From 626f87d90501fd8d7a4ea071686cd8befd0d430c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Henrik=20Bostr=C3=B6m?= Date: Mon, 13 Jun 2022 10:23:28 +0200 Subject: [PATCH] [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 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 Commit-Queue: Henrik Boström Cr-Commit-Position: refs/heads/main@{#37205} --- api/stats/rtcstats_objects.h | 123 ++++++++------------------------ pc/rtc_stats_integrationtest.cc | 25 ------- stats/rtcstats_objects.cc | 63 ---------------- 3 files changed, 30 insertions(+), 181 deletions(-) diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h index 7f93c1c91c..6984169b41 100644 --- a/api/stats/rtcstats_objects.h +++ b/api/stats/rtcstats_objects.h @@ -176,7 +176,7 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats { RTCStatsMember label; RTCStatsMember protocol; RTCStatsMember data_channel_identifier; - // TODO(hbos): Support enum types? "RTCStatsMember"? + // Enum type RTCDataChannelState. RTCStatsMember state; RTCStatsMember messages_sent; RTCStatsMember bytes_sent; @@ -185,7 +185,6 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats { }; // 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(); @@ -198,17 +197,16 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats { RTCStatsMember transport_id; RTCStatsMember local_candidate_id; RTCStatsMember remote_candidate_id; - // TODO(hbos): Support enum types? - // "RTCStatsMember"? + // Enum type RTCStatsIceCandidatePairState. RTCStatsMember state; // Obsolete: priority RTCStatsMember priority; RTCStatsMember nominated; - // 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 + // `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. RTCStatsMember writable; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062 - RTCStatsMember readable; RTCStatsMember packets_sent; RTCStatsMember packets_received; RTCStatsMember bytes_sent; @@ -216,35 +214,17 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats { RTCStatsMember total_round_trip_time; RTCStatsMember current_round_trip_time; RTCStatsMember 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 available_incoming_bitrate; RTCStatsMember requests_received; RTCStatsMember requests_sent; RTCStatsMember responses_received; RTCStatsMember responses_sent; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062 - RTCStatsMember retransmissions_received; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062 - RTCStatsMember retransmissions_sent; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062 - RTCStatsMember consent_requests_received; RTCStatsMember consent_requests_sent; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062 - RTCStatsMember consent_responses_received; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062 - RTCStatsMember consent_responses_sent; RTCStatsMember packets_discarded_on_send; RTCStatsMember 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(); @@ -261,7 +241,7 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats { RTCStatsMember port; RTCStatsMember protocol; RTCStatsMember relay_protocol; - // TODO(hbos): Support enum types? "RTCStatsMember"? + // Enum type RTCIceCandidateType. RTCStatsMember candidate_type; RTCStatsMember priority; RTCStatsMember url; @@ -300,8 +280,8 @@ class RTC_EXPORT RTCRemoteIceCandidateStats final const char* type() const override; }; -// https://w3c.github.io/webrtc-stats/#msstats-dict* -// TODO(hbos): Tracking bug crbug.com/660827 +// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats +// TODO(https://crbug.com/webrtc/14172): Deprecate and remove. class RTC_EXPORT RTCMediaStreamStats final : public RTCStats { public: WEBRTC_RTCSTATS_DECL(); @@ -315,8 +295,8 @@ class RTC_EXPORT RTCMediaStreamStats final : public RTCStats { RTCStatsMember> track_ids; }; -// https://w3c.github.io/webrtc-stats/#mststats-dict* -// TODO(hbos): Tracking bug crbug.com/659137 +// TODO(https://crbug.com/webrtc/14175): Deprecate and remove in favor of +// RTCMediaSourceStats/RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats. class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats { public: WEBRTC_RTCSTATS_DECL(); @@ -334,29 +314,20 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats { RTCStatsMember media_source_id; RTCStatsMember remote_source; RTCStatsMember ended; - // TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks. - // crbug.com/659137 + // TODO(https://crbug.com/webrtc/14173): Remove this obsolete metric. RTCStatsMember detached; - // See `RTCMediaStreamTrackKind` for valid values. + // Enum type RTCMediaStreamTrackKind. RTCStatsMember kind; RTCStatsMember jitter_buffer_delay; RTCStatsMember jitter_buffer_emitted_count; // Video-only members RTCStatsMember frame_width; RTCStatsMember frame_height; - // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137 - RTCStatsMember frames_per_second; RTCStatsMember frames_sent; RTCStatsMember huge_frames_sent; RTCStatsMember frames_received; RTCStatsMember frames_decoded; RTCStatsMember frames_dropped; - // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137 - RTCStatsMember frames_corrupted; - // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137 - RTCStatsMember partial_frames_lost; - // TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137 - RTCStatsMember full_frames_lost; // Audio-only members RTCStatsMember audio_level; // Receive-only RTCStatsMember total_audio_energy; // Receive-only @@ -370,7 +341,7 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats { RTCStatsMember inserted_samples_for_deceleration; RTCStatsMember removed_samples_for_acceleration; // Non-standard audio-only member - // TODO(kuddai): Add description to standard. crbug.com/webrtc/10042 + // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcaudioreceiverstats-jitterbufferflushes RTCNonStandardStatsMember jitter_buffer_flushes; RTCNonStandardStatsMember delayed_packet_outage_samples; RTCNonStandardStatsMember relative_packet_arrival_delay; @@ -380,14 +351,15 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats { // 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(titovartem) implement for video streams when will be requested. + // TODO(https://crbug.com/webrtc/14176): This should be moved to + // RTCInboundRtpStreamStats and it should be implemented for video as well. RTCNonStandardStatsMember jitter_buffer_target_delay; // TODO(henrik.lundin): Add description of the interruption metrics at - // https://github.com/henbos/webrtc-provisional-stats/issues/17 + // https://github.com/w3c/webrtc-provisional-stats/issues/17 RTCNonStandardStatsMember interruption_count; RTCNonStandardStatsMember total_interruption_duration; // Non-standard video-only members. - // https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict* + // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcvideoreceiverstats RTCNonStandardStatsMember freeze_count; RTCNonStandardStatsMember pause_count; RTCNonStandardStatsMember total_freezes_duration; @@ -411,7 +383,6 @@ class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats { }; // 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(); @@ -442,13 +413,6 @@ class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRTPStreamStats { 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 jitter; RTCStatsMember packets_lost; // Signed per RFC 3550 RTCStatsMember packets_discarded; @@ -475,8 +439,6 @@ class RTC_EXPORT RTCSentRtpStreamStats : public RTCRTPStreamStats { }; // 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: @@ -487,6 +449,8 @@ class RTC_EXPORT RTCInboundRTPStreamStats final RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other); ~RTCInboundRTPStreamStats() override; + // TODO(https://crbug.com/webrtc/14174): Implement trackIdentifier and kind. + RTCStatsMember remote_id; RTCStatsMember packets_received; RTCStatsMember fec_packets_received; @@ -505,48 +469,28 @@ class RTC_EXPORT RTCInboundRTPStreamStats final RTCStatsMember audio_level; RTCStatsMember total_audio_energy; RTCStatsMember total_samples_duration; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember round_trip_time; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember packets_repaired; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember burst_packets_lost; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember burst_packets_discarded; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember burst_loss_count; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember burst_discard_count; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember burst_loss_rate; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember burst_discard_rate; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember gap_loss_rate; - // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065 - RTCStatsMember gap_discard_rate; // Stats below are only implemented or defined for video. RTCStatsMember frames_received; RTCStatsMember frame_width; RTCStatsMember frame_height; - RTCStatsMember frame_bit_depth; RTCStatsMember frames_per_second; RTCStatsMember frames_decoded; RTCStatsMember key_frames_decoded; RTCStatsMember frames_dropped; RTCStatsMember total_decode_time; RTCStatsMember total_processing_delay; - // TODO(bugs.webrtc.org/13986): standardize + // TODO(https://crbug.com/webrtc/13986): standardize RTCNonStandardStatsMember total_assembly_time; RTCNonStandardStatsMember frames_assembled_from_multiple_packets; RTCStatsMember total_inter_frame_delay; RTCStatsMember total_squared_inter_frame_delay; - // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype + // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype RTCStatsMember content_type; - // TODO(asapersson): Currently only populated if audio/video sync is enabled. + // Only populated if audio/video sync is enabled. + // TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off? RTCStatsMember estimated_playout_timestamp; - // TODO(hbos): This is only implemented for video; implement it for audio as - // well. + // Only implemented for video. + // TODO(https://crbug.com/webrtc/14178): Also implement for audio. RTCStatsMember decoder_implementation; // FIR and PLI counts