Latency in Speech Feature Analysis for Telepresence Event Coding

For videoconferencing, there are network bandwidth and screen real-estate constraints that limit the number of user channels. We propose an intermediate transmission mode that transmits only at "events", where these are detected by both audio and video changes from the short-term signal average. Our objective in this paper is to determine latency until the audio portion of a single telepresence channel stabilizes. It is this stable signal from which we detect events. We describe a recursive filter approach for feature determination and experiments on the Switchboard telephone call database. Results show latency to stable signal of up to 10 seconds. Although events can be detected much more quickly (<;;1 sec) if the signal is already stable, this latency time must be considered at the start of a conversation or at changes in the short-term average.