_loss_rate;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<double> gap_discard_rate;
// Stats below are only implemented or defined for video.
RTCStatsMember<int32_t> frames_received;
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
+ RTCStatsMember<uint32_t> frame_bit_depth;
RTCStatsMember<double> frames_per_second;
RTCStatsMember<uint32_t> frames_decoded;
RTCStatsMember<uint32_t> key_frames_decoded;
RTCStatsMember<uint32_t> frames_dropped;
RTCStatsMember<double> total_decode_time;
RTCStatsMember<double> total_processing_delay;
- // TODO(https://crbug.com/webrtc/13986): standardize
+ // TODO(bugs.webrtc.org/13986): standardize
RTCNonStandardStatsMember<double> total_assembly_time;
RTCNonStandardStatsMember<uint32_t> frames_assembled_from_multiple_packets;
RTCStatsMember<double> total_inter_frame_delay;
RTCStatsMember<double> total_squared_inter_frame_delay;
- // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
+ // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
RTCStatsMember<std::string> content_type;
- // Only populated if audio/video sync is enabled.
- // TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off?
+ // TODO(asapersson): Currently only populated if audio/video sync is enabled.
RTCStatsMember<double> estimated_playout_timestamp;
- // Only implemented for video.
- // TODO(https://crbug.com/webrtc/14178): Also implement for audio.
+ // TODO(hbos): This is only implemented for video; implement it for audio as
+ // well.
RTCStatsMember<std::string> decoder_implementation;
// FIR and PLI counts are only defined for |kind == "video"|.
RTCStatsMember<uint32_t> fir_count;
@@ -503,6 +559,8 @@
};
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
+// TODO(hbos): Support the remote case |is_remote = true|.
+// https://bugs.webrtc.org/7066
class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -539,10 +597,10 @@
RTCStatsMember<std::map<std::string, double>> quality_limitation_durations;
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
RTCStatsMember<uint32_t> quality_limitation_resolution_changes;
- // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
+ // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
RTCStatsMember<std::string> content_type;
- // Only implemented for video.
- // TODO(https://crbug.com/webrtc/14178): Implement for audio as well.
+ // TODO(hbos): This is only implemented for video; implement it for audio as
+ // well.
RTCStatsMember<std::string> encoder_implementation;
// FIR and PLI counts are only defined for |kind == "video"|.
RTCStatsMember<uint32_t> fir_count;
@@ -562,6 +620,11 @@
RTCRemoteInboundRtpStreamStats(const RTCRemoteInboundRtpStreamStats& other);
~RTCRemoteInboundRtpStreamStats() override;
+ // TODO(hbos): The following RTCReceivedRtpStreamStats metrics should also be
+ // implemented: packetsReceived, packetsRepaired,
+ // burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount,
+ // burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate.
+ // RTCRemoteInboundRtpStreamStats
RTCStatsMember<std::string> local_id;
RTCStatsMember<double> round_trip_time;
RTCStatsMember<double> fraction_lost;
@@ -652,7 +715,7 @@
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<uint64_t> packets_received;
RTCStatsMember<std::string> rtcp_transport_stats_id;
- // Enum type RTCDtlsTransportState.
+ // TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
RTCStatsMember<std::string> dtls_state;
RTCStatsMember<std::string> selected_candidate_pair_id;
RTCStatsMember<std::string> local_certificate_id;
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index cc3e8e1..b5d86a9 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -486,6 +486,7 @@
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.priority);
verifier.TestMemberIsDefined(candidate_pair.nominated);
verifier.TestMemberIsDefined(candidate_pair.writable);
+ verifier.TestMemberIsUndefined(candidate_pair.readable);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(
candidate_pair.packets_discarded_on_send);
@@ -513,8 +514,13 @@
verifier.TestMemberIsNonNegative<uint64_t>(
candidate_pair.responses_received);
verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.responses_sent);
+ verifier.TestMemberIsUndefined(candidate_pair.retransmissions_received);
+ verifier.TestMemberIsUndefined(candidate_pair.retransmissions_sent);
