TOPIC:

I have a MiniDSP SHD Power, and I'm considering buying a MiniDSP Flex Eight.  
I'd like to use my MiniDSP's to apply Linkwitz Transforms (and a high pass Butterworth filter) for DIY speakers that I've simulated using WinISD.

So I'm reading the old MiniDSP page regarding Linkwitz transforms ( www.minidsp.com/applications/advanced-tools/linkwitz-transform ) and it says I need "the F and Q numbers"...
But it's not clear to me whether those are numbers correlating to my driver specs (driver specs usually have Fs, Qms, Qts, Qes) or to my calculated Linkwitz transform specs.

My driver specs are : Fs 29.5hz, Qms 4.70, Qts 0.29, Qes 0.39  
My WinISD results are : f0=80.0Hz, fp=10.0Hz, Q0=0.6, Qp=3.0

What are the F and Q numbers used in that page?

It appears you are using a ported enclosure, the Linkwitz Transform is only meant to be used on sealed enclosures.

That being said the values for f and Q used in the transform should NOT be of the driver itself but rather the driver in an enclosure. Ideally this would be directly measured with something like the Dayton Audio DATS but you could attempt to model it from manufacturer specs and something like WinISD.

Michael

Much of the implementation of the original solution was centered around analog circuit capabilities of the time.  (This was way back in the 1970's remember.) 
www.linkwitzlab.com/filters.htm#9

If you really want to dive into the theory on this you can, but nowadays DSP allows arbitrary equalization so you can create whatever you want, easily.
Any of the miniDSP platforms can create simple second-order correction of this type.
Put your woofer in a sealed box, put your microphone an inch from the cone, and see what the response is.  Correct and extend as you see fit.

Dave.

Hi Michael, thanks for the reply.  

First off, this is all for an experimental build just for me to learn and get my hands dirty, doing some things that go against the grain. 
My experiment is a small (bookcase type) speaker with a sealed subwoofer driver (TB 1139SIF) to do everything below say 250Hz, a 4" BMR driver (Cotswold audio) to perform low mids to highs, and an isodynamic planar tweeter (HiVi RT1C) for highs/ultra-highs (>7Khz). They're not meant to go loud - they'll end up in the bedroom probably.  
I already have these drivers laying around from previous projects, so this is what I'll be working with

The BMR and Tweeter are quite sensitive (90dB+) so I'll pad them down a bit, and the 1139SIF is fairly unsensitive (83dB) so I'll throw some class D power at it to match the mid and tweeter, Luckily the 1139sif has an xMax of 11-13mm so it can be pushed a bit..
 
I'd like to use the Linkwitz Transform and the raw power of a modern class D amp (250-450W TPA3255) to get the 1139SIF to go down as low as possible in a relatively small sealed cabinet.  
 

Hi Dave, thanks for your reply.  

I'm designing an active 3-way speaker where every driver has its own (Class D) amplifier. 
Will pulling up a control point on the Dirac Live curve around 20Hz until the curve looks cool on screen really have the same effect as using a calculated Linkwitz transform to boost the low-end..?

I honestly am not informed enough about the math behind both methods to know whether the result will end up the same - SPL wise, phase wise, and with respect to other things I probably don't even know about :-/