[PD] how to create a lowpass filter from first principles?

[Geoff]

Hi
I am new to Pure Data,
I downloaded and have played around with it for the last couple of  
days and it is rather wonderful.

I can quite happily create a basic synth inside it without too much  
strife.
:)

However
The reason I wanted to use Pure Data was to implement some of the  
maths that I am learning through a couple of DSP books that I am  
studying.


The DSP book I have read gives a simple lowpass filter function as

g(n) = (f(n-1) + f(n) + f(n+1))/3

i.e the average value of three consecutive samples. I understand how  
this is in effect a lowpass filter.

How do I implement that in PD?

I can easily connect a Oscillator (phasor) object to a lowpass filter  
object to the DAC and hear it working,
However I want to be able to create my own lowpass filter i.e. try to  
really understand how things are put together.

Therefore if in the function above I need 3 consecutive samples, I  
would have thought that I need some kind of buffer to hold and call  
those samples from,
so that I can then call three samples average them and then output  
that, however that would have to move at the same rate/freq as the  
audio engine is set up i.e. 44100Hz ..... errrr.

So I am thinking about how to solve the problem I just seem to be  
missing how to start.

Any guidance appreciated.
geoff
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