ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com ilnik@chromium.org webrtc-video@google.com silen@chromium.org alessiob@chromium.org webrtc-audio-uma@google.com Logs the input clipping rate in AGC1 and in AGC2. A log call is made every 30 seconds during an active WebRTC call using the input volume controller. alessiob@chromium.org silen@chromium.org webrtc-audio-uma@google.com 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. alessiob@chromium.org silen@chromium.org webrtc-audio-uma@google.com 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). Warning: this histogram was expired from 2021-12-31 to 2022-11-16; data may be missing. alessiob@chromium.org silen@chromium.org webrtc-audio-uma@google.com How long the AGC2 limiter remained in the {Region} region. Logged during an active WebRTC call whenever the limiter switches region and at the end of the call. hlundin@chromium.org webrtc-audio-uma@google.com 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 "true" meaning that the gain was in fact adjusted in response to the detected clipping, and "false" meaning that adjustment was not allowed due to limiting boundaries in the algorithm. silen@chromium.org alessiob@chromium.org webrtc-audio-uma@google.com The recommended input volume when adjusted to match the target level - i.e., not adjusted to handle clipping input. A log call is made every time that such a volume changes during an active WebRTC call. silen@chromium.org alessiob@chromium.org webrtc-audio-uma@google.com The {InputVolumeType} input volume. A log call is made every time that the volume changes during an active WebRTC call. silen@chromium.org alessiob@chromium.org webrtc-audio-uma@google.com The {InputVolumeType} input volume average {Metric} measured over 60 seconds in AudioProcessingImpl. A log call is made every 60 seconds during an active WebRTC call. silen@chromium.org alessiob@chromium.org webrtc-audio-uma@google.com The {InputVolumeType} input volume {Metric} rate measured over 60 seconds in AudioProcessingImpl. A log call is made every 60 seconds during an active WebRTC call. hlundin@chromium.org webrtc-audio-uma@google.com 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). hlundin@chromium.org webrtc-audio-uma@google.com 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). peah@chromium.org webrtc-audio-uma@google.com 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). peah@chromium.org webrtc-audio-uma@google.com 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). hlundin@chromium.org ivoc@chromium.org webrtc-audio-uma@google.com 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. fhernqvist@google.com olka@chromium.org webrtc-audio-uma@google.com Logs the AudioMixer source count whenever the mixer gets a new highest source count. This allows us to differentiate the histogram to get the number of mixers that have had a specific number of maximum sources. For example, the number of AudioMixers that have had a maximum of 5 sources is NewHighestSourceCount[5] - NewHighestSourceCount[6]. Note that this is not the same as logging the source count whenever a source is added; logging only happens when the source count exceeds the previously highest source count. hlundin@chromium.org gustaf@chromium.org webrtc-audio-uma@google.com 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. hlundin@chromium.org webrtc-audio-uma@google.com 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. hlundin@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com This histogram logs half the applied render buffer delay used in the WebRTC echo canceller. The value 0 means that no delay could be estimated, otherwise the logged delay corresponds to the actual delay + 1. 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. gustaf@chromium.org peah@chromium.org webrtc-audio-uma@google.com This histogram logs whether clockdrift is detected in the WebRTC echo canceller. A new value is logged every 10 seconds. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com This histogram logs half the estimated echo path delay in 64 sample blocks as seen by the delay estimator in the WebRTC echo canceller. The value 0 means that no delay could be estimated, 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 delay of the echo path at the time when the value is logged. gustaf@chromium.org peah@chromium.org webrtc-audio-uma@google.com 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} gustaf@chromium.org peah@chromium.org webrtc-audio-uma@google.com 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} peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org gustaf@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org gustaf@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org gustaf@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org gustaf@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com This histogram logs whether the WebRTC echo canceller has encountered persistent multichannel playout reference content during its lifetime. The value is logged at the end of the AEC lifetime if the AEC has processed at least 5 seconds of audio. