TS parameters & closed-box design

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It is far better to admit to one's self that "I don't know" and being man/woman enough to be able to admit that in public.
A far bigger failing is to never admit to your limitations, either privately or publicly.

Do not be afraid to ask, whatever the question. Reading things two or three times does not always make it any clearer. I know, I have to do that repeatedly.
 
It is far better to admit to one's self that "I don't know" and being man/woman enough to be able to admit that in public.
A far bigger failing is to never admit to your limitations, either privately or publicly.

Do not be afraid to ask, whatever the question. Reading things two or three times does not always make it any clearer. I know, I have to do that repeatedly.

Andrew, you must have a communion wafers overdose ;)

To extract the system's transfer function from the impedance measurement, see the last pages of LIMP's software documentation on ARTA's website.

The LT is described on Linkwitz' website.

The remaining is about transposing the analogue LT to DSP.
 
Bart,

Point 2 is your specifications. This is what you want to achieve.

With point 1 you get fc and Qc. The function of your system can be derived from these two parameters and is

G(s) = Tc^2 * s^2 / (Tc^2 * s^2 + Tc/Qc * s + 1)

where Tc = 1/(2*pi*fc)

You specify LT so that the system Original Box (G(s)) combined to LT(s) is giving you the roll-off you're after.
So the numerator of LT(s) must be equal to the denominator of G(s).

Then let's call Tcd and Qcd the target that you want to achieve: this is determining the denominator of LT(s).

Finally: LT(s) = (Tcd^2/Tc^2)*(Tc^2 * s^2 + Tc/Qc * s + 1)/(Tcd^2 * s^2 + Tcd/Qcd * s + 1)

Note that when you associate G(s) and LT(s) you obtain indeed your final system F(s):

F(s) = G(s) * LT(s) = Tcd^2 * s^2 / (Tcd^2 * s^2 + Tcd/Qcd * s + 1)

Which is equivalent to your original system function, where Tc has been replaced by Tcd and Qc replaced by Qcd (try it on a piece of paper if you don't believe me ;) ).

At that stage you have the specification of your LT in continuous time domain. You have 2 options:
1. You implement your LT with an op-amp. In this case just refer to Linkwitz' original notes
2. You implement it in dsp. Then you need to extract coefficients for your filter. This is point 4.
 
I hope I would understand by working it out, if you could tell me what the abbreviations stand for. G(s) LT(s) Tcd etc...
If not, then it's useless for me.

-Keep munching wafers, they're good for you AND others!- :D

Bart, it's all explained in my previous post and i'm using near standard notations. You can refer to the excellent article of Small "Direct-Radiator Loudspeaker System Analysis", which is the foundation of low-frequency loudspeaker design. If you don't have a copy, you can buy one from the AES website (or IEEE), they charge about 5 bucks if i remember correctly.
 
A quick update on this ongoing project.

I was lucky enough to find the schematics of my old CD player which originally doesn't have a digital output.
Even luckier: there's a version of the same model that has a digital output :p.
So it turned out to be easy enough to add a SPDIF output to the player and plug it to the processor.

dig.jpg
 
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