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<!--
Copyright 2020 The Chromium Authors. All rights reserved.
Use of this source code is governed by a BSD-style license that can be
found in the LICENSE file.
-->
<!--
This file is used to generate a comprehensive list of WebRTC histograms
along with a detailed description for each histogram.
For best practices on writing histogram descriptions, see
https://chromium.googlesource.com/chromium/src.git/+/HEAD/tools/metrics/histograms/README.md
Please send CLs to chromium-metrics-reviews@google.com rather than to specific
individuals. These CLs will be automatically reassigned to a reviewer within
about 5 minutes. This approach helps the metrics team to load-balance incoming
reviews. Googlers can read more about this at go/gwsq-gerrit.
-->
<histogram-configuration>
<histograms>
<variants name="IPProtocolType">
<variant name="_TCP" summary=""/>
<variant name="_UDP" summary=""/>
</variants>
<variants name="NatType">
<variant name=".NoNAT" summary=""/>
<variant name=".NonSymNAT" summary=""/>
<variant name=".SymNAT" summary=""/>
<variant name=".UnknownNAT" summary=""/>
</variants>
<variants name="ScreenshareLayerStats">
<variant name=".FrameRate"
summary="Frames per second sent, in fps. The value is reported when a
stream is removed and is calculated as the total number of
frames in this layer, divided by the duration of the call."/>
<variant name=".Qp"
summary="Average quantizer (qp) of frames sent. The value is reported
when a stream is removed and is calculated, for this layer, as
the sum of all qp values divided the number of frames."/>
<variant name=".TargetBitrate"
summary="Average target bitrate in kbps. The value is reported when a
stream is removed and is calculated as the sum of all target
bitrates for this layer (sampled after frame has been encoded)
divided by the total number of frames for this layer."/>
</variants>
<variants name="WebRTCEchoCancellerEstimate">
<variant name=".Max" summary="The maximum over the time interval"/>
<variant name=".Min" summary="The minimum over the time interval"/>
<variant name=".Value"
summary="The last estimated value of the time interval"/>
</variants>
<variants name="WebRTCMediaType">
<variant name=".Audio" summary="Audio"/>
<variant name=".Data" summary="Data"/>
<variant name=".Video" summary="Video"/>
</variants>
<variants name="WebRTCVideoExperimentGroupId">
<variant name=".ExperimentGroup0" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".ExperimentGroup1" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".ExperimentGroup2" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".ExperimentGroup3" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".ExperimentGroup4" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".S0" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".S1" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
<variant name=".S2" summary="">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
</variant>
</variants>
<histogram name="WebRTC.AecFilterHasDivergence" units="%" expires_after="M85">
<owner>grunell@chromium.org</owner>
<owner>minyue@chromium.org</owner>
<summary>
The percentage of measurement periods during a capture stream's lifetime
when the echo canceler's filter is considered to have diverged at least
once. A diverged filter could mean that there was echo och ducking
experienced. The measurement period size is fixed and in the order of one
second. The first measurement period is larger, typically 2-3 seconds.
Capture streams with shorter lifetimes (i.e. no data available) are not
recorded here.
</summary>
</histogram>
<histogram name="WebRTC.ApplicationMaxConsecutiveBytesDiscard.v2" units="units"
expires_after="M77">
<owner>guoweis@chromium.org</owner>
<summary>
The maximum consecutive discarded bytes caused by not enough buffer
available in WebRTC's socket implementation. This happens when WebRTC
IpcPacketSocket's throttling mechanism kicks in. The maximum bucket is
expanded from previous version to provide more insight when upper layer
feeds a lot of packets.
</summary>
</histogram>
<histogram name="WebRTC.ApplicationPercentPacketsDiscarded" units="%"
expires_after="M81">
<owner>guoweis@chromium.org</owner>
<summary>
The percentage of packets discarded by WebRTC's socket layer due to
EWOULDBLOCKs when WebRTC IpcPacketSocket's throttling mechanism kicks in.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc.ClippingPredictor.F1Score" units="%"
expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<owner>silen@chromium.org</owner>
<summary>
Logs the F1 score for the clipping predictor used in AgcManagerDirect. A log
call is made every 30 seconds during an active WebRTC call using the analog
gain controller and the clipping predictor feature.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc.ClippingPredictor.PredictionInterval"
units="frames" expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<owner>silen@chromium.org</owner>
<summary>
Logs the time elapsed between a clipping prediction and a matched clipping
detection for the clipping predictor used in AgcManagerDirect. A log call is
made every time a predicted clipping event is matched (theoretical upper
bound: one call every 10 ms) during an active WebRTC call using the analog
gain controller and the clipping predictor feature.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc.DigitalGainApplied" units="dB"
expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
Logs adaptive digital compression gain that is applied by AgcManagerDirect.
A log call is made once per second. The compression gain is applied to the
microphone signal at the end of the processing chain.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc.DigitalGainUpdated" units="dB"
expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
Logs adaptive digital compression gain that is applied by AgcManagerDirect.
A log call is made every time the gain changes. The compression gain is
applied to the microphone signal at the end of the processing chain.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc2.DigitalGainApplied" units="dB"
expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
Logs adaptive digital compression gain that is applied by
AdaptiveDigitalGainApplier in GainController2. A log call is made once per
second. The compression gain is applied to the microphone signal at the end
of the processing chain.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc2.EstimatedNoiseLevel" units="dBFS (negated)"
expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
This histogram reports the noise level estimation done in GainController2. A
value is reported every second. The unit is inverted dBFS. The scale goes
from 0 (very loud noise) to 100 (very faint noise).
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.Identity"
units="seconds" expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Identity' region. In this region no attenuating gain is
applied. This metric is logged from the Fixed Digital limiter in
GainController2.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.Knee"
units="seconds" expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Knee' region. In this region some attenuating gain is applied.
This metric is logged from the Fixed Digital limiter in GainController2.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.Limiter"
units="seconds" expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Limiter' region. In this region some more attenuating gain is
applied. This metric is logged from the Fixed Digital limiter in
GainController2.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.Saturation"
units="seconds" expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Saturation' region. In this region much attenuating gain is
applied. This metric is logged from the Fixed Digital limiter in
GainController2.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AgcClippingAdjustmentAllowed" enum="Boolean"
expires_after="2021-12-31">
<owner>hlundin@chromium.org</owner>
<summary>
The automatic gain control (AGC) in WebRTC tries to adjust the microphone
gain to maintain a strong audio level, but without clipping (saturation).
The histogram will log a value every time input clipping is detected. The
value is a boolean, with &quot;true&quot; meaning that the gain was in fact
adjusted in response to the detected clipping, and &quot;false&quot; meaning
that adjustment was not allowed due to limiting boundaries in the algorithm.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AgcSetLevel" units="level"
expires_after="2021-12-31">
<owner>hlundin@chromium.org</owner>
<summary>
The automatic gain control (AGC) in WebRTC tries to adjust the microphone
gain to maintain a strong audio level, but without clipping (saturation).
The histogram will log a new value every time the AGC changes the target
level.
</summary>
</histogram>
<histogram name="WebRTC.Audio.ApmCaptureInputLevelAverageRms"
units="dBFS (negated)" expires_after="2021-12-31">
<owner>hlundin@chromium.org</owner>
<summary>
This histogram reports the average RMS of the signal coming in to WebRTC's
Audio Processing Module, prior to any WebRTC processing. A new value is
reported every 10 seconds, and the average is over the latest interval. The
metric is negated dBFS, meaning that 0 is a full-scale signal, while 127
corresponds to -127 dBFS (very faint).
</summary>
</histogram>
<histogram name="WebRTC.Audio.ApmCaptureInputLevelPeakRms"
units="dBFS (negated)" expires_after="2021-12-31">
<owner>hlundin@chromium.org</owner>
<summary>
This histogram reports the peak RMS of the signal coming in to WebRTC's
Audio Processing Module, prior to any WebRTC processing. A new value is
reported every 10 seconds, and the peak is the RMS of the strongest 10 ms
block over the latest interval. The metric is negated dBFS, meaning that 0
is a full-scale signal, while 127 corresponds to -127 dBFS (very faint).
</summary>
</histogram>
<histogram name="WebRTC.Audio.ApmCaptureOutputLevelAverageRms"
units="dBFS (negated)" expires_after="2021-12-31">
<owner>peah@chromium.org</owner>
<summary>
This histogram reports the average RMS of the signal in the output of
WebRTC's Audio Processing Module, after all audio WebRTC processing. A new
value is reported every 10 seconds, and the average is over the latest
interval. The metric is negated dBFS, meaning that 0 is a full-scale signal,
while 127 corresponds to -127 dBFS (very faint).
</summary>
</histogram>
<histogram name="WebRTC.Audio.ApmCaptureOutputLevelPeakRms"
units="dBFS (negated)" expires_after="2021-12-31">
<owner>peah@chromium.org</owner>
<summary>
This histogram reports the peak RMS of the signal in the output of WebRTC's
Audio Processing Module, after all WebRTC audio processing. A new value is
reported every 10 seconds, and the peak is the RMS of the strongest 10 ms
block over the latest interval. The metric is negated dBFS, meaning that 0
is a full-scale signal, while 127 corresponds to -127 dBFS (very faint).
</summary>
</histogram>
<histogram name="WebRTC.Audio.ApmRuntimeSettingCannotEnqueue" units="counts"
expires_after="2021-12-31">
<owner>alessiob@chromium.org</owner>
<owner>peah@chromium.org</owner>
<summary>
Counts occurrences of if WebRTC's Audio Processing Module cannot enqueue a
runtime setting.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioInterruptionMs" units="ms"
expires_after="2021-11-23">
<owner>hlundin@chromium.org</owner>
<owner>ivoc@chromium.org</owner>
<summary>
Measures the duration of each audio interruption event. An audio
interruption is defined as a loss concealment (a.k.a. expand) event that
lasts more than 150 milliseconds. The metric registers each of these events.
This gives an indication of the length and prevalence of severe network
events, which are likely to be detrimental to the audio quality.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.FixedDigitalGainCurveRegion.Identity"
units="seconds" expires_after="2019-09-01">
<owner>aleloi@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Identity' region. In this region no attenuating gain is
applied. This metric is logged from the Fixed Digital limiter in the Audio
Mixer.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.FixedDigitalGainCurveRegion.Knee"
units="seconds" expires_after="2019-09-01">
<owner>aleloi@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Knee' region. In this region some attenuating gain is applied.
This metric is logged from the Fixed Digital limiter in the Audio Mixer.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.FixedDigitalGainCurveRegion.Limiter"
units="seconds" expires_after="2019-09-01">
<owner>aleloi@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Limiter' region. In this region some more attenuating gain is
applied. This metric is logged from the Fixed Digital limiter in the Audio
Mixer.
