Revert "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs."
This reverts commit 626f87d90501fd8d7a4ea071686cd8befd0d430c. Reason for revert: Breaks one downstream project, will re-land after the dependency stops referencing an unimplemented RTT metric 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: I7e9ac60eb474b44fab678d4c08ddcae846ce456c No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265800 Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37206}
This commit is contained in:
Henrik Boström
committed by
WebRTC LUCI CQ
WebRTC LUCI CQ
parent
626f87d905
commit
378b1c6826
@ -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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// Enum type RTCDataChannelState.
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// TODO(hbos): Support enum types? "RTCStatsMember<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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@ -185,6 +185,7 @@ 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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@ -197,16 +198,17 @@ 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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// Enum type RTCStatsIceCandidatePairState.
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// TODO(hbos): Support enum types?
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// "RTCStatsMember<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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// `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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// 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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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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@ -214,17 +216,35 @@ 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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@ -241,7 +261,7 @@ 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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// Enum type RTCIceCandidateType.
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// TODO(hbos): Support enum types? "RTCStatsMember<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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@ -280,8 +300,8 @@ 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/#dom-rtcmediastreamstats
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// TODO(https://crbug.com/webrtc/14172): Deprecate and remove.
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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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class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -295,8 +315,8 @@ 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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// TODO(https://crbug.com/webrtc/14175): Deprecate and remove in favor of
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// RTCMediaSourceStats/RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats.
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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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class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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@ -314,20 +334,29 @@ 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(https://crbug.com/webrtc/14173): Remove this obsolete metric.
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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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RTCStatsMember<bool> detached;
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// Enum type RTCMediaStreamTrackKind.
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// See `RTCMediaStreamTrackKind` for valid values.
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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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@ -341,7 +370,7 @@ 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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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcaudioreceiverstats-jitterbufferflushes
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// TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
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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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@ -351,15 +380,14 @@ 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(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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// TODO(titovartem) implement for video streams when will be requested.
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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/w3c/webrtc-provisional-stats/issues/17
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// https://github.com/henbos/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://w3c.github.io/webrtc-provisional-stats/#dom-rtcvideoreceiverstats
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// https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
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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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@ -383,6 +411,7 @@ 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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@ -413,6 +442,13 @@ 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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@ -439,6 +475,8 @@ 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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@ -449,8 +487,6 @@ 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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@ -469,28 +505,48 @@ 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(https://crbug.com/webrtc/13986): standardize
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// TODO(bugs.webrtc.org/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://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
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// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
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RTCStatsMember<std::string> content_type;
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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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// TODO(asapersson): Currently only populated if audio/video sync is enabled.
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RTCStatsMember<double> estimated_playout_timestamp;
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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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// TODO(hbos): This is only implemented for video; implement it for audio as
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// well.
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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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@ -503,6 +559,8 @@ 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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@ -539,10 +597,10 @@ 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://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
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// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
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RTCStatsMember<std::string> content_type;
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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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// TODO(hbos): This is only implemented for video; implement it for audio as
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// 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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@ -562,6 +620,11 @@ 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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@ -652,7 +715,7 @@ 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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// Enum type RTCDtlsTransportState.
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// TODO(hbos): Support enum types? "RTCStatsMember<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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@ -486,6 +486,7 @@ class RTCStatsReportVerifier {
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verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.priority);
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verifier.TestMemberIsDefined(candidate_pair.nominated);
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verifier.TestMemberIsDefined(candidate_pair.writable);
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verifier.TestMemberIsUndefined(candidate_pair.readable);
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verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.packets_sent);
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verifier.TestMemberIsNonNegative<uint64_t>(
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candidate_pair.packets_discarded_on_send);
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@ -513,8 +514,13 @@ class RTCStatsReportVerifier {
|
||||
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 @@ class RTCStatsReportVerifier {
|
||||
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 @@ 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);
|
||||
@ -836,6 +850,7 @@ class RTCStatsReportVerifier {
|
||||
} 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 @@ 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);
|
||||
|
||||
@ -191,6 +191,7 @@ WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair",
|
||||
&priority,
|
||||
&nominated,
|
||||
&writable,
|
||||
&readable,
|
||||
&packets_sent,
|
||||
&packets_received,
|
||||
&bytes_sent,
|
||||
@ -203,7 +204,12 @@ 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
|
||||
@ -222,6 +228,7 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string&& id,
|
||||
priority("priority"),
|
||||
nominated("nominated"),
|
||||
writable("writable"),
|
||||
readable("readable"),
|
||||
packets_sent("packetsSent"),
|
||||
packets_received("packetsReceived"),
|
||||
bytes_sent("bytesSent"),
|
||||
@ -234,7 +241,12 @@ 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") {}
|
||||
|
||||
@ -248,6 +260,7 @@ 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),
|
||||
@ -260,7 +273,12 @@ 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) {}
|
||||
|
||||
@ -394,11 +412,15 @@ 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,
|
||||
@ -443,11 +465,15 @@ 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"),
|
||||
@ -495,11 +521,15 @@ 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),
|
||||
@ -658,9 +688,20 @@ 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,
|
||||
@ -706,9 +747,20 @@ 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"),
|
||||
@ -751,9 +803,20 @@ 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),
|
||||
|
||||
Reference in New Issue
Block a user