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com This histogram logs the frequency of persistent multichannel playout reference content detected by the WebRTC echo canceller. `true` if a majority of the last 10 seconds of audio was classified as persistent multichannel. A new value is logged every 10 seconds. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. peah@chromium.org saza@chromium.org webrtc-audio-uma@google.com 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. hlundin@chromium.org jakobi@google.com tomasl@google.com webrtc-audio-uma@google.com Histogram of audio codec usage. Every sample corresponds to 5 seconds of encoding with that codec. hlundin@chromium.org webrtc-audio-uma@google.com 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. hlundin@chromium.org webrtc-audio-uma@google.com The sum of the jitter buffer delay and the sound card's buffering delay for the receiving side. That is, the sum of the metrics WebRTC.Audio.ReceiverDeviceDelayMs and WebRTC.Audio.ReceiverJitterBufferDelayMs. Sampled once every 10 ms when WebRTC audio is playing. Warning: this histogram was expired from 2020-07-06 to 2022-08-25; data may be missing. hlundin@chromium.org webrtc-audio-uma@google.com The sound card's buffering delay for the receiving side. Sampled once every 10 ms when WebRTC audio is playing. hlundin@chromium.org webrtc-audio-uma@google.com The jitter buffer delay for the receiving side. Sampled once every 10 ms when WebRTC audio is playing. hlundin@chromium.org webrtc-audio-uma@google.com 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. hlundin@chromium.org webrtc-audio-uma@google.com The target bitrate in kbps that the audio codec should try to produce on average. Sampled every time the rate is updated. hlundin@chromium.org webrtc-audio-uma@google.com The target jitter buffer delay for the receiving side. Sampled once every 10 ms per WebRTC receive stream when WebRTC audio is playing. saza@chromium.org olka@chromium.org webrtc-audio-uma@google.com Audio input channel layout in WebRTC. Logged when a processed audio input source is started, roughly once per successful getUserMedia call with some effect enabled. Warning: this histogram was expired from 2020-11-01 to 2022-04-11; data may be missing. saza@chromium.org olka@chromium.org webrtc-audio-uma@google.com Audio input sample rate for WebRTC (in Hz). Logged when a processed audio input source is started, roughly once per successful getUserMedia call with some effect enabled. Warning: this histogram was expired from 2020-11-29 to 2022-04-11; data may be missing. saza@chromium.org olka@chromium.org webrtc-audio-uma@google.com Audio input sample rate for WebRTC (atypical values, in Hz). Logged when a processed audio input source is started, roughly once per successful getUserMedia call with some effect enabled. Warning: this histogram was expired from 2021-04-30 to 2022-04-11; data may be missing. saza@chromium.org olka@chromium.org webrtc-audio-uma@google.com Audio output sample rate for WebRTC (in Hz). Logged when a WebRTC MediaStream is connected to an audio output device. saza@chromium.org olka@chromium.org webrtc-audio-uma@google.com Audio output sample rate for WebRTC (atypical values, in Hz). Logged when a WebRTC MediaStream is connected to an audio output device. Warning: this histogram was expired from 2021-04-30 to 2022-12-07; data may be missing. holmer@chromium.org The bandwidth estimate 2 seconds into a WebRTC call. holmer@chromium.org The number of video packets lost durig the first 2 seconds in a WebRTC call. holmer@chromium.org The round-trip time as measured 2 seconds into a WebRTC call. holmer@chromium.org 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. holmer@chromium.org 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. holmer@chromium.org 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. holmer@chromium.org 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. holmer@chromium.org webrtc-dev@chromium.org 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. Warning: This histogram was expired from 2022-10-11 to 2022-12-15. Data may be missing. holmer@chromium.org 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. holmer@chromium.org 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. holmer@chromium.org 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. asapersson@chromium.org The lifetime of a call. Recorded when a Call instance is destroyed. holmer@chromium.org 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. holmer@chromium.org 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. saza@chromium.org hlundin@chromium.org 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. saza@chromium.org hlundin@chromium.org 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. holmer@chromium.org 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. orphis@chromium.org toprice@chromium.org Recorded when a data channel is created. Counts the combination of reliable and ordered modes in a data channel. orphis@chromium.org toprice@chromium.org Recorded when a data channel is created. Counts a few main attributes of data channels. Expired in April 2020. Revived in M93. orphis@chromium.org toprice@chromium.org Recorded when a data channel is created. 