Path: utzoo!utgpu!news-server.csri.toronto.edu!bonnie.concordia.ca!uunet!valid!caber!lou From: lou@caber.valid.com (Louis K. Scheffer) Newsgroups: comp.dsp Subject: Re: 48k to 44.1k sample rate conversion Message-ID: <500@valid.valid.com> Date: 27 May 91 00:49:36 GMT References: <5826@media-lab.media.mit.edu.MEDIA.MIT.EDU> <1991May13.173129.18295@bernina.ethz.ch> <1991May15.105543.12165@bernina.ethz.ch> <1991May16.052920.21060@netcom.COM> <481@valid.valid.com> <4035@motcsd.csd.mot.com> Sender: news@valid.com Lines: 22 lance@motcsd.csd.mot.com (lance.norskog) writes: >Ok, now we're getting somewhere. The SPUT sound operating system >for DOS (shareware by Adrienne Cousins) has high-pass and low-pass >filtration options for playing back sound samples. SPUT is a DOS >TSR (background program) comprising 5K of 8086 code, so we know >it doesn't have 13,000 coefficients. >If interpolation is being used for the low-pass operation, is it >also being used for the high-pass problem? I don't know this software, so I'm guessing. It is probably not interpolation, but a simple digital filter. These work by taking the last N samples, multiplying them by a bunch of constants a[1],...,a[n] and adding the results. You can (optionally) take the computed output and use it in the computation on the next sample as well. You only need a few coefficients (maybe 10) to get low-pass, high-pass, and bandpass responses, with responses like analog audio equipment. The zillion coefficient case is only when you are trying to do something tricky. -Lou Scheffer