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10-11-2009 05:59 PM

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Hi,

I need to design an amplifier to conteract a low pass filter, so I need simply to build a "filter" where i can define a transfer function (I will simply invert numerator an denominator).

For simulation I used TF Construction an TF interconnection to simulate the filter and the amplifier however these VIs dont process waveform and signal variables.

So i'd like to know if there is a way to apply a waveform to tranfert function or to buid a filter with aspecific tranfer function?

thanks

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##### Re : Amplifier Design

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10-14-2009 04:11 AM - edited 10-14-2009 04:16 AM

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Hello.

To be able to apply your own transfer function, you can use special VIs available in the NI LabVIEW Control and Simulation Module.

Another solution would be to use a Mathscript node.

Regards

Message Edité par mehdi.afif le 10-14-2009 04:16 AM

10-17-2009 02:00 PM

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Hi,

I did the below processing, it seams to be correct. i'd like to verify if my way of thinkung is correct.

I need to apply this transfer function to sound acquiredwith microphone and resend it speaker:

(2.56e-14*s^4+2.56e-10*s^3+9.6e-7*s^2+0.0016*s+1)/(6.561e-17*s^4+5.2488e-12*s^3+1.57464e-7*s^2+0.00209952*s+10.4976)

It corresponds to:

(-0.1919999949e-5*pi*f+(0.3199999488e-2*I)*pi*f+1)/(-3.149279999*10^(-7)*Pi*f+(0.4199039990e-2*I)*pi*f+10.4976)

in the frequency domain.

First, I indexe the sound stream with for loop. sencond, I computes FFT.Next, I indexe the FFT. In Math Sript I calculate the response of the TF. the argument of the function is the index of the FFT array (this is the main point that embarrasses me). I multiply the result by the FFT elements, and finally I reconstruct the sound stream.

Is this the right way to do that?

ie is the result of this processing the output of the obove_cited transfer function?

thank you

02-26-2012 10:07 AM

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Hi

what do you think about this solution to filter a waveform using a TF ??

Nigeltorque