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. orphis@chromium.org toprice@chromium.org Recorded when a data channel is created. 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. Expired in M86. Revived in M93. orphis@chromium.org toprice@chromium.org Recorded when a data channel is closed unexpectedly and the error event is dispatched. Contains the SCTP error code signaled by the transport. henrika@chromium.org webrtc-dev@chromium.org A boolean is recorded to this histogram for each provided video frame where the desktop media source is given by {CapturerType}. Sampled as true if the provided frame is a results of a client/sink asking for a refresh frame using the RequestRefreshFrame() API, and sampled as false when the captured frame is newly captured frame not explicitly requested by the client/sink. Very few frames are expected to be marked as refresh frames. henrika@chromium.org webrtc-dev@chromium.org An integer value is recorded to this histogram for each captured video frame where the desktop media source is of type {CapturerType}. The value is an estimation of the current frame capture rate and the unit is in frames per second. henrika@chromium.org webrtc-dev@chromium.org A boolean is recorded to this histogram for each captured video frame where the desktop media source is given by {CapturerType}. Only recorded for capturers that support detection of whether 0Hz is active or not which requires that each captured frame must contain a representation of what regions have been updated since the last frame. Windows WGC is an example of a capturer which does not support this. henrika@chromium.org webrtc-dev@chromium.org An integer value is recorded to this histogram each time a client requests a refresh frame using the RequestRefrestFrame() API where the desktop media source is of type {CapturerType}. The value is an estimation of the current refresh-frame rate and the unit is in frames per second. alcooper@chromium.org henrika@chromium.org edgecapabilitiesdev@microsoft.com This measures the frequency of use for each desktop capturer implementation, allowing us to measure the adoption of the WGC capturer. henrika@chromium.org webrtc-dev@chromium.org This records high level errors, or success, encountered across the entire capture flow in the DirectX (DXGI) based capturer. Is recorded every time a new video frame is captured using DXGI. Only recorded on Windows. henrika@chromium.org webrtc-dev@chromium.org A boolean is recorded to this histogram for each captured video frame where the desktop media source is the full screen and a change has been detected in the mouse cursor position. The value is true when the cursor pointer is embedded in the captured desktop image and false if not. It is expected that most display adapters will report false. Only recorded when DirectX (DXGI) is used on Windows. alcooper@chromium.org henrika@chromium.org edgecapabilitiesdev@microsoft.com This records high level errors, or success, encountered across the entire capture flow in the WGC based capturer. alcooper@chromium.org henrika@chromium.org edgecapabilitiesdev@microsoft.com This records the result from retrieving a frame from the WGC based capturer. alcooper@chromium.org henrika@chromium.org edgecapabilitiesdev@microsoft.com This records the result from starting up the WGC based capturer. alcooper@chromium.org henrika@chromium.org edgecapabilitiesdev@microsoft.com This measures the performance of the {Capturer} capturer. toprice@chromium.org webrtc-dev@chromium.org Counters on creation of DesktopCaptureDevice and the first capture call. zhihuang@chromium.org Counters on different types of TCP socket error code. Collected when we hit the error code when writing. Removed 03/2022. reillyg@chromium.org device-dev@chromium.org 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. handellm@chromium.org webrtc-dev@chromium.org This histogram logs how many received UDP packets are batched in a single P2PSocketClient IPC. This is recorded every time the batched received packets transmit over IPC. handellm@chromium.org webrtc-dev@chromium.org This histogram logs how many sent UDP packets are batched in a single P2PSocket IPC. This is recorded every time a batch is transmitted over IPC. hta@chromium.org webrtc-dev@chromium.org Outcomes of adding ICE candidates to a PeerConnection. Used to check the theory that failures in candidate addition are ignored by applications. hta@chromium.org webrtc-dev@chromium.org Determines whether BUNDLE is configured. Recorded during the first DTLS connection establishment. Values are specified in https://w3c.github.io/webrtc-pc/#dom-rtcbundlepolicy Warning: This histogram was expired from 2022-08-28 to 2022-10-10. Data may be missing. hta@chromium.org webrtc-dev@chromium.org 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. Warning: This histogram was expired from 2022-08-28 to 2022-10-10. Data may be missing. hta@chromium.org webrtc-dev@chromium.org phancke@microsoft.com Counters of various ICE Endpoint types. These values are logged for the