It depends ( it should start to sound familiar...).
If you want to use an analog xover (and you use scenario proposed earlier) then nothing is needed ( maybe a dac if you don't believe in the ones used in deq2496).
If you want to keep it fully digital until dacs driving amplifiers: you'll need a loudspeaker management system ( dcx2496, BSS, Dbx,Xilica,...).
This unit usually have dacs, perform xover duty AND eq... which might make the deq2496 redundant / not needed anymore.
From there a computer with multiple output soundcard will be cheaper and more powerful overall.
Your choice.
But given the price of a deq2496 i would not bother: buy one second hand, try what it does and from there you'll define what you need or not.
If you want to use an analog xover (and you use scenario proposed earlier) then nothing is needed ( maybe a dac if you don't believe in the ones used in deq2496).
If you want to keep it fully digital until dacs driving amplifiers: you'll need a loudspeaker management system ( dcx2496, BSS, Dbx,Xilica,...).
This unit usually have dacs, perform xover duty AND eq... which might make the deq2496 redundant / not needed anymore.
From there a computer with multiple output soundcard will be cheaper and more powerful overall.
Your choice.
But given the price of a deq2496 i would not bother: buy one second hand, try what it does and from there you'll define what you need or not.
Maybe download a DAW and start playing around with the equalizer on a song/source material of your choice on headphones? Like a trial version of reaper and insert reaeq on a channel with a song you like/know. Then play around with different curves/boosts/cuts/bandwidth etc. It will give similar hands on experience as how to eq a sound system/speaker.
I make a summary, with digital EQ deq2496 and op-amp line filter I get EQ, frequency crossover between woofer and tweeter and perhaps phase rotation correction. what else can i get compared to other systems and what i can't get .
after reading the words that come after someone will laugh or others will tear their hair out but we are in 23 and what did not exist yesterday maybe exists now.
it would be very handy to have something that automatically does EQ based on the frequency SPL measurement made with REW and microphone without having to do the settings yourself. a kind of artificial intelligence. I am joking. 😀
after reading the words that come after someone will laugh or others will tear their hair out but we are in 23 and what did not exist yesterday maybe exists now.
it would be very handy to have something that automatically does EQ based on the frequency SPL measurement made with REW and microphone without having to do the settings yourself. a kind of artificial intelligence. I am joking. 😀
You won't get 'phase rotation correction' with deq2496. This kind of treatments are possible using two kind of thing: either an analog allpass ( or digital IIR filters) or by using FIR but it will be limited to dedicated hardware loudspeaker management system ( and not a dcx, it'll not be cheap) or a computer with dedicated software(s)+ multiple out soundcard.
I won't laugh at you at all, but i have concern with 'automatic routine'.
You'll need some mic+rew to do things 'seriously' and once you'll have to interpret measurements, you'll quickly understand concerns relative to robot taking descission in your place...
That said it can be partially automated for some parts of treatments and can give excellent results... but it is outside the range of question you have at the moment imho. 😉
I won't laugh at you at all, but i have concern with 'automatic routine'.
You'll need some mic+rew to do things 'seriously' and once you'll have to interpret measurements, you'll quickly understand concerns relative to robot taking descission in your place...
That said it can be partially automated for some parts of treatments and can give excellent results... but it is outside the range of question you have at the moment imho. 😉
something like this for phase delay ?
https://www.linkwitzlab.com/filters.htm
then the chain of opamps in cascade become 3 ?
https://www.linkwitzlab.com/filters.htm
then the chain of opamps in cascade become 3 ?
Allpass cell yes like the chapter '4' in the link to Linkwitz's site.
3 or more in cascade, it depends...
But be aware the delay that can be achieved are short ( eg: it is a way to compensate for difference in depth between both drivers of a coaxial unit. Tannoy used this on System 800/600 active monitors- 2 cells in serie on the woofer, which is 'in advance' wrt the tweeter emiting point).
Don't expect you could 'linearise' a 4pole Linkwitz Riley or other steep xover with analog: IIR just can't.
For this kind of things only FIR can do as they can separate and treat differently frequency and time domain, iow back to computer or professional loudspeaker management system.
Anyway it drift from your initial question and without being more accurate it's only speculation.
Maybe you could start a thread describing exactly what you want to achieve to have other pov than mine: i gave up on analog for some times now ( with the exception of the System800 passive i own on which i cloned the active circuit to implement the filtering they used without 'wasting' output on my loudspeaker management system (dsp)).
So i'm biased toward digital... and given we have some 'gifted' members about analog...
3 or more in cascade, it depends...
But be aware the delay that can be achieved are short ( eg: it is a way to compensate for difference in depth between both drivers of a coaxial unit. Tannoy used this on System 800/600 active monitors- 2 cells in serie on the woofer, which is 'in advance' wrt the tweeter emiting point).
Don't expect you could 'linearise' a 4pole Linkwitz Riley or other steep xover with analog: IIR just can't.
For this kind of things only FIR can do as they can separate and treat differently frequency and time domain, iow back to computer or professional loudspeaker management system.
Anyway it drift from your initial question and without being more accurate it's only speculation.
Maybe you could start a thread describing exactly what you want to achieve to have other pov than mine: i gave up on analog for some times now ( with the exception of the System800 passive i own on which i cloned the active circuit to implement the filtering they used without 'wasting' output on my loudspeaker management system (dsp)).
So i'm biased toward digital... and given we have some 'gifted' members about analog...
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