</summary>
</histogram>
<histogram
name="WebRTC.Audio.AudioMixer.FixedDigitalGainCurveRegion.Saturation"
units="seconds" expires_after="2019-09-01">
<owner>aleloi@chromium.org</owner>
<summary>
The Fixed-Digital part of the AGC protects from saturation by reducing the
level of too loud signals. This metric shows how long the level estimate
stays in the 'Saturation' region. In this region much attenuating gain is
applied. This metric is logged from the Fixed Digital limiter in the Audio
Mixer.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.MixingRate" enum="WebRtcNativeRate"
expires_after="M81">
<owner>aleloi@chromium.org</owner>
<summary>
Reports the sampling rate at which audio mixing is done. Lower rates result
in faster processing, higher rates can give higher quality.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.NumIncomingActiveStreams"
units="count" expires_after="M77">
<obsolete>
Removed as of 2021-05-10.
</obsolete>
<owner>aleloi@chromium.org</owner>
<summary>
Reports the number of active incoming streams in the WebRTC audio mixer. An
incoming stream is active if it is not muted. When multiple streams are
active, adding them can result in saturation and limiting is needed. Logged
every second during a WebRTC call.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.NumIncomingActiveStreams2"
units="streams" expires_after="2022-04-30">
<owner>alessiob@chromium.org</owner>
<owner>webrtc-audio@google.com</owner>
<summary>
Reports the number of active incoming streams in the WebRTC audio mixer. An
incoming stream is active if it is not muted. When multiple streams are
active, adding them can result in saturation and limiting is needed. Logged
every second during a WebRTC call.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AudioMixer.NumIncomingStreams" units="streams"
expires_after="2022-04-30">
<owner>alessiob@chromium.org</owner>
<owner>webrtc-audio@google.com</owner>
<summary>
Reports the number of streams in the WebRTC audio mixer. The mixer queries
all streams for audio and preferred sampling rate. The number of streams
affects mixing decisions. Whether to mix in floating points or use a limiter
depends on the number of incoming streams.
</summary>
</histogram>
<histogram name="WebRTC.Audio.AverageExcessBufferDelayMs" units="ms"
expires_after="2021-12-19">
<owner>hlundin@chromium.org</owner>
<summary>
Measures the average waiting time in the buffer for each packet. The waiting
time is the time elapsed from the packet arrives until it is decoded. The
metric is calculated as the average over one minute, and is logged at the
end of each such interval. A well tuned target buffer level should lead to a
low value.
</summary>
</histogram>
<histogram name="WebRTC.Audio.DelayedPacketOutageEventMs" units="ms"
expires_after="2021-11-14">
<owner>hlundin@chromium.org</owner>
<summary>
Measures the duration of each packet loss concealment (a.k.a. expand) event
that is not followed by a merge operation. The outage is measured in
milliseconds, with a range between 0 and 2000 ms. This gives an indication
of how well the jitter buffer's level adaptation is working. If the chosen
target level is too low, this value will increase.
</summary>
</histogram>
<histogram name="WebRTC.Audio.DelayedPacketOutageEventsPerMinute"
units="events/minute" expires_after="2022-01-02">
<owner>hlundin@chromium.org</owner>
<summary>
Counts the number of delayed packet outage events per minute. The range is
between 0 and 100 (corresponds to more 1.6 events per second). See
WebRTC.Audio.DelayedPacketOutageEventMs for the definition of a delayed
packet outage event, and the interpretation of such events.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ActiveRender" enum="Boolean"
expires_after="2020-12-01">
<obsolete>
No longer in use. Removed on 2020-10-19.
</obsolete>
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs a value indicating whether the WebRTC echo canceller
detects that there is active content in the render signal. A new value is
logged every 10 seconds and the logged value is averaged over this period.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.BufferDelay" units="Blocks"
expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the applied render buffer delay used in the WebRTC echo
canceller. A new value is logged every 10 seconds and the logged value
constitutes the current buffer delay at the time when the value is logged.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.CaptureSaturation" enum="Boolean"
expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs a value every time the WebRTC echo canceller has
detected saturation in the capture signal. A new value is logged every 10
seconds and the logged value indicates whether the capture signal has been
saturated during the last 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.Clockdrift" enum="ClockdriftLevel"
expires_after="2021-10-19">
<owner>gustaf@chromium.org</owner>
<owner>peah@chromium.org</owner>
<summary>
This histogram logs whether clockdrift is detected in the WebRTC echo
canceller. A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.DelayChanges"
enum="WebRTCEventFrequency" expires_after="2021-12-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the frequency of echo path delay changes that are
detected by the delay estimator in the WebRTC echo canceller. A new value is
logged every 10 seconds and the logged value indicates how frequent delay
changes have been during the last 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.EchoPathDelay" units="Blocks"
expires_after="2021-12-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the estimated echo path delay in 64 sample blocks as
seen by the delay estimator in the WebRTC echo canceller. A new value is
logged every 10 seconds and the logged value is the estimate of the delay of
the echo path at the time when the value is logged.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ErlBand0" units="dB (shifted)"
expires_after="2020-12-01">
<obsolete>
No longer in use, EchoCanceller.Erl considered sufficient. Removed on
2020-10-19.
</obsolete>
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the echo return loss achieved by the WebRTC echo
canceller in the lower 4 kHz. A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ErlBand1" units="dB (shifted)"
expires_after="2020-12-01">
<obsolete>
No longer in use, EchoCanceller.Erl considered sufficient. Removed on
2020-10-19.
</obsolete>
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the echo return loss achieved by the WebRTC echo
canceller between 4 and 8 kHz. A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ErleBand0" units="dB (shifted)"
expires_after="2020-12-01">
<obsolete>
No longer in use, EchoCanceller.Erle considered sufficient. Removed on
2020-10-19.
</obsolete>
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the echo return loss enhancement achieved by the WebRTC
echo canceller in the lower 4 kHz. A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ErleBand1" units="dB (shifted)"
expires_after="2020-12-01">
<obsolete>
No longer in use, EchoCanceller.Erle considered sufficient. Removed on
2020-10-19.
</obsolete>
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the echo return loss enhancement achieved by the WebRTC
echo canceller between 4 and 8 kHz. A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.Erle{WebRTCEchoCancellerEstimate}"
units="dB" expires_after="2021-10-19">
<owner>gustaf@chromium.org</owner>
<owner>peah@chromium.org</owner>
<summary>
This histogram logs the echo return loss enhancement achieved by the WebRTC
echo canceller as described in ITU G.168. When the echo canceller is being
used, one value is logged every 10 seconds per ongoing WebRTC call.
{WebRTCEchoCancellerEstimate}
</summary>
<token key="WebRTCEchoCancellerEstimate"
variants="WebRTCEchoCancellerEstimate">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.Erl{WebRTCEchoCancellerEstimate}"
units="dB (shifted)" expires_after="2021-10-19">
<owner>gustaf@chromium.org</owner>
<owner>peah@chromium.org</owner>
<summary>
This histogram logs the echo return loss achieved by the WebRTC echo
canceller as described in ITU G.168. When the echo canceller is being used,
one value is logged every 10 seconds per ongoing WebRTC call.
{WebRTCEchoCancellerEstimate}
</summary>
<token key="WebRTCEchoCancellerEstimate"
variants="WebRTCEchoCancellerEstimate">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.FilterDelay" units="Blocks"
expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the estimated echo path delay in 64 sample blocks as
seen by the linear filter delay in the WebRTC echo canceller. The value 0
means that no delay could be estimated from the linear filter, otherwise the
logged delay corresponds to the actual delay + 1. A new value is logged
every 10 seconds and the logged value is the estimate of the filter delay at
the time when the value is logged.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.MaxCaptureJitter"
units="frames (10 ms)" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>gustaf@chromium.org</owner>
<summary>
This histogram logs the observed maximum number of capture API calls in a
row in the unit of frames (10 ms). A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.MaxRenderJitter"
units="frames (10 ms)" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>gustaf@chromium.org</owner>
<summary>
This histogram logs the observed maximum number of render API calls in a row
in the unit of frames (10 ms). A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.MaxSkewShiftCount"
units="events/minute" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the number of times per minute that the WebRTC echo
canceller detects a shift in the skew between the total number of render and
capture calls. The metric is reported once per minute and the report is done
also for the case of 0 detected skew shifts have been detected. The amount
of skew shifts per minute is capped to 20. The metric is reported separately
for each render process. The metric is only reported for render processes
where the WebRTC echo canceller is active in and there is both audio being
played out and captured simultaneously. The reporting is only done when the
browser is awake.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.MinCaptureJitter"
units="frames (10 ms)" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>gustaf@chromium.org</owner>
<summary>
This histogram logs the observed minimum number of capture API calls in a
row in the unit of frames (10 ms). A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.MinRenderJitter"
units="frames (10 ms)" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>gustaf@chromium.org</owner>
<summary>
This histogram logs the observed minimum number of render API calls in a row
in the unit of frames (10 ms). A new value is logged every 10 seconds.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ModelBasedAecFeasible"
enum="Boolean" expires_after="2020-12-01">
<obsolete>
No longer reported. Removed on 2020-10-19.
</obsolete>
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs a value every time the WebRTC echo canceller deems that
echo path is possible to model using any of the the echo canceller echo path
models. A new value is logged every 10 seconds and the logged value is the
feasibility assessment at the time when the value is logged.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.ReliableDelayEstimates"
enum="WebRTCAecDelayEstimateReliability" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the assessed reliability of the delay estimates produced
by the delay estimator in the WebRTC echo canceller. A new value is logged
every 10 seconds and the logged value is a meausure based on how often
during that period the delay estimate was reliable.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.RenderOverruns"
enum="WebRTCEventFrequency" expires_after="2021-12-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the frequency of overruns in the render buffer of the
WebRTC echo canceller. A new value is logged every 10 seconds and the logged
value is a meausure that indicates how often overruns occurred.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.RenderUnderruns"
enum="WebRTCEventFrequency" expires_after="2021-12-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs the frequency of underruns in the render buffer of the
WebRTC echo canceller. A new value is logged every 10 seconds and the logged
value is a meausure that indicates how often underruns occurred.
</summary>
</histogram>
<histogram name="WebRTC.Audio.EchoCanceller.UsableLinearEstimate"
enum="Boolean" expires_after="2021-10-19">
<owner>peah@chromium.org</owner>
<owner>saza@chromium.org</owner>
<summary>
This histogram logs a value every time the WebRTC echo canceller deems that
the linear filter in the echo canceller is sufficiently well adapted to be
usable. A new value is logged every 10 seconds and the logged value is the
assessment of whether the filter is usable at the time when the value is
logged.
</summary>
</histogram>
<histogram name="WebRTC.Audio.Encoder.CodecType" enum="WebRtcAudioCodecs"
expires_after="2021-11-07">
<owner>aleloi@chromium.org</owner>
<summary>
Histogram of audio codec usage. Every sample corresponds to 5 seconds of
encoding with that codec.
</summary>
</histogram>
<histogram name="WebRTC.Audio.ExpandRatePercent" units="%"
expires_after="2021-11-14">
<owner>hlundin@chromium.org</owner>
<summary>
Measures the expand rate for an incoming WebRTC audio stream. The expand
rate is the fraction of samples that are generated through loss-concealemnt
algorithms instead of being decoded from incoming media packets. The metric
is calculated as the percent over a 10 second internval, and is logged at
the end of each such interval.
