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

Steffen Leve Poulsen slagmark at worldonline.dk
Thu Jan 22 22:45:19 CET 2009

Hi Geoff

Yes Pd is wonderfull!

Attached shows how to do your example the naive way in Pd.
It uses [tabsend~] to get from sig~ domain to message domain.
This is not that efficient but good for testing and development.

More efficient is [z~] from zexy or [delread~] and [delwrite~].
Also look at [biquad~] and the raw filters [czero~] and [cpole~]

mvh/ SLP

Geoff skrev:
> 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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