+ verifier.TestMemberIsUndefined(candidate_pair.consent_requests_received);
verifier.TestMemberIsNonNegative<uint64_t>(
candidate_pair.consent_requests_sent);
+ verifier.TestMemberIsUndefined(candidate_pair.consent_responses_received);
+ verifier.TestMemberIsUndefined(candidate_pair.consent_responses_sent);
return verifier.ExpectAllMembersSuccessfullyTested();
}
@@ -630,6 +636,10 @@
media_stream_track.frame_width);
verifier.TestMemberIsNonNegative<uint32_t>(
media_stream_track.frame_height);
+ verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
+ verifier.TestMemberIsUndefined(media_stream_track.frames_corrupted);
+ verifier.TestMemberIsUndefined(media_stream_track.partial_frames_lost);
+ verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
// Audio-only members should be undefined
verifier.TestMemberIsUndefined(media_stream_track.audio_level);
verifier.TestMemberIsUndefined(media_stream_track.echo_return_loss);
@@ -731,11 +741,15 @@
// Video-only members should be undefined
verifier.TestMemberIsUndefined(media_stream_track.frame_width);
verifier.TestMemberIsUndefined(media_stream_track.frame_height);
+ verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
verifier.TestMemberIsUndefined(media_stream_track.frames_received);
verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
+ verifier.TestMemberIsUndefined(media_stream_track.frames_corrupted);
+ verifier.TestMemberIsUndefined(media_stream_track.partial_frames_lost);
+ verifier.TestMemberIsUndefined(media_stream_track.full_frames_lost);
verifier.TestMemberIsUndefined(media_stream_track.freeze_count);
verifier.TestMemberIsUndefined(media_stream_track.pause_count);
verifier.TestMemberIsUndefined(media_stream_track.total_freezes_duration);
@@ -836,6 +850,7 @@
} else {
verifier.TestMemberIsUndefined(inbound_stream.frames_per_second);
}
+ verifier.TestMemberIsUndefined(inbound_stream.frame_bit_depth);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
@@ -879,6 +894,16 @@
inbound_stream.total_samples_duration);
verifier.TestMemberIsUndefined(inbound_stream.frames_received);
}
+ verifier.TestMemberIsUndefined(inbound_stream.round_trip_time);
+ verifier.TestMemberIsUndefined(inbound_stream.packets_repaired);
+ verifier.TestMemberIsUndefined(inbound_stream.burst_packets_lost);
+ verifier.TestMemberIsUndefined(inbound_stream.burst_packets_discarded);
+ verifier.TestMemberIsUndefined(inbound_stream.burst_loss_count);
+ verifier.TestMemberIsUndefined(inbound_stream.burst_discard_count);
+ verifier.TestMemberIsUndefined(inbound_stream.burst_loss_rate);
+ verifier.TestMemberIsUndefined(inbound_stream.burst_discard_rate);
+ verifier.TestMemberIsUndefined(inbound_stream.gap_loss_rate);
+ verifier.TestMemberIsUndefined(inbound_stream.gap_discard_rate);
// Test runtime too short to get an estimate (at least two RTCP sender
// reports need to be received).
verifier.MarkMemberTested(inbound_stream.estimated_playout_timestamp, true);
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index a087359..b0003c3 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -191,6 +191,7 @@
&priority,
&nominated,
&writable,
+ &readable,
&packets_sent,
&packets_received,
&bytes_sent,
@@ -203,7 +204,12 @@
&requests_sent,
&responses_received,
&responses_sent,
+ &retransmissions_received,
+ &retransmissions_sent,
+ &consent_requests_received,
&consent_requests_sent,
+ &consent_responses_received,
+ &consent_responses_sent,
&packets_discarded_on_send,
&bytes_discarded_on_send)
// clang-format on
@@ -222,6 +228,7 @@
priority("priority"),
nominated("nominated"),
writable("writable"),
+ readable("readable"),
packets_sent("packetsSent"),
packets_received("packetsReceived"),
bytes_sent("bytesSent"),
@@ -234,7 +241,12 @@
requests_sent("requestsSent"),
responses_received("responsesReceived"),
responses_sent("responsesSent"),
+ retransmissions_received("retransmissionsReceived"),
+ retransmissions_sent("retransmissionsSent"),
+ consent_requests_received("consentRequestsReceived"),
consent_requests_sent("consentRequestsSent"),
+ consent_responses_received("consentResponsesReceived"),
+ consent_responses_sent("consentResponsesSent"),
packets_discarded_on_send("packetsDiscardedOnSend"),
bytes_discarded_on_send("bytesDiscardedOnSend") {}
@@ -248,6 +260,7 @@
priority(other.priority),
nominated(other.nominated),