</summary>
</histogram>
<histogram name="WebRTC.Audio.JitterBufferFullPerMinute" units="events/minute"
expires_after="2020-08-23">
<owner>minyue@chromium.org</owner>
<summary>
Frequency that audio packets hits the capacity of WebRTC jitter buffer. A
larger value indicates that the capacity is not big enough, and/or audio
packets are not processed quickly enough. The metric is recorded every
minute.
</summary>
</histogram>
<histogram name="WebRTC.Audio.ReceiverDeviceDelayMs" units="ms"
expires_after="2022-01-02">
<owner>hlundin@chromium.org</owner>
<summary>
The sound card's buffering delay for the receiving side. Sampled once every
10 ms when WebRTC audio is playing.
</summary>
</histogram>
<histogram name="WebRTC.Audio.ReceiverJitterBufferDelayMs" units="ms"
expires_after="2022-01-02">
<owner>hlundin@chromium.org</owner>
<summary>
The jitter buffer delay for the receiving side. Sampled once every 10 ms
when WebRTC audio is playing.
</summary>
</histogram>
<histogram name="WebRTC.Audio.ResidualEchoDetector.EchoLikelihood" units="%"
expires_after="2021-05-09">
<obsolete>
No longer reported. Removed in M84.
</obsolete>
<owner>hlundin@chromium.org</owner>
<owner>ivoc@chromium.org</owner>
<summary>
The estimated likelihood percentage of echo as detected by the residual echo
detector. The residual echo detector can be used to detect cases where the
AEC (hardware or software) is not functioning properly. The detector can be
non-causal and operates on larger timescales with more delay than the
regular AEC.
</summary>
</histogram>
<histogram name="WebRTC.Audio.SpeechExpandRatePercent" units="%"
expires_after="2021-11-28">
<owner>hlundin@chromium.org</owner>
<summary>
Measures the audible expand rate for an incoming WebRTC audio stream. The
metric is very similar to WebRTC.Audio.ExpandRate, with the difference that
this metric only reports loss-concealement that generates audible samples.
The metric is calculated as the percent over a 10 second internval, and is
logged at the end of each such interval.
</summary>
</histogram>
<histogram name="WebRTC.Audio.TargetBitrateInKbps" units="kbps"
expires_after="2021-11-07">
<owner>hlundin@chromium.org</owner>
<summary>
The target bitrate in kbps that the audio codec should try to produce on
average. Sampled every time the rate is updated.
</summary>
</histogram>
<histogram name="WebRTC.Audio.TargetJitterBufferDelayMs" units="ms"
expires_after="2021-12-26">
<owner>hlundin@chromium.org</owner>
<summary>
The target jitter buffer delay for the receiving side. Sampled once every 10
ms per WebRTC receive stream when WebRTC audio is playing.
</summary>
</histogram>
<histogram name="WebRTC.AudioInputChannelLayout" enum="ChannelLayout"
expires_after="2020-11-01">
<owner>henrika@chromium.org</owner>
<owner>webrtc-audio@google.com</owner>
<summary>Audio input channel layout in WebRTC.</summary>
</histogram>
<histogram name="WebRTC.AudioInputSampleRate" enum="AudioSampleRate"
expires_after="2020-11-29">
<owner>henrika@chromium.org</owner>
<summary>Audio input sample rate for WebRTC (in Hz).</summary>
</histogram>
<histogram name="WebRTC.AudioInputSampleRateUnexpected" units="Hz"
expires_after="2021-04-30">
<owner>henrika@chromium.org</owner>
<owner>webrtc-audio@google.com</owner>
<summary>
Audio input sample rate for WebRTC (atypical values, in Hz).
</summary>
</histogram>
<histogram name="WebRTC.AudioOutputSampleRate" enum="AudioSampleRate"
expires_after="2021-11-07">
<owner>henrika@chromium.org</owner>
<owner>webrtc-audio@google.com</owner>
<summary>Audio output sample rate for WebRTC (in Hz).</summary>
</histogram>
<histogram name="WebRTC.AudioOutputSampleRateUnexpected" units="Hz"
expires_after="2021-04-30">
<owner>henrika@chromium.org</owner>
<owner>webrtc-audio@google.com</owner>
<summary>
Audio output sample rate for WebRTC (atypical values, in Hz).
</summary>
</histogram>
<histogram name="WebRTC.BWE.InitialBandwidthEstimate" units="kbps"
expires_after="2021-11-07">
<owner>holmer@chromium.org</owner>
<summary>The bandwidth estimate 2 seconds into a WebRTC call.</summary>
</histogram>
<histogram name="WebRTC.BWE.InitiallyLostPackets" units="packets"
expires_after="2021-12-19">
<owner>holmer@chromium.org</owner>
<summary>
The number of video packets lost durig the first 2 seconds in a WebRTC call.
</summary>
</histogram>
<histogram name="WebRTC.BWE.InitialRtt" units="ms" expires_after="2021-12-19">
<owner>holmer@chromium.org</owner>
<summary>
The round-trip time as measured 2 seconds into a WebRTC call.
</summary>
</histogram>
<histogram name="WebRTC.BWE.InitialVsConvergedDiff" units="kbps"
expires_after="2022-12-27">
<owner>holmer@chromium.org</owner>
<summary>
The difference between the bandwidth estimate at 2 seconds and 20 seconds
into a WebRTC call, with a min at 0, which is supposed to capture the how
much the initial bandwidth estimate overshot the actual bandwidth available.
</summary>
</histogram>
<histogram name="WebRTC.BWE.MidCallProbing.Initiated" units="kbps"
expires_after="2020-02-16">
<owner>philipel@chromium.org</owner>
<summary>
The bitrate that will be probed, triggered by an update to the max
configured bitrate.
</summary>
</histogram>
<histogram name="WebRTC.BWE.MidCallProbing.ProbedKbps" units="kbps"
expires_after="2020-03-01">
<owner>philipel@chromium.org</owner>
<summary>
The resulting bitrate probed, triggered by an update to the max configured
bitrate.
</summary>
</histogram>
<histogram name="WebRTC.BWE.MidCallProbing.Success" units="kbps"
expires_after="2020-02-16">
<owner>philipel@chromium.org</owner>
<summary>
A successful probing attempt for a given bitrate, triggered by an update to
the max configured bitrate. NOTE! This is not the resulting bitrate from a
probing attempt, see WebRTC.BWE.MidCallProbing.ProbedKbps.
</summary>
</histogram>
<histogram name="WebRTC.BWE.Probing.ProbeClusterSizeInBytes" units="bytes"
expires_after="2021-11-21">
<owner>jonasolsson@chromium.org</owner>
<owner>crodbro@chromium.org</owner>
<summary>
The size in bytes of the probe cluster. Counted for each cluster once all
its packets have been sent.
</summary>
</histogram>
<histogram name="WebRTC.BWE.Probing.ProbesPerCluster" units="units"
expires_after="2021-11-21">
<owner>jonasolsson@chromium.org</owner>
<owner>crodbro@chromium.org</owner>
<summary>
The size in packets of the probe cluster. Counted for each cluster once all
its packets have been sent.
</summary>
</histogram>
<histogram name="WebRTC.BWE.Probing.TimePerProbeCluster" units="ms"
expires_after="2021-11-14">
<owner>jonasolsson@chromium.org</owner>
<owner>crodbro@chromium.org</owner>
<summary>
The time from sending the first to the last packet of the probe cluster.
Counted for each cluster once all its packets have been sent.
</summary>
</histogram>
<histogram name="WebRTC.BWE.Probing.TotalFailedProbeClusters" units="units"
expires_after="2020-10-01">
<owner>jonasolsson@chromium.org</owner>
<owner>crodbro@chromium.org</owner>
<summary>
Counts the amount of probe clusters that timed out. Counted when the prober
is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.BWE.Probing.TotalProbeClustersRequested" units="units"
expires_after="2021-11-14">
<owner>jonasolsson@chromium.org</owner>
<owner>crodbro@chromium.org</owner>
<summary>
Counts the amount of probe clusters that the bitrate prober created. Counted
when the prober is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.BWE.RampUpTimeTo1000kbpsInMs" units="ms"
expires_after="2022-10-25">
<owner>holmer@chromium.org</owner>
<summary>
The time it takes the estimated bandwidth to reach 1000 kbps from the first
RTCP packet received. Used to measure the bandwidth ramp-up time.
</summary>
</histogram>
<histogram name="WebRTC.BWE.RampUpTimeTo2000kbpsInMs" units="ms"
expires_after="2022-02-28">
<owner>holmer@chromium.org</owner>
<summary>
The time it takes the estimated bandwidth to reach 2000 kbps from the first
RTCP packet received. Used to measure the bandwidth ramp-up time.
</summary>
</histogram>
<histogram name="WebRTC.BWE.RampUpTimeTo500kbpsInMs" units="ms"
expires_after="2022-12-27">
<owner>holmer@chromium.org</owner>
<summary>
The time it takes the estimated bandwidth to reach 500 kbps from the first
RTCP packet received. Used to measure the bandwidth ramp-up time.
</summary>
</histogram>
<histogram name="WebRTC.BWE.Types" enum="WebRtcBweType"
expires_after="2022-10-11">
<owner>holmer@chromium.org</owner>
<summary>
The bandwidth estimation used in WebRTC calls. Records whether the BWE is
running on the sender or the receiver and what BWE related RTP header
extensions are in use.
</summary>
</histogram>
<histogram name="WebRTC.Call.AudioBitrateReceivedInKbps" units="kbps"
expires_after="2022-02-21">
<owner>holmer@chromium.org</owner>
<summary>
Average audio bitrate received during a call, counted from first packet
received until Call instance is destroyed. Only mesured for calls that are
at least 10 seconds long.
</summary>
</histogram>
<histogram name="WebRTC.Call.BitrateReceivedInKbps" units="kbps"
expires_after="2022-02-21">
<owner>holmer@chromium.org</owner>
<summary>
Average total bitrate received during a call (audio + video + RTCP), counted
from first packet received until Call instance is destroyed. Only mesured
for calls that are at least 10 seconds long.
</summary>
</histogram>
<histogram name="WebRTC.Call.EstimatedSendBitrateInKbps" units="kbps"
expires_after="2022-04-11">
<owner>holmer@chromium.org</owner>
<summary>
Average estimated send bitrate during a call, counted from first packet sent
until Call instance is destroyed. Only mesured for calls that are at least
10 seconds long.
</summary>
</histogram>
<histogram name="WebRTC.Call.LifetimeInSeconds" units="seconds"
expires_after="2022-02-21">
<owner>asapersson@chromium.org</owner>
<summary>
The lifetime of a call. Recorded when a Call instance is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.Call.PacerBitrateInKbps" units="kbps"
expires_after="2022-02-21">
<owner>holmer@chromium.org</owner>
<summary>
Average pacer bitrate during a call, counted from first packet sent until
Call instance is destroyed. Only mesured for calls that are at least 10
seconds long.
