c160-3ay@ucbzooey.BERKELEY.EDU (Scrooge) (12/15/85)
In article <701@abic.UUCP> mjm@abic.UUCP (Mark Medovich) writes: > Adaptive Delta Modulation seems to be a nice idea, but I've found that >no manufacturer is using this technique with the exception of Delta Lab >Inc. My impression is that the sample frequency is a function of the slope >of the input, (and hence, d(-)/dt?). Can you tell me more?... > ...I think algorithms of >this sort will reduce processing time and conserve memory. I can see how such a technique would conserve memory, but it seems to me that if you wanted to manipulate the sampled sound in any way (digital filtering, pitch change, etc) the algorithms would have to be much more complicated to deal with this more complex method of storing information. If the sampling frequency is constant, that's one less thing for the algorithm to deal with. Ranjit Bhatnagar (PLEASE mail to Disclaimer: I don't know anything ...ucbvax!bugs!ranjit, not about ad. del. mod.! to c160-3ay)