Modeling non-linearities with LaPlace Tranfer functions

Apr 18, 2006 7 Replies

Dear All,



I'm working on modeling the behavior of an analog equalizer at high frequency.



The way I look at the system is that it should include 2 main responses : DC gain and AC gain. The DC gain is easily obtained by DC sweeping the input and measuring the resultant DC output voltage. The resultant DC response looks more like a tanh function with a linear operation region and a saturation region at its extremes. As for the AC response, based on small-signal analysis, it's basically a transfer function with a certain number of poles and zeros that I implement using laplace transform function.



To obtain the overall gain, the DC response is multiplied by the AC response, where the latter is first normalized to its DC gain to make sure the DC gain is not included twice.



Results are as expected for the range of inputs that fall within the linear region of the DC tanh function. On the other hand, when the input amplitude exceeds the linear region, the gain values can not be anticipated with hand analysis.



Did anybody encounter a similar problem ? Do you have any recommendations on ways to include such "large-signal" effects into the model.



Your contribution is highly appreciated.



Thanks, Tamer.



Unfortunately there's no simple answer to this question. As soon as a system goes outside the linear region, all bets are off. There are heaps of examples of apparently 'simple' nonlinear systems exhibiting massively complex behaviour that can't be predicted with any formal method. If you want an example, google on Mandelbrot set or Henon maps.

Depending on what you want to do, a couple of techniques that *might* prove useful are piecewise linearisation, which is deriving a linear relationship about a chosen operating point for small deviations, and describing functions which extend frequency domain analysis into the nonlinear regime - slightly.

A practical approach for dealing with nonlinear systems is rigorous system modelling in the time domain, this requires experience, time and patience to get results.

The base of the fourier/laplace transform is linearity. So if youf system is not linear, then you have to look very precise what actually you're doing. A thorough understanding of the involved theory is necessary.

Start with the Schwartz space, where the transform lives in. IMO, the ground to use the laplace transform becomes soft, wobbly and slowly unuseable.

Rene

Ing.Buero R.Tschaggelar - http://www.ibrtses.com & commercial newsgroups - http://www.talkto.net

On approach is to approximate the nonlinear response by the sum of functions (non necessarily linear) for which the LaPlace transform exists.

This is excactly what does not work. A property of the Laplace transform L() :

L(a*A+b*B+c*C)=aL(A)+bL(B)+cL(C) //a,b,c : Constant

Now since the input of the Transform is not linear: { Be f the nonlinear function } f(a*A+b*B) != af(A)+bf(B) //a,b,c : Constant

Then the property above is not true either. Sorry.

Rene

Ing.Buero R.Tschaggelar - http://www.ibrtses.com & commercial newsgroups - http://www.talkto.net

That looks like a differential amplifier. You can develop the 1+tanh function like this:

1 +Vd/2Vt - Vd^3/24Vt^3... which is enough. Now you can substitute the Vd=differential voltage with Vd*sinwt and get Vt is the temperatur voltage (26mV) the linear range is around +/- 2Vt 1 + Vd/2Vt *sinwt - Vd^3/96Vt^3 *(3sinWt - sin3wt) this is the distortion which is rising with the squared amplitude. Vd= peak amplitude
ciao Ban Apricale, Italy

I understand that this solution does include the nonlinearities in some way but I don't see how this will include the Gain as a function of frequency ?

Thanks for your feedback.

Timo, this is adding harmonics depending on the signal level. So you take your signal to be processed and apply the filter functtion and after each filter step calculate the level and then add those harmonics depending on the envelope and do the next filter stage with the addaed harmonics. those

3rd order ones are pretty awful, but you can use 2nd order ones, that rise lineary with amplitude. It is a difficult operation and you have to go from frequency domain to time domain and back.
ciao Ban Apricale, Italy

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