</summary>
</histogram>
<histogram name="WebRTC.Call.RtcpBitrateReceivedInBps" units="bits/s"
expires_after="2022-02-21">
<owner>holmer@chromium.org</owner>
<summary>
Average RTCP bitrate received during a call, counted from first packet
received until Call instance is destroyed. Only mesured for calls that are
at least 10 seconds long.
</summary>
</histogram>
<histogram name="WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds" units="s"
expires_after="2021-12-05">
<owner>saza@chromium.org</owner>
<summary>
The amount of time between the arrival of the first and last audio RTP
packets to pass through a Call object. This is logged when the Call object
is destroyed. This is only logged if audio RTP packets are at some point in
time received, and is a way to omit temporary objects that do not send any
actual media.
</summary>
</histogram>
<histogram name="WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds" units="s"
expires_after="2021-12-12">
<owner>saza@chromium.org</owner>
<summary>
The amount of time between the arrival of the first and last video RTP
packets to pass through a Call object. This is logged when the Call object
is destroyed. This is only logged if video RTP packets are at some point in
time received, and is a way to omit temporary objects that do not send any
actual media.
</summary>
</histogram>
<histogram name="WebRTC.Call.VideoBitrateReceivedInKbps" units="kbps"
expires_after="2022-02-21">
<owner>holmer@chromium.org</owner>
<summary>
Average video bitrate received during a call, counted from first packet
received until Call instance is destroyed. Only mesured for calls that are
at least 10 seconds long.
</summary>
</histogram>
<histogram name="WebRTC.DataChannelCounters" enum="DataChannelCounters"
expires_after="2020-04-19">
<owner>perkj@chromium.org</owner>
<summary>
Counters on creation, opening, and a few main attributes of data channels.
</summary>
</histogram>
<histogram name="WebRTC.DataChannelMaxRetransmits" units="units"
expires_after="M85">
<owner>perkj@chromium.org</owner>
<summary>
The maximum number of retransmissions that are attempted in unreliable mode.
It is set to the value used in the configuration when a RTCDataChannel is
created.
</summary>
</histogram>
<histogram name="WebRTC.DataChannelMaxRetransmitTime" units="ms"
expires_after="M77">
<owner>perkj@chromium.org</owner>
<summary>
The length of the time window during which transmissions and retransmissions
may occur in unreliable mode. It is set to the value used in the
configuration when a RTCDataChannel is created.
</summary>
</histogram>
<histogram name="WebRTC.DesktopCapture.Win.DesktopCapturerImpl"
enum="WebRtcDesktopCapturerImpl" expires_after="2021-12-31">
<owner>jamiewalch@chromium.org</owner>
<owner>auorion@microsoft.com</owner>
<owner>edgecapabilitiesdev@microsoft.com</owner>
<summary>
This measures the frequency of use for each desktop capturer implementation,
allowing us to measure the adoption of the WGC capturer.
</summary>
</histogram>
<histogram name="WebRTC.DesktopCapture.Win.WgcCapturerResult"
enum="WebRtcWgcCapturerResult" expires_after="2021-12-31">
<owner>jamiewalch@chromium.org</owner>
<owner>auorion@microsoft.com</owner>
<owner>edgecapabilitiesdev@microsoft.com</owner>
<summary>
This records high level errors, or success, encountered across the entire
capture flow in the WGC based capturer.
</summary>
</histogram>
<histogram name="WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult"
enum="WebRtcWgcCaptureSessionGetFrameResult" expires_after="2021-12-31">
<owner>jamiewalch@chromium.org</owner>
<owner>auorion@microsoft.com</owner>
<owner>edgecapabilitiesdev@microsoft.com</owner>
<summary>
This records the result from retrieving a frame from the WGC based capturer.
</summary>
</histogram>
<histogram name="WebRTC.DesktopCapture.Win.WgcCaptureSessionStartResult"
enum="WebRtcWgcCaptureSessionStartResult" expires_after="2021-12-31">
<owner>jamiewalch@chromium.org</owner>
<owner>auorion@microsoft.com</owner>
<owner>edgecapabilitiesdev@microsoft.com</owner>
<summary>
This records the result from starting up the WGC based capturer.
</summary>
</histogram>
<histogram name="WebRTC.DesktopCapture.Win.{Capturer}CapturerFrameTime"
units="ms" expires_after="2021-12-31">
<owner>jamiewalch@chromium.org</owner>
<owner>auorion@microsoft.com</owner>
<owner>edgecapabilitiesdev@microsoft.com</owner>
<summary>This measures the performance of the {Capturer} capturer.</summary>
<token key="Capturer">
<variant name="DirectX" summary="DirectX based"/>
<variant name="Magnifier" summary="Magnifier based"/>
<variant name="ScreenGDI" summary="GDI based screen"/>
<variant name="Wgc" summary="WGC based"/>
<variant name="WindowGDI" summary="GDI based window"/>
</token>
</histogram>
<histogram name="WebRTC.DesktopCaptureCounters" enum="DesktopCaptureCounters"
expires_after="2021-12-31">
<owner>guidou@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Counters on creation of DesktopCaptureDevice and the first capture call.
</summary>
</histogram>
<histogram name="WebRTC.ICE.TcpSocketWriteErrorCode" enum="SocketErrorCode"
expires_after="M77">
<owner>zhihuang@chromium.org</owner>
<summary>
Counters on different types of TCP socket error code. Collected when we hit
the error code when writing.
</summary>
</histogram>
<histogram name="WebRTC.ICE.UdpSocketWriteErrorCode" enum="SocketErrorCode"
expires_after="M77">
<owner>zhihuang@chromium.org</owner>
<summary>
Counters on different types of UDP socket error code. Collected when we hit
the error code when writing.
</summary>
</histogram>
<histogram name="WebRTC.MediaStreamDevices.HasPanTiltZoomCamera"
enum="BooleanAvailable" expires_after="M100">
<owner>reillyg@chromium.org</owner>
<owner>device-dev@chromium.org</owner>
<summary>
Records whether a user would potentially see a permission prompt for moving
the camera. It is recorded when showing a camera permission prompt
regardless of whether or not the site requested camera movement.
</summary>
</histogram>
<histogram name="WebRTC.NAT.Metrics" enum="NatTypeCounters" expires_after="M85">
<owner>guoweis@chromium.org</owner>
<summary>
Counters on various types of NAT observed. This is logged once per session.
</summary>
</histogram>
<histogram name="WebRTC.NumDataChannelsPerPeerConnection" units="units"
expires_after="2020-02-16">
<owner>perkj@chromium.org</owner>
<summary>
Number of data channels created per PeerConnection. Sample added to the
histogram when the PeerConnection is destroyed. Note that this is done
purely on the renderer side, so no sample will be generated when the
renderer process is destroyed (as in the fast shutdown path for the
renderer) before the PeerConnection is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.AddIceCandidate"
enum="AddIceCandidateResult" expires_after="2022-02-09">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Outcomes of adding ICE candidates to a PeerConnection. Used to check the
theory that failures in candidate addition are ignored by applications.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.BundlePolicy"
enum="PeerConnectionBundlePolicy" expires_after="2022-01-01">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Determines whether BUNDLE is configured. Recorded during the first DTLS
connection establishment. Values are specified in
https://w3c.github.io/webrtc-pc/#dom-rtcbundlepolicy
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.BundleUsage"
enum="PeerConnectionBundleUsage" expires_after="2022-01-01">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Determines whether BUNDLE is used in remote offers or answers. Recorded
during setRemoteDescription calls. Distinguishes between simple, complex,
datachannel-only and legacy plan-b usage.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.CandidatePairType{IPProtocolType}"
enum="IceCandidatePairTypes" expires_after="2020-04-05">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Counters of various ICE Endpoint types. These values are logged for the
first selected candidate pair of a PeerConnection. {IPProtocolType}
</summary>
<token key="IPProtocolType" variants="IPProtocolType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.PeerConnection.CandidatePoolUsage.{BundlePolicy}"
units="components" expires_after="2022-01-01">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Measures the configured ice candidate poolsize for the {BundlePolicy}
bundlePolicy:
https://w3c.github.io/webrtc-pc/#dom-rtcconfiguration-icecandidatepoolsize
Recorded during the first DTLS connection establishment. See also
WebRTC.PeerConnection.BundlePolicy.
</summary>
<token key="BundlePolicy">
<variant name="Balanced"/>
<variant name="MaxBundle"/>
<variant name="MaxCompat"/>
</token>
</histogram>
<histogram name="WebRTC.PeerConnection.ConnectionState"
enum="IceConnectionStates" expires_after="2021-12-12">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Counters of ICE Connection states. These values are logged when the
PeerConnection gets into that state for the first time or after the ICE
restart.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.DtlsHandshakeError"
enum="DtlsHandshakeError" expires_after="M81">
<owner>zhihuang@chromium.org</owner>
<summary>
Records the error whenever the Dtls handshake fails. There are only two
types of errors, incompatitable cipher suite and unknown error, for now.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Duration.Network" units="microseconds"
expires_after="2022-05-24">
<owner>handellm@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Duration between the moment the WebRTC network thread's JingleThreadWrapper
begins running a task and the moment it ends executing it. It only measures
durations of tasks posted to rtc::Thread. Samples are acquired periodically
every several seconds by JingleThreadWrapper.
Warning: This metric does not include reports from clients with
low-resolution clocks (i.e. on Windows, ref.
|TimeTicks::IsHighResolution()|).
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Duration.Signaling" units="microseconds"
expires_after="2022-05-24">
<owner>handellm@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Duration between the moment the WebRTC signaling thread's
JingleThreadWrapper begins running a task and the moment it ends executing
it. It only measures durations of tasks posted to rtc::Thread. Samples are
acquired periodically every several seconds by JingleThreadWrapper.
Warning: This metric does not include reports from clients with
low-resolution clocks (i.e. on Windows, ref.
|TimeTicks::IsHighResolution()|). solution.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Duration.Worker" units="microseconds"
expires_after="2022-05-24">
<owner>handellm@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Duration between the moment the WebRTC worker thread's JingleThreadWrapper
begins running a task and the moment it ends executing it. It only measures
durations of tasks posted to rtc::Thread. Samples are acquired periodically
every several seconds by JingleThreadWrapper.
Warning: This metric does not include reports from clients with
low-resolution clocks (i.e. on Windows, ref.
|TimeTicks::IsHighResolution()|).
