RTCInboundRtpStreamStats: insertedSamplesForDeceleration property

The insertedSamplesForDeceleration property of the RTCInboundRtpStreamStats dictionary accumulates the difference between the number of samples received and the number of samples played out of the jitter buffer while audio playout is slowed down.

The WebRTC jitter buffer sets a target playout delay level such that the inflow and outflow of the jitter buffer should be approximately the same. If the jitter buffer empties too quickly the audio sample that is next in line to be output may be "ahead of schedule", and the jitter buffer may slow down playout. If the jitter buffer slows down the playout of the sample by inserting additional audio samples, this property indicates the accumulated number of such added samples.

Slowing down and/or speeding up the audio (as tracked with removedSamplesForAcceleration) may result in audible warbling or other distortion. The totals at the end of the call also give you some indication of how many samples or seconds were impacted, and insertedSamplesForDeceleration can be correlated with totalSamplesReceived to get a relative measure of deceleration. Logging insertedSamplesForDeceleration and removedSamplesForAcceleration in timeslices can be helpful for isolating the times at which the problem occurred and you can then correlate other metrics in the same timeslice to determine likely causes.