Question about filters

Can someone please explain what the difference between a 1, 2, 4, 6, and 8 pole filter is??
I don't understand it's function
thanx!!

Not sure about all the tech terms, but i'll do my best.


From what I understand, the poles are otherwise looked at as "areas of alteration."


For example. a 1 pole filter will only allow you to filter one area of frequency at a time. Notice the change the envelope when you mess around with the knobs/controls.

So if you wanted to say, filter out some low end, you'd have that.


Now let's say you wanted to be able to filter more than one area...lets say you wanted to fool with the mids too. Rather than run two separate filters, you'd be--in essence--adding another "pole". So now the 2-pole would let you edit two areas of the frequency spectrum, so you'd have two envelopes.

And so on.

The filter's pole is the depth of slope in the cutoff region. A one pole filter means it has a 6dB per octave cutoff (in fact each pole represents 6dB, so 8 poles is 48dB per octave in signal reduction). So if you use a lowpass 1 pole filter, with a cutoff at lets say 440Hz, that would reduce the signal by 6dB at 880.

Think of it this way: an ideal filter would reject all frequencies in its stopband (in the example above the stopband would be all frequencies above 440, and the passband would be all frequencies below). This is not possible, in the real world there is a slope. That means that some frequencies in the passband are reduced, and some frequencies in the stopband are passed. The higher the filter's pole, the closer it is to an ideal filter. Its not necessarily an advantage to be closer to the ideal, but an 8 pole filter can have its uses...

This picture illustrated the idea. The blue line would represent the steepest slope, hence the highest pole.



The Moog filter, the most famous of all time was 4 poles. Most other analog synth designers (Roland with their Juno's, Akai's AX, Oberheim, Sequential Prophet 5 and others) used a 2 pole filter.

Great, thanks for the info!

Don't forget to mention the filter's distortion and frequency dependent phase shifting. If you see it simply as a coil (and remember some of your science lessons) it needs some time to let current pass. A capacitor gives you current nearly instantly. Taking a real world signal causes different parts of the signal to be treated different. For example a low pass filter (coil in series) delays parts of the filtered signal more than others. And at it's resonance frequency it acts as a sort of resonant circuit.

Depending on an analog filter's structure (not only contributing to it's pole count) it creates an own sound. So you don't influence overtones in a linear way, e.g. twice the frequency with half strength. A Moog filter typically has 4 poles (24dB) and consists of 4 6dB filters with a resonance path (output of F#4 fed back to input of F#1). It's phase shift is nearly 1/8th of a wave cycle per 6dB block summing up to half a wave...180°.

And while you turn the knobs the analog filter heats up and it's parts change their parameters. That's why you can always distinguish a (digital) filter emulation from a real (analog) filter. Maybe that's why the MPC comes with special filters in addition to normal LP, HP, BP and notch filters - their resonance and sound shaping capability is poor (when you're used to a Sherman Filterbank or at least Akai's MFC 42).

Something "short" about Moog filters: http://www.xs4all.nl/~rhordijk/G2Pages/ResFilters.htm
A few years ago I got one in original packaging, new! http://www.akaipro.com/mfc42

P.S.: Turning MFC42's resonance to 100% and switching on some of these fancy buttons creates great sounds, even without feeding any signal into this pretty little thing.