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Move more non-standard metrics to inbound-rtp.

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They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.

To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.

Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
This commit is contained in:
Henrik Boström
2022-10-06 13:37:11 +02:00
committed by WebRTC LUCI CQ
parent 8ad5e393c4
commit 2fb83072db
6 changed files with 96 additions and 23 deletions
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11
api/stats/rtcstats_objects.h
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@ -346,13 +346,10 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
RTCStatsMember<uint64_t> concealment_events;
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
// Non-standard audio-only member
// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcaudioreceiverstats-jitterbufferflushes
// TODO(crbug.com/webrtc/14524): These metrics have been moved, delete them.
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
// https://github.com/w3c/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// Non-standard video-only members.
@ -503,6 +500,12 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
RTCStatsMember<uint32_t> pli_count;
RTCStatsMember<uint32_t> nack_count;
RTCStatsMember<uint64_t> qp_sum;
// Non-standard audio metrics.
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// The former googMinPlayoutDelayMs (in seconds).
RTCNonStandardStatsMember<double> min_playout_delay;