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IceRegatheringReason"
enum="IceRegatheringReason" expires_after="M77">
<owner>honghaiz@chromium.org</owner>
<summary>Records the reasons for ICE re-gathering.</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IceRestartState" enum="IceRestartState"
expires_after="M77">
<owner>honghaiz@chromium.org</owner>
<summary>
Records the transport channel states when ICE restart happens.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IPMetrics" enum="PeerConnectionCounters"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Counters on IPv4 and IPv6 usage in PeerConnection. These values are logged
once per PeerConnection.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IPPermissionStatus"
enum="IPPermissionStatus" expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Whether the permission to collect the local IP addresses in WebRTC has been
requested and/or granted. This is collected the first time when networks
updated event is reported or if never reported, during the destruction phase
of a call.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IPv4Interfaces" units="units"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Number of IPv4 network interfaces discovered in a PeerConnection Session.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IPv4LocalCandidates" units="units"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Number of IPv4 local Candidates gathered in a PeerConnection Session once
the ICE address gathering process reaches the Completed status. To avoid
miscounting, this only includes the first m line's first component.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IPv6Interfaces" units="units"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Number of IPv6 network interfaces discovered in a PeerConnection Session.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.IPv6LocalCandidates" units="units"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Number of IPv6 local Candidates gathered in a PeerConnection Session once
the ICE address gathering process reaches the Completed status. To avoid
miscounting, this only includes the first m line's first component.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.KeyProtocol"
enum="PeerConnectionKeyProtocol" expires_after="2021-11-07">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
What key exchange protocol (DTLS or SDES) is used to establish the crypto
keys for a PeerConnection's RTP transport. Note: This histogram was expired
after M82, and resurrected in M89.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.KeyProtocolByMedia"
enum="PeerConnectionKeyProtocolByMedia" expires_after="2021-11-07">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
What key exchange protocol (DTLS or SDES) is used to establish the crypto
keys for a PeerConnection's RTP transport, specified per media type. Note:
This histogram was expired after M82, and resurrected in M89.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Latency.Network" units="microseconds"
expires_after="2022-05-24">
<owner>handellm@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Latency defined as the duration between the moment a task is scheduled from
the WebRTC network thread's JingleThreadWrapper's task runner, and the
moment it begins running. Samples are acquired periodically every several
seconds by JingleThreadWrapper.
Warning: This metric does not include reports from clients with
low-resolution clocks (i.e. on Windows, ref.
|TimeTicks::IsHighResolution()|).
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Latency.Signaling" units="microseconds"
expires_after="2022-05-24">
<owner>handellm@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Latency defined as the duration between the moment a task is scheduled from
the WebRTC signaling thread's JingleThreadWrapper's task runner, and the
moment it begins running. Samples are acquired periodically every several
seconds by JingleThreadWrapper.
Warning: This metric does not include reports from clients with
low-resolution clocks (i.e. on Windows, ref.
|TimeTicks::IsHighResolution()|).
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Latency.Worker" units="microseconds"
expires_after="2022-05-24">
<owner>handellm@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Latency defined as the duration between the moment a task is scheduled from
the WebRTC worker thread's JingleThreadWrapper's task runner, and the moment
it begins running. Samples are acquired periodically every several seconds
by JingleThreadWrapper.
Warning: This metric does not include reports from clients with
low-resolution clocks (i.e. on Windows, ref.
|TimeTicks::IsHighResolution()|).
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.OfferExtmapAllowMixed"
enum="PeerConnectionOfferExtmapAllowMixed" expires_after="2022-01-02">
<owner>kron@chromium.org</owner>
<summary>
What setting for the SDP attribute extmap-allow-mixed has been asked for by
the creator of a PeerConnection. This is specified to the constructor
through the dictionary property offerExtmapAllowMixed which can be set to
either true or false. A default value will be used if it's not specified.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.ProvisionalAnswer"
enum="PeerConnectionProvisionalAnswer" expires_after="2022-06-01">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Whether provisional answers are used. Recorded during the first DTLS
connection establishment.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.RtcpMux" enum="PeerConnectionRtcpMux"
expires_after="2020-02-23">
<owner>pthatcher@chromium.org</owner>
<summary>
Whether RTCP-mux is used for the PeerConnection (both the local and remote
description enable RTCP-mux). Recorded after SetLocalDescription and
SetRemoteDescription are called, once per PeerConnection.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpComplexUsage.CreateAnswer"
enum="PeerConnectionSdpUsageCategory" expires_after="2022-05-07">
<owner>hbos@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
The SDP usage category (&quot;safe&quot;, &quot;unsafe&quot; or
&quot;unknown&quot;) of createAnswer(). Using complex SDP without explicitly
specifying the sdpSemantics is considered unsafe in this context because
such usage is sensitive to the rollout of a different default SDP semantic.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpComplexUsage.CreateOffer"
enum="PeerConnectionSdpUsageCategory" expires_after="2022-05-07">
<owner>hbos@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
The SDP usage category (&quot;safe&quot;, &quot;unsafe&quot; or
&quot;unknown&quot;) of createOffer(). Using complex SDP without explicitly
specifying the sdpSemantics is considered unsafe in this context because
such usage is sensitive to the rollout of a different default SDP semantic.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpComplexUsage.SetLocalAnswer"
enum="PeerConnectionSdpUsageCategory" expires_after="2022-05-07">
<owner>hbos@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
The SDP usage category (&quot;safe&quot;, &quot;unsafe&quot; or
&quot;unknown&quot;) of setLocalDescription(answer). Using complex SDP
without explicitly specifying the sdpSemantics is considered unsafe in this
context because such usage is sensitive to the rollout of a different
default SDP semantic.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpComplexUsage.SetLocalOffer"
enum="PeerConnectionSdpUsageCategory" expires_after="2022-05-07">
<owner>hbos@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
The SDP usage category (&quot;safe&quot;, &quot;unsafe&quot; or
&quot;unknown&quot;) of setLocalDescription(offer). Using complex SDP
without explicitly specifying the sdpSemantics is considered unsafe in this
context because such usage is sensitive to the rollout of a different
default SDP semantic.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpComplexUsage.SetRemoteAnswer"
enum="PeerConnectionSdpUsageCategory" expires_after="2022-05-07">
<owner>hbos@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
The SDP usage category (&quot;safe&quot;, &quot;unsafe&quot; or
&quot;unknown&quot;) of setRemoteDescription(answer). Using complex SDP
without explicitly specifying the sdpSemantics is considered unsafe in this
context because such usage is sensitive to the rollout of a different
default SDP semantic.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpComplexUsage.SetRemoteOffer"
enum="PeerConnectionSdpUsageCategory" expires_after="2022-05-07">
<owner>hbos@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
The SDP usage category (&quot;safe&quot;, &quot;unsafe&quot; or
&quot;unknown&quot;) of setRemoteDescription(offer). Using complex SDP
without explicitly specifying the sdpSemantics is considered unsafe in this
context because such usage is sensitive to the rollout of a different
default SDP semantic.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpFormatReceived"
enum="PeerConnectionSdpFormatReceived" expires_after="2021-12-12">
<owner>steveanton@chromium.org</owner>
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
What SDP format is received in the remote offer. The value &quot;no
tracks&quot; means that no audio or video tracks were received. The value
&quot;simple&quot; means that at most one audio and at most one video track
was received. The value &quot;complex&quot; means that more than one audio
or more than one video track was received, and how this was signaled is
indicated (&quot;Plan B&quot; meaning with a=ssrc lines within the same m=
section and &quot;Unified Plan&quot; meaning with a separate m= section).
This is recorded when calling setRemoteDescription with an SDP Offer.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpFormatReceivedAnswer"
enum="PeerConnectionSdpFormatReceived" expires_after="2021-12-12">
<owner>steveanton@chromium.org</owner>
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
What SDP format is received in the remote answer. See
WebRTC.PeerConnection.SdpFormatReceived for the description of the values.
This is recorded when calling setRemoteDescription with an SDP Answer.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpSemanticNegotiated"
enum="PeerConnectionSdpSemanticNegotiated" expires_after="2021-11-07">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
What SDP semantic (Unified Plan or Plan B) was detected when completing
negotiation of a PeerConnection. This is recorded when accepting an SDP
Answer. The value &quot;mixed&quot; means that the accepted answer included
both types of marker in the SDP. The value &quot;none&quot; will happen when
the answerer sends no media.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SdpSemanticRequested"
enum="PeerConnectionSdpSemanticRequested" expires_after="2021-12-05">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
What SDP semantic (Unified Plan, Plan B or &quot;use default&quot;) has been
asked for by the creator of a PeerConnection.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Simulcast.ApplyLocalDescription"
enum="SimulcastApiVersion" expires_after="2020-08-23">
<owner>amithi@chromium.org</owner>
<summary>
Was simulcast applied to the local description and with which API surface.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription"
enum="SimulcastApiVersion" expires_after="2020-08-30">
<owner>amithi@chromium.org</owner>
<summary>
Was simulcast applied to the remote description and with which API surface.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Simulcast.Disabled" units="units"
expires_after="M81">
<owner>amithi@chromium.org</owner>
<summary>
Simulcast was disabled because it is not supported by the remote party. This
is a counter that can take on only a single value (1).
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings"
units="units" expires_after="2020-08-30">
<owner>amithi@chromium.org</owner>
<summary>
Counts the number of send encodings given to PeerConnection::AddTransceiver.
This histogram will be used to understand if and how the API is used.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SrtcpUnprotectError"
enum="SrtpErrorCode" expires_after="never">
<!-- expires-never: Needed for long-term tracking of the ecosystem. -->
<owner>steveanton@chromium.org</owner>
<summary>
What error code is reported by libsrtp when failing to unprotect an incoming
SRTCP (secured media control) packet.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SrtpCryptoSuite{WebRTCMediaType}"
enum="DTLS_SRTPCryptoSuite" expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Counters on the type of SRTP crypto suites used by WebRTC. This is collected
whenever the transport signals the OnCompleted event. {WebRTCMediaType}
</summary>
<token key="WebRTCMediaType" variants="WebRTCMediaType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.PeerConnection.SrtpUnprotectError" enum="SrtpErrorCode"
expires_after="never">
<!-- expires-never: Needed for long-term tracking of the ecosystem. -->
<owner>steveanton@chromium.org</owner>
<summary>
What error code is reported by libsrtp when failing to unprotect an incoming
SRTP (secured media) packet.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.SslCipherSuite{WebRTCMediaType}"
enum="SSLCipherSuite" expires_after="never">
<!-- expires-never: This is useful for deprecating old cipher suites; the
need for this can occur at long intervals. -->
<owner>hta@google.com</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Counters on the type of SSL cipher suites used by WebRTC. This is collected
whenever the transport signals the OnCompleted event. {WebRTCMediaType}
</summary>
<token key="WebRTCMediaType" variants="WebRTCMediaType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.PeerConnection.ThermalState" enum="ThermalState"
expires_after="2021-12-05">
<owner>eshr@google.com</owner>
<owner>hbos@chromium.org</owner>
<summary>
Measures computer thermal state, sampled every 60s when a PeerConnection is
open with a video sender. Most quick toggles between thermal states are thus
not sampled, but thermal states are generally stable so these toggles should
be rare.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.TimeToConnect" units="ms"
expires_after="2020-09-06">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>Time to setup a peer to peer call with PeerConnection.</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.TimeToNetworkUpdated" units="ms"
expires_after="M77">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
Time to receive the first SignalNetworksChanged from the request to start
updating network in PeerConnection.
</summary>
</histogram>
<histogram name="WebRTC.PeerConnection.UsagePattern"
enum="WebRtcPeerConnectionUsagePattern" expires_after="2021-12-05">
<owner>hta@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Capsule history of a WebRTC PeerConnection, encoded as a sequence of bits
encapsulated in an integer. Only a few values will be deemed interesting,
but the interesting values may change over time.