are only defined for |kind == "video"|. RTCStatsMember fir_count; @@ -559,8 +503,6 @@ class RTC_EXPORT RTCInboundRTPStreamStats final }; // 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(); @@ -597,10 +539,10 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats { RTCStatsMember> quality_limitation_durations; // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges RTCStatsMember quality_limitation_resolution_changes; - // https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype + // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype RTCStatsMember content_type; - // TODO(hbos): This is only implemented for video; implement it for audio as - // well. + // Only implemented for video. + // TODO(https://crbug.com/webrtc/14178): Implement for audio as well. RTCStatsMember encoder_implementation; // FIR and PLI counts are only defined for |kind == "video"|. RTCStatsMember fir_count; @@ -620,11 +562,6 @@ class RTC_EXPORT RTCRemoteInboundRtpStreamStats final 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 local_id; RTCStatsMember round_trip_time; RTCStatsMember fraction_lost; @@ -715,7 +652,7 @@ class RTC_EXPORT RTCTransportStats final : public RTCStats { RTCStatsMember bytes_received; RTCStatsMember packets_received; RTCStatsMember rtcp_transport_stats_id; - // TODO(hbos): Support enum types? "RTCStatsMember"? + // Enum type RTCDtlsTransportState. RTCStatsMember dtls_state; RTCStatsMember selected_candidate_pair_id; RTCStatsMember local_certificate_id; diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc index b5d86a9bf5..cc3e8e114b 100644 --- a/pc/rtc_stats_integrationtest.cc +++ b/pc/rtc_stats_integrationtest.cc @@ -486,7 +486,6 @@ class RTCStatsReportVerifier { verifier.TestMemberIsNonNegative(candidate_pair.priority); verifier.TestMemberIsDefined(candidate_pair.nominated); verifier.TestMemberIsDefined(candidate_pair.writable); - verifier.TestMemberIsUndefined(candidate_pair.readable); verifier.TestMemberIsNonNegative(candidate_pair.packets_sent); verifier.TestMemberIsNonNegative( candidate_pair.packets_discarded_on_send); @@ -514,13 +513,8 @@ class RTCStatsReportVerifier { verifier.TestMemberIsNonNegative( candidate_pair.responses_received); verifier.TestMemberIsNonNegative(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( candidate_pair.consent_requests_sent); - verifier.TestMemberIsUndefined(candidate_pair.consent_responses_received); - verifier.TestMemberIsUndefined(candidate_pair.consent_responses_sent); return verifier.ExpectAllMembersSuccessfullyTested(); } @@ -636,10 +630,6 @@ class RTCStatsReportVerifier { media_stream_track.frame_width); verifier.TestMemberIsNonNegative( 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); @@ -741,15 +731,11 @@ class RTCStatsReportVerifier { // 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); @@ -850,7 +836,6 @@ class RTCStatsReportVerifier { } else { verifier.TestMemberIsUndefined(inbound_stream.frames_per_second); } - verifier.TestMemberIsUndefined(inbound_stream.frame_bit_depth); verifier.TestMemberIsNonNegative( inbound_stream.jitter_buffer_delay); verifier.TestMemberIsNonNegative( @@ -894,16 +879,6 @@ class RTCStatsReportVerifier { 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 b0003c3952..a0873599e0 100644 --- a/stats/rtcstats_objects.cc +++ b/stats/rtcstats_objects.cc @@ -191,7 +191,6 @@ WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair", &priority, &nominated, &writable, - &readable, &packets_sent, &packets_received, &bytes_sent, @@ -204,12 +203,7 @@ WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair", &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 @@ -228,7 +222,6 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string&& id, priority("priority"), nominated("nominated"), writable("writable"), - readable("readable"), packets_sent("packetsSent"), packets_received("packetsReceived"), bytes_sent("bytesSent"), @@ -241,12 +234,7 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string&& id, 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") {} @@ -260,7 +248,6 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats( 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), @@ -273,12 +260,7 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats( 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) {} @@ -412,15 +394,11 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track", &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, @@ -465,15 +443,11 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id, 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"), @@ -521,15 +495,11 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats( 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), @@ -688,20 +658,9 @@ WEBRTC_RTCSTATS_IMPL( &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, @@ -747,20 +706,9 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id, 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"), @@ -803,20 +751,9 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats( 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),