In a previous article, kolstad@cae.wisc.edu (Joel Kolstad) writes: >In article <hcbC5un9L.DD0@netcom.com> hcb@netcom.com (H. C. Bowman) writes: >> >>I just bought a new portable CD player for the office, and I notice that >>it proudly proclaims "8 TIMES OVERSAMPLING" on the box.  Now while I think >>I understand what oversampling is (the rate of discrete "samples" >>exceeds the highest frequency component of interest by some factor), >>I don't understand this "8 TIMES" business...  It seems to me that when >>I bought my first CD player (was it REALLY 10 years ago?!), the specs >>said "4 TIMES" ...  Could someone please tell me whether I'm getting >>senile?  If I'm not, then what good does it do for the player to take >>samples at a higher rate?    >The data is only ever read once (barring mistracks and such, of course), >and eventually gets turned into 44.1 KHz, 16 bit, two channel data. >Oversampling takes two discrete data points, and interpolates n-1 points >between them for n times oversampling.  When I asked, people said that the >interpolation was not simply linear interpolation, but significantly more >complicated.  You are quite correct in your understanding.  The filtering is not interpolation, as that would distort the frequency content of the signal you are listening to.  Generally, these players run the samples thru an all-pass filter network.  I have done this for ECG waveforms from a person's heart, and the effect is rather spooky.  It actually reconstructs peaks that weren't there (correctly, too!) and fills in the gaps with the properly computed values, just as if there had been a real sample taken at that point.  I use a CPU to do all the math.  It takes a decent (but not unreasonable) amount of CPU time to do this.  You can keep up with things in realtime if you write efficient code.  In case you care, the filtering method uses an FIR (finite impulse response) filter.  I'd guess that CD makers use the same kind of method.  Anybody out there know the real answer?  I'd say that they use a tapped delay line with resistor/op-amp weighting to accomplish the filtering.  This strikes me as the most cost effective method for volume production runs.   >Anyway, then, the purpose of oversampling is to move the "effective" >sampling rate up to n times 44.1 KHz, in order to use higher frequency >antialiasing filters.  For the same quality filter, higher oversampling >lets you build cheaper filters, whereas for the same price filter, higher >oversamplings lets you build better filters.  So, assuming the quality of >all other components in a CD player remained the same, oversampling should >allow a manufacturer to produce _slightly_ better sound due to anti-alias >filtering.  Actually, I think the only reason they do this is so that they can say that they have a marketting gimic.  I would guess that it is acutally cheaper to filter an oversampled signal than not.  You can use sloppier components and give the filter a roll-off that isn't so sharp.   It's too bad that they charge more for something that (I think) is actually less costly to build.  I seriously doubt that the filters cost the same but are better.  They are built to a price spec, and that spec says "cheap as possible!".    >					---Joel Kolstad    -dave