Help: How to design the audio pre-emphas(J.17) filter.

hello,

I assume that you would always have a 15kHz low-pass in series. If that is
so, and if you are familiar with MATLAB/OCTAVE and PSPICE/SwCAD, can't you
do it like this:

A1) use "fir1"-function in MATLAB/OCTAVE to create the low-pass (get the
coefficients)

B1) create the given circuit for the pre-emphasis in PSPICE/SwCAD
B2) store the frequency response up to nyquist
B3) use the stored frequency curve with the "fir2"-function of
MATLAB/OCTAVE and get coefficients

C1) convolute the two sets of coefficients. 

=> Resulting coefficients are the ones you are looking for.

I'd apreciate it if you'd correct me, if I'm wrong.

Marcel Roth

marcel.roth wrote:
> hello,
> 
> I assume that you would always have a 15kHz low-pass in series. If that is
> so, and if you are familiar with MATLAB/OCTAVE and PSPICE/SwCAD, can't you
> do it like this:
> 
> A1) use "fir1"-function in MATLAB/OCTAVE to create the low-pass (get the
> coefficients)
> 
> B1) create the given circuit for the pre-emphasis in PSPICE/SwCAD
> B2) store the frequency response up to nyquist
> B3) use the stored frequency curve with the "fir2"-function of
> MATLAB/OCTAVE and get coefficients
> 
> C1) convolute the two sets of coefficients. 
> 
> => Resulting coefficients are the ones you are looking for.
> 
> I'd apreciate it if you'd correct me, if I'm wrong.

You need to know a mathematical expression for the frequency response 
you want in order to create the circuit for SPICE. You can design the 
digital filter directly with that information. There are several design 
procedures for digital filters, many of them in books accessible to 
people who can write what you did. There's a list of such books at 
http://www.dspguru.com

One more thing: the verb you want is "convolve", not "convolute". Just 
the same as "revolve" going with "revolution". English can be quite odd.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
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