Practical FIR

Part 3

As ever so often in engineering, if some new technology is introduced we go into  'nec plus ultra' modus. Suddenly you will find a huge range of applications where all and unique is about that new tech. Same for FIR filtering. 

The first appearance of FIR in (diy) audio was as a means to do room EQ-ing. The idea was to measure a systems response in a room, do quite some averaging and apply an overall EQ that would correct all and everything. First there is of course no such thing as room EQ: the only way to change the acoustics of a room is by changing it's dimensions and/or changing the absorbing/reflecting surfaces. But the solution is better then using only conventional (min. phase) EQ as we have been doing with our 31b. graphics.

Some of the room effects are min. phase and thus one could use a graphic to try to address the problems. The definition of min. phase behaviour dictates that your ideal EQ point will have both the necessary magnitude and phase compensation. 

So for these min. phase phenomena the EQ will NOT introduce 'PHASE ISSUES' (crinch).

However. Loads of stuff you would like to EQ in acoustics is NOT min. phase. Remember playing a festival and during nightfall HF content of your PA changes drastically? And changing your graph. EQ doesn't help in the way you wanted? Well that's a non min.phase phenomena and this would call for a lin. phase EQ, which as we know now is only possible with FIR.

 

Another example would be (conventional) EQ-ing across a line array. Hopefully everybody does understand you can't apply different EQ settings to different elements in your line array, right? The different phase behaviour of the different elements will in that case wreak havoc on the overall dispersion of the array! The only way you can get away with this is by using lin.phase EQ. Something that French manufacturer of brown boxes does understand.

So when and where to apply lin.phase FIR filters as a means of filtering without affecting phase is something that needs thought!

It is not the miracle solution for everything, it is just another tool and it does have pro's and con's like any other tool.

So yes, as has been explained on million places: FIR filters can be computational heavy, but why should that be mentioned over and over as a problem? 

A more interesting difference, if you really would like to set IIR versus FIR, is the way the precision of the calculations work out sonically. Fixed point vs floating point. Internal bit depth. The way rounding errors work in your algorithm. Audibility of the pre-rings (or not?) That sort of thing. In the MCU/DSP part of this blog I will write a bit about this.

If all this is cleared, one big topic still is to be debated:

IS PHASE AUDIBLE?

(or more precise does applying an allpass filter change the way (stereo) audio is percieved?)

One will understand that I (and my peers) do have an opinion contrary to that of the big names in our industry. This is the internet: I can't demonstrate, but I encourage you to find out for yourself!