Xref: utzoo sci.electronics:18070 comp.dsp:1305 Path: utzoo!utgpu!news-server.csri.toronto.edu!bonnie.concordia.ca!ccu.umanitoba.ca!herald.usask.ca!alberta!cpsc.ucalgary.ca!ctycal!ingoldsb From: ingoldsb@ctycal.UUCP (Terry Ingoldsby) Newsgroups: sci.electronics,comp.dsp Subject: Re: A question about the Nyquist theorm Summary: More questions about practical sampling Message-ID: <625@ctycal.UUCP> Date: 26 Feb 91 00:18:04 GMT References: <20408@shlump.nac.dec.com> Organization: The City of Calgary, Ab Lines: 50 Pursuing the discussion of the Nyquist theorem, I have a question about practical sampling applications. If you have a sine wave at frequency f, which you sample at just over 2f samples per second then the Nyquist theorem is satisfied. I know that by performing a Fourier transform it is possible to recover all of the signal, i.e. deduce that the original wave was at frequency f. Note that this is different than just playing connect the dots with the samples. Most of the algorithms I've heard of used with CD players perform a variety of interpolation, oversampling, etc., but these all seem to be elaborate versions of connect the dots. I'm not aware that the digital signal processing customarily done will restore the wave to anything resembling its original. I suspect that there is something I am missing here. Can anyone clarify the situation? E.g. Original: x x x x x x x x x x x x x x x x x x x x x x x x x x x ^ ^ ^ ^ ^ ^ ^ Sample points Connect the dots reproduction (you can draw in the lines, I hate ascii drawings) x x x x x x x The only thing I can think is that the resulting waveform must contain frequencies greater than the Nyquist limit allows, thus permitting them to be filtered out with a brick wall filter (approachable with digital filtering) letting the orignal come through unaltered. Can someone confirm my belief? -- Terry Ingoldsby ingoldsb%ctycal@cpsc.ucalgary.ca Land Information Services or The City of Calgary ...{alberta,ubc-cs,utai}!calgary!ctycal!ingoldsb