first selected candidate pair of a PeerConnection. {IPProtocolType} Warning: this histogram was expired from 2020-04-05 to 2023-06-16; data may be missing. This measurement showed that numbers higher than 1 are in frequent use for all policies. Experiment is no longer needed. hta@chromium.org webrtc-dev@chromium.org 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. hta@chromium.org webrtc-dev@chromium.org 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. Warning: This histogram was expired from 2022-08-21 to 2022-10-29. Data may be missing. zhihuang@chromium.org Records the error whenever the Dtls handshake fails. There are only two types of errors, incompatitable cipher suite and unknown error, for now. handellm@chromium.org webrtc-dev@chromium.org 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()|). handellm@chromium.org webrtc-dev@chromium.org 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. handellm@chromium.org webrtc-dev@chromium.org 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()|). honghaiz@chromium.org Records the reasons for ICE re-gathering. honghaiz@chromium.org Records the transport channel states when ICE restart happens. Removed 10/2022. hta@chromium.org webrtc-dev@chromium.org Records the number of configured ICE servers when the peerconnection is created or gets connected, depending on {MeasuredAt}. hta@chromium.org webrtc-dev@chromium.org phancke@microsoft.com Counters on IPv4 and IPv6 usage in PeerConnection. These values are logged once per PeerConnection. Warning: this histogram was expired from M81 to 2023-06-16; data may be missing. hta@chromium.org webrtc-dev@chromium.org Number of IPv4 network interfaces discovered in a PeerConnection Session. Warning: This histogram was expired from M81 to M110 Canary (2022-10-29). Data may be missing. hta@chromium.org webrtc-dev@chromium.org Number of IPv6 network interfaces discovered in a PeerConnection Session. Warning: This histogram was expired from M81 to M110 Canary (2022-10-29). Data may be missing. The SDES protocol is no longer available in Chrome as of M98. DTLS is the only possible value. hta@chromium.org webrtc-dev@chromium.org 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. handellm@chromium.org webrtc-dev@chromium.org 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()|). handellm@chromium.org webrtc-dev@chromium.org 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()|). handellm@chromium.org webrtc-dev@chromium.org 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()|). Removed 12/2022. hta@chromium.org webrtc-dev@chromium.org Records the length of the SDP mid attribute in order to limit its length to 16 characters. hta@chromium.org webrtc-dev@chromium.org Whether provisional answers are used. Recorded during the first DTLS connection establishment. hta@chromium.org webrtc-dev@chromium.org 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. Warning: This histogram was expired from 2020-02-23 to 2022-10-10. Data may be missing. hta@chromium.org webrtc-dev@chromium.org phancke@microsoft.com Which RTCPMuxPolicy was requested for the PeerConnection. Recorded during the first DTLS connection establishment. Removed 04/2022. hbos@chromium.org webrtc-dev@chromium.org The SDP usage category ("safe", "unsafe" or "unknown") 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. Removed 04/2022. hbos@chromium.org webrtc-dev@chromium.org The SDP usage category ("safe", "unsafe" or "unknown") 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. Removed 04/2022. hbos@chromium.org webrtc-dev@chromium.org The SDP usage category ("safe", "unsafe" or "unknown") 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. Removed 04/2022. hbos@chromium.org webrtc-dev@chromium.org The SDP usage category ("safe", "unsafe" or "unknown") 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. Removed 04/2022. hbos@chromium.org webrtc-dev@chromium.org The SDP usage category ("safe", "unsafe" or "unknown") 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. Removed 04/2022. hbos@chromium.org webrtc-dev@chromium.org The SDP usage category ("safe", "unsafe" or "unknown") 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. Removed 09/2022. steveanton@chromium.org hta@chromium.org webrtc-dev@chromium.org What SDP format is received in the remote offer. The value "no tracks" means that no audio or video tracks were received. The value "simple" means that at most one audio and at most one video track was received. The value "complex" means that more than one audio or more than one video track was received, and how this was signaled is indicated ("Plan B" meaning with a=ssrc lines within the same m= section and "Unified Plan" meaning with a separate m= section). This is recorded when calling setRemoteDescription with an SDP Offer. Removed 09/2022. steveanton@chromium.org hta@chromium.org webrtc-dev@chromium.org 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. Removed 09/2022. hta@chromium.org webrtc-dev@chromium.org 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 "mixed" means that the accepted answer included both types of marker in the SDP. The value "none" will happen when the answerer sends no media. amithi@chromium.org Counts the number of send encodings given to PeerConnection::AddTransceiver. This histogram will be used to understand if and how the API is used. handellm@google.com hbos@chromium.org Measures computer speed limit state, sampled every 60s when a PeerConnection is open. The value indicates the current operating point in percent of the nominal capacity of the CPUs. The computer decreases the capacity during thermal overload events. steveanton@chromium.org What error code is reported by libsrtp when failing to unprotect an incoming SRTCP (secured media control) packet. steveanton@chromium.org What error code is reported by libsrtp when failing to unprotect an incoming SRTP (secured media) packet. hta@google.com webrtc-dev@chromium.org Counters on the type of SSL cipher suites used by WebRTC. This is collected whenever the transport signals the OnCompleted event. {WebRTCMediaType} eshr@google.com hbos@chromium.org 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. handellm@google.com hbos@chromium.org Sampled every 60s when the peer connection is open. True when the computer's speed limit is between 0 and 99 inclusively, false otherwise. handellm@google.com hbos@chromium.org The number of thermal throttling episodes the peer connection encountered. Reported when the peer connection is torn down if the peer connection had at least one track. hta@chromium.org webrtc-dev@chromium.org 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. hbos@chromium.org webrtc-dev@chromium.org phancke@microsoft.com Measures whether the remote description contains a payload type collision that is not allowed by the IETFs BUNDLE specification. Enforcing this would allow simplifying the statistics collector. Recorded for every remote description. hta@chromium.org webrtc-dev@chromium.org phancke@microsoft.com Measures whether the ice-ufrag/ice-pwd consists of characters allowed by the specification. Currently some characters outside the range are allowed for backward compability with deployed software. Recorded during the first DTLS connection establishment. handellm@chromium.org hta@chromium.org 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. Warning: this histogram was expired between 2023-04-02 to 2023-04-19; data may be missing. handellm@chromium.org hta@chromium.org 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. Warning: this histogram was expired from 2022-11-27 to 2022-12-10 and from 2023-04-02 to 2023-04-19; data may be missing. perkj@chromium.org 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. jiayl@chromium.org Time for capturing one frame in screen capturing. kron@chromium.org webrtc-video@google.com Tracks the usage of the WebRTC Fullscreen Detector. The Fullscreen Detector is used when capturing presentation applications such as PowerPoint. If the editor window is captured and a slideshow is started in fullscreen mode, the Fullscreen Detector will automatically change so that the fullscreen window is captured. Once the slideshow stops, the editor will be captured again. This is recorded the first time this feature is activated, but only once per screensharing session. handellm@google.com ilnik@chromium.org The number of milliseconds from the point a zero-Hz frame cadence adapter is created until the first frame arrives from a connected source. handellm@chromium.org hta@chromium.org 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. handellm@chromium.org hta@chromium.org 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. qingsi@google.com jeroendb@google.com 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} hta@chromium.org webrtc-dev@chromium.org Records outcome of integrity checks for STUN requests. This is used in investigation of crbug.com/1177125, in order to figure out if it is safe to turn on checking of these attributes. qingsi@google.com jeroendb@google.com 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} qingsi@google.com jeroendb@google.com 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} perkj@chromium.org 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. andresp@chromium.org toprice@chromium.org agpalak@chromium.org benjaminwagner@google.com The state of a UserMediaRequest when it gets destroyed before having a result. Warning: this histogram was expired from M77 to M108(excluded); data may be missing. Note: "Explicitly Cancelled" means MediaStreamImpl::cancelUserMediaRequest was called and not necessarily that the user cancelled. Those are likely tracked as UserMediaRequest with a result of permission denied. toprice@chromium.org agpalak@chromium.org Counters for UserMediaRequests results such as failure reasons. The standard specification error names are in parenthesis. asapersson@chromium.org webrtc-video@google.com The average number of adaptation changes per minute for a sent video stream. Recorded when a stream is removed. {VideoAdaptationReason} holmer@chromium.org The average round-trip time of a WebRTC call in milliseconds. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org Percentage of sent frames that are limited in resolution due to bandwidth for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org webrtc-video@google.com 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. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org Percentage of frames that are limited in resolution due to CPU for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org The average QP (quantizer value) per frame for a received VP8 video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} asapersson@chromium.org 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. asapersson@chromium.org The average decode time per frame for a received video stream. Recorded when a stream is removed. asapersson@chromium.org Percentage of delayed frames to renderer for a received video stream. Recorded when a stream is removed. asapersson@chromium.org The average delay of delayed frames to renderer for a received video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by a capturer for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by an encoder's internal rate limiter for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped because encoder queue is full for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by a WebRTC rate limiter (in MediaOpt) for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by a WebRTC on the receive side because they are incomplete or undecodable. asapersson@chromium.org The average QP (quantizer value) per frame for a sent video stream. Recorded when a stream is removed. {VideoEncodedQpStats} asapersson@chromium.org webrtc-video@google.com Configured video codec for a sent video stream. Recorded when a VideoSendStream is destroyed (for streams whose lifetime is longer than 10 seconds). asapersson@chromium.org The average encode time per frame for a sent video stream. Recorded when a stream is removed. sprang@chromium.org webrtc-video@google.com webrtc-dev@chromium.org The average bitrate overshoot percent for a sent stream. Recorded when a stream is removed. asapersson@chromium.org The average end-to-end delay per frame for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The maximum end-to-end delay per frame for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. henrika@chromium.org webrtc-video@google.com The time it took between a video frame was captured until it started being encoded. Recorded on the receiving side when a video frame has been decoded using timestamps generated on the sending side if the VideoSendTiming flag is set. henrika@chromium.org webrtc-video@google.com The time it took to decode a video frame. Recorded on the receiving side when a video frame has been decoded using locally generated timestamps. henrika@chromium.org webrtc-video@google.com The time it took to encode a video frame. Recorded on the receiving side when a video frame has been decoded using timestamps generated on the sending side if the VideoSendTiming flag is set. henrika@chromium.org webrtc-video@google.com Total time it took to compensate for network jitter of a video frame. Recorded on the receiving side when a video frame has been decoded using locally generated timestamps. henrika@chromium.org webrtc-video@google.com The time a video frame spent in the pacer and packetizer before being transmitted. Recorded on the receiving side when a video frame has been decoded using timestamps generated on the sending side if the VideoSendTiming flag is set. henrika@chromium.org webrtc-video@google.com The time it took to receive a full video frame. Recorded on the receiving side when a video frame has been decoded using locally generated timestamps. hbos@chromium.org 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). hbos@chromium.org 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). kron@chromium.org webrtc-video@google.com The number of hardware decoded frames between fallbacks to software decoder for a received video stream. {WebRtcCodecs} asapersson@chromium.org 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. asapersson@chromium.org The average input height per frame (for incoming frames to video engine) for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org The average input width per frame (for incoming frames to video engine) for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com The 95th percentile of interframe delay for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The average interframe delay for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The maximum interframe delay for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} asapersson@chromium.org Average jitter buffer delay for a received video stream. Recorded when a stream is removed. asapersson@chromium.org Permille of frames that are key frames for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} asapersson@chromium.org Permille of frames that are key frames for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com The average duration of a freeze in video playback. Recorded when a received stream is removed or content type changes. ilnik@chromium.org webrtc-video@google.com The average duration of a smooth video playback. Recorded when a received stream is removed or content type changes. asapersson@chromium.org 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} asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. ssilkin@chromium.org The number of video freezes per minute for a received video stream. Recorded when a stream is removed or content type changes. asapersson@chromium.org The number of times a video stream has been paused/resumed during a call. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com The average number of resolution down-switches per minute for a received video stream. Recorded when a stream is removed or content type changes. asapersson@chromium.org Average delay (network delay (rtt/2) + jitter delay + decode time + render delay) for a received video stream. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org Percentage of time that the video has been paused for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org Percentage of sent frames that are downscaled in resolution due to quality for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org Percentage of received FEC packets for a received video stream. Recorded when a stream is removed. asapersson@chromium.org The average received height per frame for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} asapersson@chromium.org Percentage of received packets lost for a received video stream. Recorded when a stream is removed. asapersson@chromium.org The average received width per frame for a received video stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} asapersson@chromium.org The lifetime of a video receive stream. Recorded when a VideoReceiveStream instance is destroyed. asapersson@chromium.org Percentage of recovered media packets from FEC packets for a received video stream. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. sprang@chromium.org webrtc-video@google.com webrtc-dev@chromium.org The RMSE(Root Mean Square Error) between sent bits per second for a sent stream and target bitrate set by WebRTC. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. orphis@chromium.org toprice@chromium.org Counts usage of each scalability mode used to configure the video senders in RTCRtpEncodingParameters from addTransceiver and setParameters APIs. This is part of the WebRTC-SVC specification. sprang@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. ilnik@chromium.org webrtc-video@google.com The average QP (quantizer value) per frame for a received VP8 screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by a capturer for a sent screenshare stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by an encoder's internal rate limiter for a sent screenshare stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped because encoder queue is full for a sent screenshare stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Total number of frames dropped by a WebRTC rate limiter (in MediaOpt) for a sent screenshare stream. Recorded when a stream is removed. sprang@chromium.org The average encode time per frame for a sent (screen content) video stream. Recorded when a stream is removed. sprang@chromium.org webrtc-video@google.com webrtc-dev@chromium.org The average bitrate overshoot percent for a sent screenshare stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com The average end-to-end delay per frame for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The maximum end-to-end delay per frame for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} sprang@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. asapersson@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. ilnik@chromium.org webrtc-video@google.com The 95th percentile of interframe delay for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The average interframe delay for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The maximum interframe delay for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com Permille of frames that are key frames for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} sprang@chromium.org Permille of frames that are key frames for a sent (screen content) video stream. Recorded when a stream is removed. sprang@chromium.org Stats for the lower layer (TL0) of a screenshare stream in conference mode. {ScreenshareLayerStats} sprang@chromium.org Stats for the higher layer (TL1) of a screenshare stream in conference mode. {ScreenshareLayerStats} ilnik@chromium.org webrtc-video@google.com The average duration of a freeze in screenshare playback. Recorded then a received stream is removed or content type changes. ilnik@chromium.org webrtc-video@google.com The average duration of a smooth video playback for screenshare stream. Recorded when a received stream is removed or content type changes. ilnik@chromium.org webrtc-video@google.com 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} sprang@chromium.org 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. sprang@chromium.org 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. Removed 12/2022. ilnik@chromium.org webrtc-video@google.com The average number of resolution down-switches per minute for a receive screenshare stream. Recorded