writable(other.writable),
+ readable(other.readable),
packets_sent(other.packets_sent),
packets_received(other.packets_received),
bytes_sent(other.bytes_sent),
@@ -260,7 +273,12 @@
requests_sent(other.requests_sent),
responses_received(other.responses_received),
responses_sent(other.responses_sent),
+ retransmissions_received(other.retransmissions_received),
+ retransmissions_sent(other.retransmissions_sent),
+ consent_requests_received(other.consent_requests_received),
consent_requests_sent(other.consent_requests_sent),
+ consent_responses_received(other.consent_responses_received),
+ consent_responses_sent(other.consent_responses_sent),
packets_discarded_on_send(other.packets_discarded_on_send),
bytes_discarded_on_send(other.bytes_discarded_on_send) {}
@@ -394,11 +412,15 @@
&jitter_buffer_emitted_count,
&frame_width,
&frame_height,
+ &frames_per_second,
&frames_sent,
&huge_frames_sent,
&frames_received,
&frames_decoded,
&frames_dropped,
+ &frames_corrupted,
+ &partial_frames_lost,
+ &full_frames_lost,
&audio_level,
&total_audio_energy,
&echo_return_loss,
@@ -443,11 +465,15 @@
jitter_buffer_emitted_count("jitterBufferEmittedCount"),
frame_width("frameWidth"),
frame_height("frameHeight"),
+ frames_per_second("framesPerSecond"),
frames_sent("framesSent"),
huge_frames_sent("hugeFramesSent"),
frames_received("framesReceived"),
frames_decoded("framesDecoded"),
frames_dropped("framesDropped"),
+ frames_corrupted("framesCorrupted"),
+ partial_frames_lost("partialFramesLost"),
+ full_frames_lost("fullFramesLost"),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
echo_return_loss("echoReturnLoss"),
@@ -495,11 +521,15 @@
jitter_buffer_emitted_count(other.jitter_buffer_emitted_count),
frame_width(other.frame_width),
frame_height(other.frame_height),
+ frames_per_second(other.frames_per_second),
frames_sent(other.frames_sent),
huge_frames_sent(other.huge_frames_sent),
frames_received(other.frames_received),
frames_decoded(other.frames_decoded),
frames_dropped(other.frames_dropped),
+ frames_corrupted(other.frames_corrupted),
+ partial_frames_lost(other.partial_frames_lost),
+ full_frames_lost(other.full_frames_lost),
audio_level(other.audio_level),
total_audio_energy(other.total_audio_energy),
echo_return_loss(other.echo_return_loss),
@@ -658,9 +688,20 @@
&audio_level,
&total_audio_energy,
&total_samples_duration,
+ &round_trip_time,
+ &packets_repaired,
+ &burst_packets_lost,
+ &burst_packets_discarded,
+ &burst_loss_count,
+ &burst_discard_count,
+ &burst_loss_rate,
+ &burst_discard_rate,
+ &gap_loss_rate,
+ &gap_discard_rate,
&frames_received,
&frame_width,
&frame_height,
+ &frame_bit_depth,
&frames_per_second,
&frames_decoded,
&key_frames_decoded,
@@ -706,9 +747,20 @@
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration"),
+ round_trip_time("roundTripTime"),
+ packets_repaired("packetsRepaired"),
+ burst_packets_lost("burstPacketsLost"),
+ burst_packets_discarded("burstPacketsDiscarded"),
+ burst_loss_count("burstLossCount"),
+ burst_discard_count("burstDiscardCount"),
+ burst_loss_rate("burstLossRate"),
+ burst_discard_rate("burstDiscardRate"),
+ gap_loss_rate("gapLossRate"),
+ gap_discard_rate("gapDiscardRate"),
frames_received("framesReceived"),
frame_width("frameWidth"),
frame_height("frameHeight"),
+ frame_bit_depth("frameBitDepth"),
frames_per_second("framesPerSecond"),
frames_decoded("framesDecoded"),
key_frames_decoded("keyFramesDecoded"),
@@ -751,9 +803,20 @@
audio_level(other.audio_level),
total_audio_energy(other.total_audio_energy),
total_samples_duration(other.total_samples_duration),
+ round_trip_time(other.round_trip_time),
+ packets_repaired(other.packets_repaired),
+ burst_packets_lost(other.burst_packets_lost),
+ burst_packets_discarded(other.burst_packets_discarded),
+ burst_loss_count(other.burst_loss_count),
+ burst_discard_count(other.burst_discard_count),
+ burst_loss_rate(other.burst_loss_rate),
+ burst_discard_rate(other.burst_discard_rate),
+ gap_loss_rate(other.gap_loss_rate),
+ gap_discard_rate(other.gap_discard_rate),
frames_received(other.frames_received),
frame_width(other.frame_width),
frame_height(other.frame_height),
+ frame_bit_depth(other.frame_bit_depth),
frames_per_second(other.frames_per_second),
frames_decoded(other.frames_decoded),
key_frames_decoded(other.key_frames_decoded),
