[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}
This commit is contained in:
Henrik Boström
committed by
WebRTC LUCI CQ
WebRTC LUCI CQ
parent
6daffe90e2
commit
626f87d905
@ -176,7 +176,7 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
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RTCStatsMember<std::string> label;
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RTCStatsMember<std::string> protocol;
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RTCStatsMember<int32_t> data_channel_identifier;
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// TODO(hbos): Support enum types? "RTCStatsMember<RTCDataChannelState>"?
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// Enum type RTCDataChannelState.
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RTCStatsMember<std::string> state;
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RTCStatsMember<uint32_t> messages_sent;
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RTCStatsMember<uint64_t> bytes_sent;
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@ -185,7 +185,6 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
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};
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// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
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// TODO(hbos): Tracking bug https://bugs.webrtc.org/7062
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class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -198,17 +197,16 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
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RTCStatsMember<std::string> transport_id;
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RTCStatsMember<std::string> local_candidate_id;
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RTCStatsMember<std::string> remote_candidate_id;
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// TODO(hbos): Support enum types?
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// "RTCStatsMember<RTCStatsIceCandidatePairState>"?
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// Enum type RTCStatsIceCandidatePairState.
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RTCStatsMember<std::string> state;
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// Obsolete: priority
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RTCStatsMember<uint64_t> priority;
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RTCStatsMember<bool> nominated;
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// TODO(hbos): Collect this the way the spec describes it. We have a value for
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// it but it is not spec-compliant. https://bugs.webrtc.org/7062
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// `writable` does not exist in the spec and old comments suggest it used to
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// exist but was incorrectly implemented.
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// TODO(https://crbug.com/webrtc/14171): Standardize and/or modify
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// implementation.
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RTCStatsMember<bool> writable;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<bool> readable;
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RTCStatsMember<uint64_t> packets_sent;
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RTCStatsMember<uint64_t> packets_received;
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RTCStatsMember<uint64_t> bytes_sent;
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@ -216,35 +214,17 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
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RTCStatsMember<double> total_round_trip_time;
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RTCStatsMember<double> current_round_trip_time;
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RTCStatsMember<double> available_outgoing_bitrate;
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// TODO(hbos): Populate this value. It is wired up and collected the same way
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// "VideoBwe.googAvailableReceiveBandwidth" is, but that value is always
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// undefined. https://bugs.webrtc.org/7062
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RTCStatsMember<double> available_incoming_bitrate;
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RTCStatsMember<uint64_t> requests_received;
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RTCStatsMember<uint64_t> requests_sent;
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RTCStatsMember<uint64_t> responses_received;
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RTCStatsMember<uint64_t> responses_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> retransmissions_received;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> retransmissions_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> consent_requests_received;
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RTCStatsMember<uint64_t> consent_requests_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> consent_responses_received;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> consent_responses_sent;
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RTCStatsMember<uint64_t> packets_discarded_on_send;
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RTCStatsMember<uint64_t> bytes_discarded_on_send;
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};
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// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
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// TODO(hbos): `RTCStatsCollector` only collects candidates that are part of
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// ice candidate pairs, but there could be candidates not paired with anything.
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// crbug.com/632723
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// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
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// them in the new PeerConnection::GetStats.
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class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -261,7 +241,7 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
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RTCStatsMember<int32_t> port;
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RTCStatsMember<std::string> protocol;
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RTCStatsMember<std::string> relay_protocol;
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// TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
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// Enum type RTCIceCandidateType.
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RTCStatsMember<std::string> candidate_type;
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RTCStatsMember<int32_t> priority;
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RTCStatsMember<std::string> url;
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@ -300,8 +280,8 @@ class RTC_EXPORT RTCRemoteIceCandidateStats final
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const char* type() const override;
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};
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// https://w3c.github.io/webrtc-stats/#msstats-dict*
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// TODO(hbos): Tracking bug crbug.com/660827
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamstats
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// TODO(https://crbug.com/webrtc/14172): Deprecate and remove.
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class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -315,8 +295,8 @@ class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
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RTCStatsMember<std::vector<std::string>> track_ids;
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};
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// https://w3c.github.io/webrtc-stats/#mststats-dict*
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// TODO(hbos): Tracking bug crbug.com/659137
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// TODO(https://crbug.com/webrtc/14175): Deprecate and remove in favor of
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// RTCMediaSourceStats/RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats.