when a stream is removed or content type changes. sprang@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. ilnik@chromium.org webrtc-video@google.com The average received height per frame for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com Percentage of received packets lost for a received screenshare stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com The average received width per frame for a received screenshare stream. Recorded when a stream is removed. {WebRTCVideoExperimentGroupId} ilnik@chromium.org webrtc-video@google.com The lifetime of a screenshare receive stream. Recorded when a VideoReceiveStream instance is destroyed. sprang@chromium.org 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. sprang@chromium.org webrtc-video@google.com webrtc-dev@chromium.org The RMSE(Root Mean Square Error) between sent bits per second for a sent screenshare stream and target bitrate set by WebRTC. Recorded when a stream is removed. sprang@chromium.org 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. sprang@chromium.org 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. sprang@chromium.org 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. ilnik@chromium.org webrtc-video@google.com 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. sprang@chromium.org The average sent height per frame for a sent (screen content) video stream. Recorded when a stream is removed. sprang@chromium.org Percentage of sent packets lost for a sent screenshare stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com 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. sprang@chromium.org The average sent width per frame for a sent (screen content) video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Percentage of time the received screenshare stream playbacks low quality blocky video. Recorded when a stream is removed or content type changes. ilnik@chromium.org webrtc-video@google.com Percentage of time the received screenshare stream playbacks HD resolution. Recorded when a stream is removed or content type changes. sprang@chromium.org Percentage of unique RTCP NACK requests that are received in response to a sent screenshare stream. Recorded when a stream is removed. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org 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. asapersson@chromium.org The lifetime of a video send stream. Recorded when a VideoSendStream instance is destroyed. asapersson@chromium.org 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. asapersson@chromium.org The average sent height per frame for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org Percentage of sent packets lost for a sent video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com 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. asapersson@chromium.org The average sent width per frame for a sent video stream. Recorded when a stream is removed. asapersson@chromium.org Average target delay (jitter delay + decode time + render delay) for a received video stream. Recorded when a stream is removed. ilnik@chromium.org webrtc-video@google.com Percentage of time the receive video stream playbacks low quality blocky video. Recorded when a stream is removed or content type changes. ilnik@chromium.org webrtc-video@google.com Percentage of time the receive video stream playbacks HD resolution. Recorded when a stream is removed or content type changes. asapersson@chromium.org Percentage of unique RTCP NACK requests that are received in response to a sent video stream. Recorded when a stream is removed. asapersson@chromium.org Percentage of unique RTCP NACK requests that are sent in response to a received video stream. Recorded when a stream is removed. perkj@chromium.org 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. toprice@chromium.org hbos@chromium.org mcasas@chromium.org emircan@chromium.org armax@chromium.org Counts number of calls to WebRTC APIs from JavaScript. toprice@chromium.org hbos@chromium.org 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. eladalon@chromium.org manj@google.com dmitriyg@google.com The result of calls to the API for the collection and uploading of WebRTC event logs. eladalon@chromium.org manj@google.com dmitriyg@google.com NetError returned by the SimpleURLLoader object in charge of uploading a WebRTC event log file. eladalon@chromium.org manj@google.com dmitriyg@google.com Tracks the uploading or discarding of WebRTC event logs that were previously collected. toprice@chromium.org olka@chromium.org webrtc-dev@chromium.org Network error codes for WebRTC text log upload failures. Recorded when an upload attempt fails. Note: The histogram was disabled between May 2022 and M110, so has no data for that period. toprice@chromium.org olka@chromium.org webrtc-dev@chromium.org Counts upload failures for WebRTC text log. Error codes for network errors are logged in WebRtcTextLogging.UploadFailureNetErrorCode. Recorded when an upload attempt fails. toprice@chromium.org olka@chromium.org webrtc-dev@chromium.org Counts the number of WebRTC text log events per web application. Suffixed by event. {WebRtcLoggingEvent}