webrtc-stats
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w3c
WebRTC Statistics
The current tests are happy with either video or audio metrics, but some are only defined for one of the kinds. Having WPTs have separate tests for different kinds would...
video frame maybe dropped prior to encode, [according to webrtc code](https://chromium.googlesource.com/external/webrtc/+/master/api/video_codecs/video_encoder.h#65) the capture video frame maybe dropped by encoder's internal rate limiter or MediaOptimizations , but no stats about this...
Check the [summary table](https://www.w3.org/TR/webrtc-stats/#summary), "inbound-rtp", and you see the field `kind` listed twice. `RTCInboundRtpStreamStats` contains the field `kind`. `RTCInboundRtpStreamStats` dictionary extends `RtcReceivedRtpStreamStats` `RtcReceivedRtpStreamStats` dictionary extends `RtcRtpStreamStats` `RtcRtpStreamStats` dictionary contains `kind`...
It seems there is no field describe packet loss rate of specfic audio/video inbound rtp stream, the fractionLost field exist in [RTCRemoteInboundRtpStreamStats](https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats), but it‘s about outbound loss rate. packet loss...
What happens if the source is stereo? We should be consistent in all the audio metrics on how stereo audio streams are handled.
Transferred from https://github.com/w3c/webrtc-svc/issues/72 It would be nice to have SVC-related fields in WebRTC stats. For example, adding the DOMString scalabilityMode field of RTCOutboundRtpStreamStats to indicate which scalability mode is used...
This was previously discussed in #550. The reason for creating a separate issue specifically for `codec` is that this information is already exposed in `RTCRtpSender/RTCRtpReceiver.getCapabilities(kind).codecs`. Therefore it is likely that...
The MDN docs for [RTCStats.timestamp](https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidatePairStats/timestamp) say this for the value: > A [DOMHighResTimeStamp](https://developer.mozilla.org/en-US/docs/Web/API/DOMHighResTimeStamp) value indicating the time at which the activity described by the statistics in this object was recorded,...
Spec for `RTCCodecStats.clockRate` says it is the "media sampling rate". However the parameter that is set is defined in `RTCRtpCodec.clockRate` as _The codec clock rate expressed in Hertz_, which on...
This [language on RTP stats](https://w3c.github.io/webrtc-stats/#the-rtp-statistics-hierarchy) seems to go against https://github.com/w3c/webrtc-stats/issues/26: _"RTP monitored objects are deleted when the corresponding RTP sender or RTP receiver is reconfigured to remove the corresponding RTP...