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class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -334,29 +314,20 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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RTCStatsMember<std::string> media_source_id;
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RTCStatsMember<bool> remote_source;
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RTCStatsMember<bool> ended;
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// TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks.
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// crbug.com/659137
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// TODO(https://crbug.com/webrtc/14173): Remove this obsolete metric.
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RTCStatsMember<bool> detached;
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// See `RTCMediaStreamTrackKind` for valid values.
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// Enum type RTCMediaStreamTrackKind.
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RTCStatsMember<std::string> kind;
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RTCStatsMember<double> jitter_buffer_delay;
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RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
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// Video-only members
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RTCStatsMember<uint32_t> frame_width;
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RTCStatsMember<uint32_t> frame_height;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<double> frames_per_second;
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RTCStatsMember<uint32_t> frames_sent;
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RTCStatsMember<uint32_t> huge_frames_sent;
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RTCStatsMember<uint32_t> frames_received;
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RTCStatsMember<uint32_t> frames_decoded;
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RTCStatsMember<uint32_t> frames_dropped;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<uint32_t> frames_corrupted;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<uint32_t> partial_frames_lost;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<uint32_t> full_frames_lost;
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// Audio-only members
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RTCStatsMember<double> audio_level; // Receive-only
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RTCStatsMember<double> total_audio_energy; // Receive-only
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@ -370,7 +341,7 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
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RTCStatsMember<uint64_t> removed_samples_for_acceleration;
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// Non-standard audio-only member
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// TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcaudioreceiverstats-jitterbufferflushes
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RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
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RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
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RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
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@ -380,14 +351,15 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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// delay, in seconds, at the time that the sample was emitted from the jitter
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// buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
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// Currently it is implemented only for audio.
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// TODO(titovartem) implement for video streams when will be requested.
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// TODO(https://crbug.com/webrtc/14176): This should be moved to
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// RTCInboundRtpStreamStats and it should be implemented for video as well.
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RTCNonStandardStatsMember<double> jitter_buffer_target_delay;
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// TODO(henrik.lundin): Add description of the interruption metrics at
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// https://github.com/henbos/webrtc-provisional-stats/issues/17
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// https://github.com/w3c/webrtc-provisional-stats/issues/17
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RTCNonStandardStatsMember<uint32_t> interruption_count;
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RTCNonStandardStatsMember<double> total_interruption_duration;
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// Non-standard video-only members.
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// https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcvideoreceiverstats
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RTCNonStandardStatsMember<uint32_t> freeze_count;
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RTCNonStandardStatsMember<uint32_t> pause_count;
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RTCNonStandardStatsMember<double> total_freezes_duration;
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@ -411,7 +383,6 @@ class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats {
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};
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// https://w3c.github.io/webrtc-stats/#streamstats-dict*
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// TODO(hbos): Tracking bug crbug.com/657854
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class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -442,13 +413,6 @@ class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRTPStreamStats {
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RTCReceivedRtpStreamStats(const RTCReceivedRtpStreamStats& other);
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~RTCReceivedRtpStreamStats() override;
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// TODO(hbos) The following fields need to be added and migrated
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// both from RTCInboundRtpStreamStats and RTCRemoteInboundRtpStreamStats:
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// packetsReceived, packetsRepaired, burstPacketsLost,
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// burstPacketDiscarded, burstLossCount, burstDiscardCount, burstLossRate,
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// burstDiscardRate, gapLossRate, gapDiscardRate, framesDropped,
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// partialFramesLost, fullFramesLost
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// crbug.com/webrtc/12532
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RTCStatsMember<double> jitter;
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RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
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RTCStatsMember<uint64_t> packets_discarded;
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@ -475,8 +439,6 @@ class RTC_EXPORT RTCSentRtpStreamStats : public RTCRTPStreamStats {
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};
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// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
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// TODO(hbos): Support the remote case |is_remote = true|.
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// https://bugs.webrtc.org/7065
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class RTC_EXPORT RTCInboundRTPStreamStats final
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: public RTCReceivedRtpStreamStats {
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public:
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@ -487,6 +449,8 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
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RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
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~RTCInboundRTPStreamStats() override;
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// TODO(https://crbug.com/webrtc/14174): Implement trackIdentifier and kind.
