drewfx1 recently pointed out that FIR and IIR filters both have benefits, and that each can be applied to suit different priorities.
 
I went to study up and found this video helpful:
 

Thanx Mike! Will watch this after all my other "stuff" is done!

mike_mccue
 
I went to study up and found this video helpful:
 

Mike, if you truly understood this video tutorial, then I'm duly impressed.  When I was studying engineering in college, I frequently encountered the kind of math involved in DSP calculations (although I never directly studied DSP itself).  However, that ship sailed a long time ago.  This stuff isn't for the faint of mind.
 

That's a pretty cool video! FIR vs IIR filters have been the subject of debate (intense debate) within the live sound community for the last couple of years. Almost every DSP device (think BSS, Biamp, Symetrix, MediaMatrix, etc) offers both, and the latency can now  be reduced to near manageable levels. Among other things, this makes it possible to create some really hairy - I mean complex) cross-over networks... cross-overs being one of the weak points in most multi-way sound reinforcement systems.

There are a couple other resources you might enjoy - I have to find out if they are freely sharable, but if they are I will post them here.

Not sure what hardware/software you have at your disposal - but if you get the chance try creating the same filter in both topologies and see if you can hear a difference!

dmbaer

mike_mccue
 
I went to study up and found this video helpful:
 

Mike, if you truly understood this video tutorial, then I'm duly impressed.  When I was studying engineering in college, I frequently encountered the kind of math involved in DSP calculations (although I never directly studied DSP itself).  However, that ship sailed a long time ago.  This stuff isn't for the faint of mind.
 

Hi David,
 I don't think "truly understand" applies for me, but I did find that I was able to follow the explanation as the overview it seemed it was meant to be, and I felt like I understood more at the end than I did at the beginning.
 
 I was especially interested in the idea that FIR refers to a data table, that so called Linear Phase EQs are FIR, and that the reference to a large data table can cause lots of loading. It made me think of the infamous LP64 EQ and all its glitches.
 
 I'd like to follow some examples of IIR where the variables are replaced with actual numbers so I can get a better understanding of what goes on.
 
 best regards,
mike

mike_mccue 
 I'd like to follow some examples of IIR where the variables are replaced with actual numbers so I can get a better understanding of what goes on.
 

What goes on is if you take any set of coefficients and a feed a single sample with a value of "1" surrounded by silence and do the math you get the impulse response.

drewfx1

mike_mccue 
 I'd like to follow some examples of IIR where the variables are replaced with actual numbers so I can get a better understanding of what goes on.
 

What goes on is if you take any set of coefficients and a feed a single sample with a value of "1" surrounded by silence and do the math you get the impulse response.

+1
(see watt I did there?)

drewfx1

mike_mccue 
 I'd like to follow some examples of IIR where the variables are replaced with actual numbers so I can get a better understanding of what goes on.
 

What goes on is if you take any set of coefficients and a feed a single sample with a value of "1" surrounded by silence and do the math you get the impulse response.

It seems like seeing first hand how a series of samples go in one side and come out the other with a 1 octave wide boost at 5kHz would be something that would help with furthering an understanding.
 
best regards,
mike

mike_mccue
It seems like seeing first hand how a series of samples go in one side and come out the other with a 1 octave wide boost at 5kHz would be something that would help with furthering an understanding.
 

That's exactly what the impulse response shows you.
 
A single sample impulse input to the filter contains all frequencies with flat response and the impulse response output by the filter contains the filtered response. You have a series of filter coefficients that produce a given impulse response. The impulse response completely defines what the filter does in the time domain - it equates to the frequency and phase response of the filter in the frequency domain. 
 
The filter coefficients define either an FIR or an IIR filter and you just have a series of single sample delays, multiply the delayed samples by each associated coefficient and add each result back to your signal (with an IIR some of the coefficients feed back as well). IOW, the basic structure of the filter is either FIR or IIR, but is otherwise the same regardless of what the filter is actually doing - what changes is the coefficients.
 
The tricky part is getting the filter coefficients that produce the desired results. 

Yes, I'd like to see a specific example with a impulse response that is described as something I can relate to and with an input that is described as something I can relate to.
 
The video showed some filters, but I have no idea what they were.
 
You mentioned an input of "1" surrounded by silence, and I can not readily imagine what that an input of 1 sounds like.
 
I think it might be interesting to see a specific example where a input, perhaps a G chord "pad" voiced on a Hammond B3 went in and came out the other side.
 
Perhaps what I am missing is that I don't actually need to recognize the input to see the "filter", but at the very least it would seem like the knowing what the filter is supposed to be doing, in some specific example, might be a good next step.
 
I have seen transfer functions for analog EQ design but the little bit I know about them hasn't prepared me to recognize how/what/if Impulse Response techniques are similar or different.
 
best regards,
mike