</summary>
</histogram>
<histogram name="WebRTC.ReceivedAudioTrackDuration" units="ms"
expires_after="2021-11-28">
<owner>perkj@chromium.org</owner>
<summary>
Durations of audio tracks received over a PeerConnection. The stopwatch
starts when the track first becomes connected, and ends when it is
disconnected or very soon thereafter.
</summary>
</histogram>
<histogram name="WebRTC.ReceivedVideoTrackDuration" units="ms"
expires_after="2021-11-28">
<owner>perkj@chromium.org</owner>
<summary>
Durations of video tracks received over a PeerConnection. The stopwatch
starts when the track first becomes connected, and ends when it is
disconnected or very soon thereafter.
</summary>
</histogram>
<histogram name="WebRTC.ReliableDataChannelMessageSize" units="bytes"
expires_after="2021-12-05">
<owner>perkj@chromium.org</owner>
<summary>
Sizes of messages sent over reliable data channels. The size of an
individual message is added to the histogram as a sample immediately when a
message is sent.
</summary>
</histogram>
<histogram name="WebRTC.ScreenCaptureTime" units="ms"
expires_after="2020-10-18">
<owner>jiayl@chromium.org</owner>
<summary>Time for capturing one frame in screen capturing.</summary>
</histogram>
<histogram name="WebRTC.SentAudioTrackDuration" units="ms"
expires_after="2021-11-28">
<owner>perkj@chromium.org</owner>
<summary>
Durations of audio tracks sent over a PeerConnection. The stopwatch starts
when the track first becomes connected, and ends when it is disconnected or
very soon thereafter.
</summary>
</histogram>
<histogram name="WebRTC.SentVideoTrackDuration" units="ms"
expires_after="2021-11-07">
<owner>perkj@chromium.org</owner>
<summary>
Durations of video tracks sent over a PeerConnection. The stopwatch starts
when the track first becomes connected, and ends when it is disconnected or
very soon thereafter.
</summary>
</histogram>
<histogram name="WebRTC.Stun.BatchSuccessPercent{NatType}" units="%"
expires_after="M85">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
For clients using a shared source port per STUN binding request toward the
specified servers, success rate for requests which received a response with
various intervals between requests. Only the first instance of renderers
will conduct the trial and log this result. The STUN binding requests are
grouped into multiple batches and the success rate is calculated for an
individual batch. {NatType}
</summary>
<token key="NatType" variants="NatType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Stun.ResponseLatency{NatType}" units="ms"
expires_after="M85">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
For clients using a shared source port per STUN binding request, average RTT
for requests which received a response with various intervals between
requests. Only the first instance of renderers will conduct the trial and
log this result. {NatType}
</summary>
<token key="NatType" variants="NatType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Stun.SuccessPercent{NatType}" units="%"
expires_after="M85">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
For clients using a shared source port per STUN binding request, success
rate for requests which received a response with various intervals between
requests. Only the first instance of renderers will conduct the trial and
log this result. {NatType}
</summary>
<token key="NatType" variants="NatType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.SystemMaxConsecutiveBytesDelayed{IPProtocolType}"
units="units" expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
The maximum of consecutive delayed bytes caused by EWOULDBLOCKs from system.
This happens when system can't send any packet synchronously at that moment.
{IPProtocolType}
</summary>
<token key="IPProtocolType" variants="IPProtocolType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.SystemPercentPacketsDelayed{IPProtocolType}" units="%"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
The percentage of packets delayed due to ERR_IO_PENDING from system in a
WebRTC socket. This happens when system can't send any packet synchronously
at that moment. {IPProtocolType}
</summary>
<token key="IPProtocolType" variants="IPProtocolType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.SystemSendPacketDuration{IPProtocolType}" units="ms"
expires_after="M81">
<owner>qingsi@google.com</owner>
<owner>jeroendb@google.com</owner>
<summary>
The duration that it takes to send out a packet in system layer. This
includes both the queuing time (under the condition when socket returns
EWOULDBLOCK from system) as well as the time system takes to finish the
asynchronous send. For UDP, it's the time from P2PSocketHostUdp::Send to
P2PSocketHostUdp::HandleSendResult. Tcp part is to be implemented.
{IPProtocolType}
</summary>
<token key="IPProtocolType" variants="IPProtocolType">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.UnreliableDataChannelMessageSize" units="bytes"
expires_after="2021-11-21">
<owner>perkj@chromium.org</owner>
<summary>
Sizes of messages sent over unreliable data channels. The size of an
individual message is added to the histogram as a sample immediately when a
message is sent.
</summary>
</histogram>
<histogram name="WebRTC.UserMediaRequest.NoResultState"
enum="MediaStreamRequestState" expires_after="M77">
<owner>andresp@chromium.org</owner>
<summary>
The state of a UserMediaRequest when it gets destroyed before having a
result.
Note: &quot;Explicitly Cancelled&quot; means
MediaStreamImpl::cancelUserMediaRequest was called and not necessarily that
the user cancelled. Those are likely tracked as UserMediaRequest with a
result of permission denied.
</summary>
</histogram>
<histogram name="WebRTC.UserMediaRequest.Result2"
enum="MediaStreamRequestResult2" expires_after="2022-02-07">
<owner>guidou@chromium.org</owner>
<owner>agpalak@chromium.org</owner>
<summary>
Counters for UserMediaRequests results such as failure reasons. The standard
specification error names are in parenthesis.
</summary>
</histogram>
<histogram name="WebRTC.Video.AdaptChangesPerMinute{VideoAdaptationReason}"
units="changes/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average number of adaptation changes per minute for a sent video stream.
Recorded when a stream is removed. {VideoAdaptationReason}
</summary>
<token key="VideoAdaptationReason">
<variant name=""/>
<variant name=".Cpu" summary="Adapt reason: CPU."/>
<variant name=".Quality" summary="Adapt reason: quality."/>
</token>
</histogram>
<histogram name="WebRTC.Video.AverageRoundTripTimeInMilliseconds" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>holmer@chromium.org</owner>
<summary>
The average round-trip time of a WebRTC call in milliseconds. Recorded when
a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.AVSyncOffsetInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The absolute value of the sync offset between a rendered video frame and the
latest played audio frame is measured per video frame. The average offset
per received video stream is recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.BandwidthLimitedResolutionInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of sent frames that are limited in resolution due to bandwidth
for a sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.BandwidthLimitedResolutionsDisabled"
units="disabled resolutions" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
For frames that are limited in resolution due to bandwidth, the average
number of disabled resolutions is recorded for a sent video stream. Recorded
when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.BitrateReceivedInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received bits per second for a received video stream. Recorded
when a stream is removed. The total number of bytes is divided by the time
the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.BitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent bits per second for a sent video stream. Recorded when a
stream is removed. The total number of bytes is divided by the time the
video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.CpuLimitedResolutionInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of frames that are limited in resolution due to CPU for a sent
video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.CurrentDelayInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Average current delay for a received video stream. This is the actual delay
imposed on frames (where the goal is to reach the target delay (see
WebRTC.Video.TargetDelayInMs)). Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Decoded.Vp8.Qp{WebRTCVideoExperimentGroupId}"
units="qp value" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average QP (quantizer value) per frame for a received VP8 video stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.DecodedFramesPerSecond" units="fps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of decoded frames per second for a received video stream.
Recorded when a stream is removed. The total number of frames is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.DecodeTimeInMs" units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average decode time per frame for a received video stream. Recorded when
a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DecodeTimePerFrameInMs{CodecInfo}" units="ms"
expires_after="2022-02-23">
<owner>kron@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The decode time per frame for a received video stream. Continously updated
after each frame has been decoded. {CodecInfo}
</summary>
<token key="CodecInfo">
<variant name="">
<obsolete>
Base histogram. Use suffixes of this histogram instead.
</obsolete>
</variant>
<variant name=".H264.4k.Hw" summary=""/>
<variant name=".H264.4k.Sw" summary=""/>
<variant name=".H264.Hd.Hw" summary=""/>
<variant name=".H264.Hd.Sw" summary=""/>
<variant name=".Vp9.4k.Hw" summary=""/>
<variant name=".Vp9.4k.Sw" summary=""/>
<variant name=".Vp9.Hd.Hw" summary=""/>
<variant name=".Vp9.Hd.Sw" summary=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.DelayedFramesToRenderer" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of delayed frames to renderer for a received video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average delay of delayed frames to renderer for a received video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DroppedFrames.Capturer" units="frames"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by a capturer for a sent video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DroppedFrames.Encoder" units="frames"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by an encoder's internal rate limiter for a
sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DroppedFrames.EncoderQueue" units="frames"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped because encoder queue is full for a sent
video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DroppedFrames.Ratelimiter" units="frames"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by a WebRTC rate limiter (in MediaOpt) for a
sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.DroppedFrames.Receiver" units="frames"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by a WebRTC on the receive side because they
are incomplete or undecodable.
</summary>
</histogram>
<histogram name="WebRTC.Video.Encoded.Qp{VideoEncodedQpStats}" units="qp value"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average QP (quantizer value) per frame for a sent video stream. Recorded
when a stream is removed. {VideoEncodedQpStats}
</summary>
<token key="VideoEncodedQpStats">
<variant name=""/>
<variant name=".H264"
summary="Video codec: H264. QP range: 0-51. No spatial layers."/>
<variant name=".Vp8"
summary="Video codec: VP8. QP range: 0-127. Single stream sent."/>
<variant name=".Vp8.S0"
summary="Video codec: VP8. QP range: 0-127. Spatial index 0."/>
<variant name=".Vp8.S1"
summary="Video codec: VP8. QP range: 0-127. Spatial index 1."/>
<variant name=".Vp8.S2"
summary="Video codec: VP8. QP range: 0-127. Spatial index 2."/>
<variant name=".Vp9"
summary="Video codec: VP9. QP range: 0-255. No spatial layers."/>
<variant name=".Vp9.S0"
summary="Video codec: VP9. QP range: 0-255. Spatial layer 0."/>
<variant name=".Vp9.S1"
summary="Video codec: VP9. QP range: 0-255. Spatial layer 1."/>
<variant name=".Vp9.S2"
summary="Video codec: VP9. QP range: 0-255. Spatial layer 2."/>
</token>
</histogram>
<histogram name="WebRTC.Video.Encoder.CodecType" enum="WebRtcVideoCodecs"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Configured video codec for a sent video stream. Recorded when a
VideoSendStream is destroyed (for streams whose lifetime is longer than 10
seconds).
</summary>
</histogram>
<histogram name="WebRTC.Video.EncodeTimeInMs" units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average encode time per frame for a sent video stream. Recorded when a
stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.EndToEndDelayInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average end-to-end delay per frame for a received video stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.EndToEndDelayMaxInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The maximum end-to-end delay per frame for a received video stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.FecBitrateReceivedInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received FEC bits per second for a received video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.FecBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent FEC bits per second for a sent video stream. Recorded
when a stream is removed. The total number of bytes is divided by the time
the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.FirPacketsReceivedPerMinute"
units="packets/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received RTCP FIR packets per minute for a sent video stream.
