Linear Phase,FIR Filter based EQ

i tend to ignore FIR, so i dont know much about it, but for a start...

- afaik (which usually does not mean much) it will be phase linear by definition - but it wont be very frequency linear. unless you are using a really big buffer.

- to make it 96 dB/A per ocatve, you´d simply put several of them in series, like you also would do with a biquad filter.

here is a very old implementation of a high pass with only 31 samples. it sounds terrible. you might want to try 511 for now, but in max 7 the size is no longer limited at all i think.

-110

For a linear Phase, the impulse response of the filter needs to be symmetrical.
A 96db/oct FIR will be very performance hungry.
Why do you need the filter to be linear phase? In many many applications this is not necessary and actually sounds bad/messes with transients..
(of course it sounds good if a company makes an EQ and say it's 'linear phase' but again, it really depends on the situation and most of the time they are not a good idea.)

some more detailed answers to romans comments:

I've been ignoring FIR

for a good reason.

– afaik (which usually does not mean much) it will be phase linear by definition – but it wont be very frequency linear. unless you are using a really big buffer.

FIR filters are not always phase linear. They can be.
IIR filters are always not phase linear.

The frequency response being 'linear', i guess you meant you can't get a nice linear passband. Yeah, I mean you can, but as you said, you need many coefficients=many delaytaps=a big buffer=poor performance to get a steep roll off.

– to make it 96 dB/A per ocatve, you´d simply put several of them in series, like you also would do with a biquad filter."

yes, you could do that, or you could do it all in one operation, which might be more efficient. Not sure, can't the filterdesign object design FIRs also? Otherwise, it's really easy in matlab...

ok one more:

nd can you explain to me were in the equation is the Q the Freq and the slope? how to make it changeable?
thank you for reading and sorry for my lack of knowledge.

so, Q is typically a feedback related parameter, wich you don't have in a FIR. But anyway, let's take slope/rolloff and frequency..
I really don't know how people do this. But you could easily do the following:
You just draw your desired frequency/amplitude response into a buffer, do the inverse fourier transform and you have the impulse response of the filter.
Or: you use a IIR, contriol the Freq/Q of this filter. Everytime the parameters change, you get the impulse response of that filter, window it (since it is potentially infinite) , make it symnmetrical and use that as your convolution kernel/FIR coeffs..

"needs to be symmetrical."

yeah, now that you remind me. well that at least saves from building the whole signal chain symetrical (as you need to do in analog hardware)

but that doubles the latency, right?

"I really don’t know how people do this."

i believe it makes the most sense to design such a filter parallel, i.e. you mix the FIR filter against the input to get a lower slope.

You might be able to glean a few insights from this, which just went up today:
https://cycling74.com/tutorials/crossover-filter-design-video-tutorial

The important part of the phase here is that the two signals produced have exactly the same phase as each-other. The fact that the phase is not linear only becomes a potential problem if you want to mix one of the filter output signals with the original input signal.

In this case it is quite common to linearize one of the signals using an allpass filter.

@Timothy, thats really a great video, once again!
And yes thats also my understanding, that linear phase is only important if you plan to mix with the original signal. I think in all other cases it only wastes cpu cycles. Especially if you build such a steep FIR.
Also I think its not a good idea to build the filter in parallel, dry/wet mixing with the original, as roman suggested, in order to get a variable filter steepness. Its just wasting performance if you imagine the case of a very slight filtering but having a 96db/oct filter only slightly mixed in.
"the filter kernel being symmetrical doubles the latency"
I don't think thats true. Or rather, doubles it in comparison to what? For a symmetrical kernel you get a latency of kernellength/2.
Timothy, do you mean it is possible to make a linear phase filter with an IIR in combination with a compensating allpass? I've never heard of this, can you post some link/paper/example? thank you!

@WOYTEG Yes, it is possible to combine an allpass with an IIR to make the phase more linear.

One cannot (obviously) remove the inherent delay, however, so it is often more effective (in the case stated here) to add the allpass to the input signal to impart to it the same phase response as the crossover.

Depending on how much you want to go crazy with this, here is an old paper on allpass design: http://www.ece.rice.edu/dsp/publicationsold/pub/designappl.pdf

It is a little challenging to sort through all of the noise when doing a web search on this subject as there is a lot of disagreement and actually hearing the results is highly dependent on the source sound material used for the testing.