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RTCStatsMember<std::string> remote_id;
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RTCStatsMember<uint32_t> packets_received;
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RTCStatsMember<uint64_t> fec_packets_received;
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@ -505,48 +469,28 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
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RTCStatsMember<double> audio_level;
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RTCStatsMember<double> total_audio_energy;
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RTCStatsMember<double> total_samples_duration;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<double> round_trip_time;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<uint32_t> packets_repaired;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<uint32_t> burst_packets_lost;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<uint32_t> burst_packets_discarded;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<uint32_t> burst_loss_count;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<uint32_t> burst_discard_count;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<double> burst_loss_rate;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<double> burst_discard_rate;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<double> gap_loss_rate;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
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RTCStatsMember<double> gap_discard_rate;
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// Stats below are only implemented or defined for video.
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RTCStatsMember<int32_t> frames_received;
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RTCStatsMember<uint32_t> frame_width;
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RTCStatsMember<uint32_t> frame_height;
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RTCStatsMember<uint32_t> frame_bit_depth;
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RTCStatsMember<double> frames_per_second;
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RTCStatsMember<uint32_t> frames_decoded;
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RTCStatsMember<uint32_t> key_frames_decoded;
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RTCStatsMember<uint32_t> frames_dropped;
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RTCStatsMember<double> total_decode_time;
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RTCStatsMember<double> total_processing_delay;
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// TODO(bugs.webrtc.org/13986): standardize
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// TODO(https://crbug.com/webrtc/13986): standardize
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RTCNonStandardStatsMember<double> total_assembly_time;
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RTCNonStandardStatsMember<uint32_t> frames_assembled_from_multiple_packets;
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RTCStatsMember<double> total_inter_frame_delay;
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RTCStatsMember<double> total_squared_inter_frame_delay;
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// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
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RTCStatsMember<std::string> content_type;
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// TODO(asapersson): Currently only populated if audio/video sync is enabled.
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// Only populated if audio/video sync is enabled.
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// TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off?
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RTCStatsMember<double> estimated_playout_timestamp;
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// TODO(hbos): This is only implemented for video; implement it for audio as
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// well.
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// Only implemented for video.
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// TODO(https://crbug.com/webrtc/14178): Also implement for audio.
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RTCStatsMember<std::string> decoder_implementation;
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// FIR and PLI counts are only defined for |kind == "video"|.
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RTCStatsMember<uint32_t> fir_count;
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@ -559,8 +503,6 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
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};
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// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
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// TODO(hbos): Support the remote case |is_remote = true|.
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// https://bugs.webrtc.org/7066
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class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -597,10 +539,10 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
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RTCStatsMember<std::map<std::string, double>> quality_limitation_durations;
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
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RTCStatsMember<uint32_t> quality_limitation_resolution_changes;
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// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
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RTCStatsMember<std::string> content_type;
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// TODO(hbos): This is only implemented for video; implement it for audio as
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// well.
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// Only implemented for video.
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// TODO(https://crbug.com/webrtc/14178): Implement for audio as well.
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RTCStatsMember<std::string> encoder_implementation;
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// FIR and PLI counts are only defined for |kind == "video"|.
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RTCStatsMember<uint32_t> fir_count;
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@ -620,11 +562,6 @@ class RTC_EXPORT RTCRemoteInboundRtpStreamStats final
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RTCRemoteInboundRtpStreamStats(const RTCRemoteInboundRtpStreamStats& other);
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~RTCRemoteInboundRtpStreamStats() override;
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// TODO(hbos): The following RTCReceivedRtpStreamStats metrics should also be
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// implemented: packetsReceived, packetsRepaired,
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// burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount,
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// burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate.
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// RTCRemoteInboundRtpStreamStats
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RTCStatsMember<std::string> local_id;
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RTCStatsMember<double> round_trip_time;
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RTCStatsMember<double> fraction_lost;
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@ -715,7 +652,7 @@ class RTC_EXPORT RTCTransportStats final : public RTCStats {
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RTCStatsMember<uint64_t> bytes_received;
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RTCStatsMember<uint64_t> packets_received;
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RTCStatsMember<std::string> rtcp_transport_stats_id;
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// TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
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// Enum type RTCDtlsTransportState.
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RTCStatsMember<std::string> dtls_state;
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RTCStatsMember<std::string> selected_candidate_pair_id;
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RTCStatsMember<std::string> local_certificate_id;
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