Recorded when a stream is removed. The total number of packets is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.FirPacketsSentPerMinute" units="packets/minute"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent RTCP FIR packets per minute for a received video stream.
Recorded when a stream is removed. The total number of packets is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.H264DecoderImpl.Event"
enum="WebRtcH264DecoderImplEvent" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>hbos@chromium.org</owner>
<summary>
The number of |H264DecoderImpl| events, such as an initialization or
decoding error, that have occurred. At most one Init and one Error is
reported per |H264DecoderImpl| instance. This is to avoid the same event
from being reported multiple times (e.g. if there is an error you might
re-initialize or get a decode error every frame which would otherwise
pollute the data).
</summary>
</histogram>
<histogram name="WebRTC.Video.H264EncoderImpl.Event"
enum="WebRtcH264EncoderImplEvent" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>hbos@chromium.org</owner>
<summary>
The number of |H264EncoderImpl| events, such as an initialization or
encoding error, that have occurred. At most one Init and one Error is
reported per |H264EncoderImpl| instance. This is to avoid the same event
from being reported multiple times (e.g. if there is an error you might
re-initialize or get an encode error every frame which would otherwise
pollute the data).
</summary>
</histogram>
<histogram
name="WebRTC.Video.HardwareDecodedFramesBetweenSoftwareFallbacks{WebRtcCodecs}"
units="frames" expires_after="2022-02-23">
<owner>kron@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The number of hardware decoded frames between fallbacks to software decoder
for a received video stream. {WebRtcCodecs}
</summary>
<token key="WebRtcCodecs">
<variant name="">
<obsolete>
Base histogram. Use suffixes of this histogram instead.
</obsolete>
</variant>
<variant name=".Av1" summary=""/>
<variant name=".Generic" summary=""/>
<variant name=".H264" summary=""/>
<variant name=".Multiplex" summary=""/>
<variant name=".Vp8" summary=""/>
<variant name=".Vp9" summary=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.InputFramesPerSecond" units="fps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of incoming frames per second for a sent video stream. Recorded
when a stream is removed. The total number of frames is divided by the time
the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.InputHeightInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average input height per frame (for incoming frames to video engine) for
a sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.InputWidthInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average input width per frame (for incoming frames to video engine) for
a sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.InterframeDelay95PercentileInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The 95th percentile of interframe delay for a received video stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.InterframeDelayInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average interframe delay for a received video stream. Recorded when a
stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.InterframeDelayMaxInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The maximum interframe delay for a received video stream. Recorded when a
stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.JitterBufferDelayInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Average jitter buffer delay for a received video stream. Recorded when a
stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.KeyFramesReceivedInPermille{WebRTCVideoExperimentGroupId}"
units="permille" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Permille of frames that are key frames for a received video stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.KeyFramesSentInPermille" units="permille"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Permille of frames that are key frames for a sent video stream. Recorded
when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.MeanFreezeDurationMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average duration of a freeze in video playback. Recorded when a received
stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.MeanTimeBetweenFreezesMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average duration of a smooth video playback. Recorded when a received
stream is removed or content type changes.
</summary>
</histogram>
<histogram
name="WebRTC.Video.MediaBitrateReceivedInKbps{WebRTCVideoExperimentGroupId}"
units="kbps" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received media payload bits per second for a received video
stream. Recorded when a stream is removed. The total number of bytes is
divided by the time the video stream exists. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.MediaBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent media payload bits per second for a sent video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.NackPacketsReceivedPerMinute"
units="packets/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received RTCP NACK packets per minute for a sent video stream.
Recorded when a stream is removed. The total number of packets is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.NackPacketsSentPerMinute" units="packets/minute"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent RTCP NACK packets per minute for a received video stream.
Recorded when a stream is removed. The total number of packets is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.NumberFreezesPerMinute" units="freezes/minute"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ssilkin@chromium.org</owner>
<summary>
The number of video freezes per minute for a received video stream. Recorded
when a stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.NumberOfPauseEvents" units="pause events"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of times a video stream has been paused/resumed during a call.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.NumberResolutionDownswitchesPerMinute"
units="switches/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average number of resolution down-switches per minute for a received
video stream. Recorded when a stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.OnewayDelayInMs" units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Average delay (network delay (rtt/2) + jitter delay + decode time + render
delay) for a received video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.PaddingBitrateReceivedInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received padding bits per second for a received video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.PaddingBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent padding bits per second for a sent video stream. Recorded
when a stream is removed. The total number of bytes is divided by the time
the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.PausedTimeInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of time that the video has been paused for a sent video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.PliPacketsReceivedPerMinute"
units="packets/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received RTCP PLI packets per minute for a sent video stream.
Recorded when a stream is removed. The total number of packets is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.PliPacketsSentPerMinute" units="packets/minute"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent RTCP PLI packets per minute for a received video stream.
Recorded when a stream is removed. The total number of packets is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.QualityLimitedResolutionDownscales"
units="downscales" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
For frames that are downscaled in resolution due to quality, the average
number of downscales is recorded for a sent video stream. Recorded when a
stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.QualityLimitedResolutionInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of sent frames that are downscaled in resolution due to quality
for a sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.ReceivedFecPacketsInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of received FEC packets for a received video stream. Recorded
when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.ReceivedHeightInPixels{WebRTCVideoExperimentGroupId}"
units="pixels" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average received height per frame for a received video stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.ReceivedPacketsLostInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of received packets lost for a received video stream. Recorded
when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.ReceivedWidthInPixels{WebRTCVideoExperimentGroupId}"
units="pixels" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average received width per frame for a received video stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.ReceiveStreamLifetimeInSeconds" units="seconds"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The lifetime of a video receive stream. Recorded when a VideoReceiveStream
instance is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.Video.RecoveredMediaPacketsInPercentOfFec" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of recovered media packets from FEC packets for a received video
stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.RenderFramesPerSecond" units="fps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent frames to the renderer per second for a received video
stream. Recorded when a stream is removed. The total number of frames is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.RenderSqrtPixelsPerSecond" units="pps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of pixels (sqrt(width*height)) of sent frames to the renderer per
second for a received video stream. Recorded when a stream is removed. The
total number of pixels is divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.RetransmittedBitrateReceivedInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of retransmitted bits per second for a received video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.RetransmittedBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of retransmitted bits per second for a sent video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.RtpToNtpFreqOffsetInKhz" units="kHz"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The absolute value of the difference between the estimated frequency during
RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is
measured per received video frame. The max offset during 40 second intervals
is stored. The average of these stored offsets per received video stream is
recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.RtxBitrateReceivedInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of received bits over RTX per second for a received video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.RtxBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent bits over RTX per second for a sent video stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.BandwidthLimitedResolutionInPercent"
units="%" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Percentage of sent frames that are limited in resolution due to bandwidth
for a sent (screen content) video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.BandwidthLimitedResolutionsDisabled"
units="disabled resolutions" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
For frames that are limited in resolution due to bandwidth, the average
number of disabled resolutions is recorded for a sent (screen content) video
stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.BitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of sent bits per second for a sent screenshare stream. Recorded
when a stream is removed. The total number of bytes is divided by the time
the video stream exists.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.Decoded.Vp8.Qp{WebRTCVideoExperimentGroupId}"
units="qp value" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average QP (quantizer value) per frame for a received VP8 screenshare
stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.DroppedFrames.Capturer"
units="frames" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by a capturer for a sent screenshare stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.DroppedFrames.Encoder" units="frames"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by an encoder's internal rate limiter for a
sent screenshare stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.DroppedFrames.EncoderQueue"
units="frames" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped because encoder queue is full for a sent
screenshare stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.DroppedFrames.Ratelimiter"
units="frames" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Total number of frames dropped by a WebRTC rate limiter (in MediaOpt) for a
sent screenshare stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.EncodeTimeInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average encode time per frame for a sent (screen content) video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.EndToEndDelayInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average end-to-end delay per frame for a received screenshare stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.EndToEndDelayMaxInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The maximum end-to-end delay per frame for a received screenshare stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.FecBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of sent FEC bits per second for a sent screenshare stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.FirPacketsReceivedPerMinute"
units="packets/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of received RTCP FIR packets per minute for a sent screenshare
stream. Recorded when a stream is removed. The total number of packets is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.FramesPerDrop"
units="sent/dropped ratio" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Ratio of sent frames to dropped frames at the encoder. The value is reported
when a stream is removed and is calculated as the total number frames sent
divided by the number of dropped frames.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.FramesPerOvershoot"
units="sent/overshoot ratio" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Ratio of sent frames to number of re-encoded frames (due to target bitrate
overshoot). The value is reported when a stream is removed and is calculated
as the total number frames sent divided by the number of re-encoded frames.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.InputFramesPerSecond" units="fps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of incoming frames per second for a sent (screen content) video
stream. Recorded when a stream is removed. The total number of frames is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.InputHeightInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average input height per frame (for incoming frames to video engine) for
a sent (screen content) video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.InputWidthInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average input width per frame (for incoming frames to video engine) for
a sent (screen content) video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.InterframeDelay95PercentileInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The 95th percentile of interframe delay for a received screenshare stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.InterframeDelayInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average interframe delay for a received screenshare stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.InterframeDelayMaxInMs{WebRTCVideoExperimentGroupId}"
units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The maximum interframe delay for a received screenshare stream. Recorded
when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.KeyFramesReceivedInPermille{WebRTCVideoExperimentGroupId}"
units="permille" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Permille of frames that are key frames for a received screenshare stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.KeyFramesSentInPermille"
units="permille" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Permille of frames that are key frames for a sent (screen content) video
stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.Layer0{ScreenshareLayerStats}"
units="units" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Stats for the lower layer (TL0) of a screenshare stream in conference mode.
{ScreenshareLayerStats}
</summary>
<token key="ScreenshareLayerStats" variants="ScreenshareLayerStats">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.Layer1{ScreenshareLayerStats}"
units="units" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Stats for the higher layer (TL1) of a screenshare stream in conference mode.
{ScreenshareLayerStats}
</summary>
<token key="ScreenshareLayerStats" variants="ScreenshareLayerStats">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.MeanFreezeDurationMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average duration of a freeze in screenshare playback. Recorded then a
received stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.MeanTimeBetweenFreezesMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average duration of a smooth video playback for screenshare stream.
Recorded when a received stream is removed or content type changes.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.MediaBitrateReceivedInKbps{WebRTCVideoExperimentGroupId}"
units="kbps" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The number of received media payload bits per second for a received
screenshare stream. Recorded when a stream is removed. The total number of
bytes is divided by the time the video stream exists.
{WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.MediaBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of sent media payload bits per second for a sent screenshare
stream. Recorded when a stream is removed. The total number of bytes is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.NackPacketsReceivedPerMinute"
units="packets/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of received RTCP NACK packets per minute for a sent screenshare
stream. Recorded when a stream is removed. The total number of packets is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.NumberResolutionDownswitchesPerMinute"
units="switches/minute" expires_after="M88">
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average number of resolution down-switches per minute for a receive
screenshare stream. Recorded when a stream is removed or content type
changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.PaddingBitrateSentInKbps"
units="kbps" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of sent padding bits per second for a sent screenshare stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.PliPacketsReceivedPerMinute"
units="packets/minute" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of received RTCP PLI packets per minute for a sent screenshare
stream. Recorded when a stream is removed. The total number of packets is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.QualityLimitedResolutionDownscales"
units="downscales" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
For frames that are downscaled in resolution due to quality, the average
number of downscales is recorded for a sent (screen content) video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.QualityLimitedResolutionInPercent"
units="%" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Percentage of sent frames that are downscaled in resolution due to quality
for a sent (screen content) video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.ReceivedHeightInPixels{WebRTCVideoExperimentGroupId}"
units="pixels" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average received height per frame for a received screenshare stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.ReceivedPacketsLostInPercent"
units="%" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Percentage of received packets lost for a received screenshare stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram
name="WebRTC.Video.Screenshare.ReceivedWidthInPixels{WebRTCVideoExperimentGroupId}"
units="pixels" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The average received width per frame for a received screenshare stream.
Recorded when a stream is removed. {WebRTCVideoExperimentGroupId}
</summary>
<token key="WebRTCVideoExperimentGroupId"
variants="WebRTCVideoExperimentGroupId">
<variant name=""/>
</token>
</histogram>
<histogram name="WebRTC.Video.Screenshare.ReceiveStreamLifetimeInSeconds"
units="seconds" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The lifetime of a screenshare receive stream. Recorded when a
VideoReceiveStream instance is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.RetransmittedBitrateSentInKbps"
units="kbps" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of retransmitted bits per second for a sent screenshare stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.RtxBitrateSentInKbps" units="kbps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The number of sent bits over RTX per second for a sent screenshare stream.
Recorded when a stream is removed. The total number of bytes is divided by
the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SendSideDelayInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average delay (of average delays) of sent packets for a sent (screen
content) video stream. Recorded when a stream is removed. The delay is
measured from a frame is input to video engine until a packet is sent to the
network. For each sent packet, the average delay of all sent packets over
the last second is reported. The average of these reported delays is
recorded.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SendSideDelayMaxInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average delay (of max delays) of sent packets for a sent (screen
content) video stream. Recorded when a stream is removed. The delay is
measured from a frame is input to video engine until a packet is sent to the
network. For each sent packet, the maximum delay of all sent packets over
the last second is reported. The average of these reported delays is
recorded.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SentFramesPerSecond" units="fps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
The number of sent frames per second for a sent (screen content) video
stream. Recorded when a stream is removed. The total number of frames is
divided by the time the video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SentHeightInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average sent height per frame for a sent (screen content) video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SentPacketsLostInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Percentage of sent packets lost for a sent screenshare stream. Recorded when
a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SentToInputFpsRatioPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Ratio between Screenshare.SentFramesPerSecond and
Screenshare.InputFramesPerSecond in percents. Recorded when a stream is
removed. The total number of sent frames is divided by the total number of
input frames and multiplied by 100.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.SentWidthInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
The average sent width per frame for a sent (screen content) video stream.
Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.TimeInBlockyVideoPercentage"
units="%" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Percentage of time the received screenshare stream playbacks low quality
blocky video. Recorded when a stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.TimeInHdPercentage" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Percentage of time the received screenshare stream playbacks HD resolution.
Recorded when a stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.Screenshare.UniqueNackRequestsReceivedInPercent"
units="%" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>sprang@chromium.org</owner>
<summary>
Percentage of unique RTCP NACK requests that are received in response to a
sent screenshare stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.SendDelayInMs" units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average send delay of sent packets for a sent video stream. Recorded
when a stream is removed. The delay is measured from a packet is sent to the
transport until leaving the socket.
</summary>
</histogram>
<histogram name="WebRTC.Video.SendSideDelayInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average delay (of average delays) of sent packets for a sent video
stream. Recorded when a stream is removed. The delay is measured from a
frame is input to video engine until a packet is sent to the network. For
each sent packet, the average delay of all sent packets over the last second
is reported. The average of these reported delays is recorded.
</summary>
</histogram>
<histogram name="WebRTC.Video.SendSideDelayMaxInMs" units="ms"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average delay (of max delays) of sent packets for a sent video stream.
Recorded when a stream is removed. The delay is measured from a frame is
input to video engine until a packet is sent to the network. For each sent
packet, the maximum delay of all sent packets over the last second is
reported. The average of these reported delays is recorded.
</summary>
</histogram>
<histogram name="WebRTC.Video.SendStreamLifetimeInSeconds" units="seconds"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The lifetime of a video send stream. Recorded when a VideoSendStream
instance is destroyed.
</summary>
</histogram>
<histogram name="WebRTC.Video.SentFramesPerSecond" units="fps"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The number of sent frames per second for a sent video stream. Recorded when
a stream is removed. The total number of frames is divided by the time the
video stream exists.
</summary>
</histogram>
<histogram name="WebRTC.Video.SentHeightInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average sent height per frame for a sent video stream. Recorded when a
stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.SentPacketsLostInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of sent packets lost for a sent video stream. Recorded when a
stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.SentToInputFpsRatioPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Ratio between SentFramesPerSecond and InputFramesPerSecond in percents.
Recorded when a stream is removed. The total number of sent frames is
divided by the total number of input frames and multiplied by 100.
</summary>
</histogram>
<histogram name="WebRTC.Video.SentWidthInPixels" units="pixels"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
The average sent width per frame for a sent video stream. Recorded when a
stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.TargetDelayInMs" units="ms" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Average target delay (jitter delay + decode time + render delay) for a
received video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.TimeInBlockyVideoPercentage" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Percentage of time the receive video stream playbacks low quality blocky
video. Recorded when a stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.TimeInHdPercentage" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>ilnik@chromium.org</owner>
<owner>webrtc-video@google.com</owner>
<summary>
Percentage of time the receive video stream playbacks HD resolution.
Recorded when a stream is removed or content type changes.
</summary>
</histogram>
<histogram name="WebRTC.Video.UniqueNackRequestsReceivedInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of unique RTCP NACK requests that are received in response to a
sent video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.UniqueNackRequestsSentInPercent" units="%"
expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>asapersson@chromium.org</owner>
<summary>
Percentage of unique RTCP NACK requests that are sent in response to a
received video stream. Recorded when a stream is removed.
</summary>
</histogram>
<histogram name="WebRTC.Video.VP8DecoderImpl.TooManyPendingFrames"
units="counts" expires_after="never">
<!-- expires-never: WebRTC health metric. -->
<owner>perkj@chromium.org</owner>
<summary>
Counts occurences of if the VP8 software decoder runs out of buffers due to
that they are not returned to the buffer pool. See http://crbug/652923 and
http://crbug/542522.
</summary>
</histogram>
<histogram name="WebRTC.webkitApiCount" enum="RTCAPIName"
expires_after="2021-11-14">
<owner>guidou@chromium.org</owner>
<owner>hbos@chromium.org</owner>
<owner>mcasas@chromium.org</owner>
<owner>emircan@chromium.org</owner>
<owner>armax@chromium.org</owner>
<summary>Counts number of calls to WebRTC APIs from JavaScript.</summary>
</histogram>
<histogram name="WebRTC.webkitApiCountPerSession" enum="RTCAPIName"
expires_after="2022-03-07">
<owner>guidou@chromium.org</owner>
<owner>hbos@chromium.org</owner>
<summary>
Counts the number of calls to WebRTC APIs from JavaScript once per session.
A session is a crude estimate since its implemented as the lifetime of the
render process that called the WebRTC API.
</summary>
</histogram>
<histogram name="WebRTC.WindowCaptureTime" units="ms" expires_after="M77">
<owner>jiayl@chromium.org</owner>
<summary>Time for capturing one frame in window capturing.</summary>
</histogram>
<histogram name="WebRtcEventLogging.Api" enum="WebRtcEventLoggingApiEnum"
expires_after="2021-11-07">
<owner>eladalon@chromium.org</owner>
<owner>saeedj@google.com</owner>
<owner>manj@google.com</owner>
<owner>dmitriyg@google.com</owner>
<summary>
The result of calls to the API for the collection and uploading of WebRTC
event logs.
</summary>
</histogram>
<histogram name="WebRtcEventLogging.NetError" units="units"
expires_after="2021-07-07">
<owner>eladalon@chromium.org</owner>
<owner>saeedj@google.com</owner>
<owner>manj@google.com</owner>
<owner>dmitriyg@google.com</owner>
<summary>
NetError returned by the SimpleURLLoader object in charge of uploading a
WebRTC event log file.
</summary>
</histogram>
<histogram name="WebRtcEventLogging.Upload" enum="WebRtcEventLoggingUploadEnum"
expires_after="2021-11-14">
<owner>eladalon@chromium.org</owner>
<owner>saeedj@google.com</owner>
<owner>manj@google.com</owner>
<owner>dmitriyg@google.com</owner>
<summary>
Tracks the uploading or discarding of WebRTC event logs that were previously
collected.
</summary>
</histogram>
<histogram name="WebRtcTextLogging.UploadFailureNetErrorCode"
enum="NetErrorCodes" expires_after="2021-12-01">
<owner>guidou@chromium.org</owner>
<owner>olka@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Network error codes for WebRTC text log upload failures. Recorded when an
upload attempt fails.
</summary>
</histogram>
<histogram name="WebRtcTextLogging.UploadFailureReason"
enum="WebRtcLoggingUploadFailureReason" expires_after="2021-12-01">
<owner>guidou@chromium.org</owner>
<owner>olka@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Counts upload failures for WebRTC text log. Error codes for network errors
are logged in WebRtcTextLogging.UploadFailureNetErrorCode. Recorded when an
upload attempt fails.
</summary>
</histogram>
<histogram name="WebRtcTextLogging{WebRtcLoggingEvent}"
enum="WebRtcLoggingWebAppIdHash" expires_after="2021-12-01">
<owner>guidou@chromium.org</owner>
<owner>olka@chromium.org</owner>
<owner>webrtc-dev@chromium.org</owner>
<summary>
Counts the number of WebRTC text log events per web application. Suffixed by
event. {WebRtcLoggingEvent}
</summary>
<token key="WebRtcLoggingEvent">
<variant name="">
<obsolete>
Base histogram. Use suffixes of this histogram instead.
</obsolete>
</variant>
<variant name=".Discard" summary="Discard"/>
<variant name=".Start" summary="Start"/>
<variant name=".UploadFailed" summary="Upload failed"/>
<variant name=".UploadStarted" summary="Upload started"/>
<variant name=".UploadStoredStarted"
summary="Upload of a stored log started"/>
<variant name=".UploadSuccessful" summary="Upload successful"/>
</token>
</histogram>
</histograms>
</histogram-configuration>