Path: utzoo!utgpu!news-server.csri.toronto.edu!cs.utexas.edu!sun-barr!lll-winken!unixhub!shelby!csli!poser From: poser@csli.Stanford.EDU (Bill Poser) Newsgroups: comp.dsp Subject: Re: A simple, practical sound board Message-ID: <16113@csli.Stanford.EDU> Date: 31 Oct 90 08:33:36 GMT References: <15912@netcom.UUCP> Reply-To: poser@csli.stanford.edu (Bill Poser) Organization: Center for the Study of Language and Information, Stanford U. Lines: 25 There is no way you can get an hour of speech into a megabyte with reasonable quality if you just digitize waveforms. Suppose you sample at a resolution of 8 bits, which is the resolution of the cheapo ADDACs you can buy for PCs and Macs. For research purposes and hi-fi people use higher resolution. That means your megabyte gets you 1024000 samples. At 3600 seconds in an hour, that means 284.44 samples per second is the maximum sampling rate you can use. The corresponding Nyquist frequency is 142.22 Hz, meaning that you can only represent frequencies below this level. This is WAY too low. For music people use sampling rates around 44K samples/sec in order to get frequencies up to over 20KHz. For speech you don't need anything that high. For speech research we typically sample at 20K samples/sec and low pass filter at 8KHz. That covers everything significant for speech. For some purposes we sample at 10KHz with low-pass filtering at 4KHz. Engineers often sample at 8K samples/sec because they expect to be working with telephone speech, which is limited to the region below about 3200 Hz. Already we're talking about degraded, though intelligible, speech. So you can see that if you just want to record and edit waveforms, there is no way you can cram an hour of speech into a megabyte. To store speech at around 2000 bits/second as you wish to do is possible but requires non-trivial coding, and I'm not sure that you will like the quality that results. I think you're going to need a bigger disk.