Explain this DSP notation?

So the z 1 just means a one-unit delay Let's take H p (1-p)/(1-pz 1 ) If we follow the convention of "x" for input and "y" for output, the transfer function H = y/x (=output/input) so we get y/x = (1-p)/(1-pz 1 ) or (1-p)x = (1-pz 1 )y (1-p)xn = yn - pyn-1 or: yn = pyn-1 + (1-p)xn In C code this can be implemented y += (1-p)*(x-y) without any additional state beyond using the output "y" as a state variable itself. Or you can go for the more literal approach: y_delayed_1 = y; y = p*y_delayed_1 + (1-p)*x As far as the other equations go, they're all typical equations except for that second equation which looks like maybe it's a way of selecting either H? 1-z 1 OR 1-z 2 (what's N?) The filters are kind of vague and they'll be tougher for you to deal with unless you can find some prepackaged filters.

In general they're of the form H = H0*(1+az 1 bz 2 cz 3 )/(1+rz 1 sz 2 tz 3 ) and all you do is write down H = y/x, cross multiply to get H0 * (1+az 1 bz 2 cz 3 ) * x = (1+rz 1 sz 2 tz 3 ) * y and then isolate "y" by itself, making the output "y" a linear function of various delays of itself and of the input But designing filters (picking the a,b,c,etc. ) is tougher than implementing them, for the most part.

So the z-1 just means a one-unit delay. Let's take Hp = (1-p)/(1-pz-1). If we follow the convention of "x" for input and "y" for output, the transfer function H = y/x (=output/input) so we get y/x = (1-p)/(1-pz-1) or (1-p)x = (1-pz-1)y (1-p)xn = yn - pyn-1 or: yn = pyn-1 + (1-p)xn In C code this can be implemented y += (1-p)*(x-y); without any additional state beyond using the output "y" as a state variable itself.

Or you can go for the more literal approach: y_delayed_1 = y; y = p*y_delayed_1 + (1-p)*x; As far as the other equations go, they're all typical equations except for that second equation which looks like maybe it's a way of selecting either H? = 1-z-1 OR 1-z-2. (what's N?) The filters are kind of vague and they'll be tougher for you to deal with unless you can find some prepackaged filters.

In general they're of the form H = H0*(1+az-1+bz-2+cz-3...)/(1+rz-1+sz-2+tz-3...) and all you do is write down H = y/x, cross multiply to get H0 * (1+az-1+bz-2+cz-3...) * x = (1+rz-1+sz-2+tz-3...) * y and then isolate "y" by itself, making the output "y" a linear function of various delays of itself and of the input. But designing filters (picking the a,b,c,etc.) is tougher than implementing them, for the most part.

Beautiful. Thank you! – Grumdrig Dec 17 '09 at 2:49 BTW to answer your question, N is the period of the string vibration being modeled, in samples.

– Grumdrig Dec 17 '09 at 21:57.

I cant really gove you an answer,but what I can give you is a way to a solution, that is you have to find the anglde that you relate to or peaks your interest. A good paper is one that people get drawn into because it reaches them ln some way.As for me WW11 to me, I think of the holocaust and the effect it had on the survivors, their families and those who stood by and did nothing until it was too late.