FDNR in active filters (Frequency Dependent Negative Resistor) - diyAudio
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Old 8th April 2003, 09:31 AM   #1
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Default FDNR in active filters

Hi!

Has anyone used FDNR (frequency dependent negative resistances) in an anctive filter? I started long time ago to design such a filter but other things got in the way... still FNDR looks very cool.

Look at page 10

http://search1.analog.com/searchProx...3DFNDR%3Bfs%3D
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Old 8th April 2003, 03:14 PM   #2
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Old 8th April 2003, 04:27 PM   #3
BrianL is offline BrianL  United States
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Check out Burr-Brown (TI) ap note AB-026A
by Rick Downs, circa 1991:
http://www-s.ti.com/sc/psheets/sbaa001/sbaa001.pdf

They are very flexible though my experience
is that they must be carefully laid out and
care given to scaling impedances and picking
the op-amps so that the op-amps don't oscillate.
PC board with ground plane is a VERY good idea.

Also, don't believe any of the B.S. about the
"op-amps not being in the signal path". One
can make the argument is that this is true
in some literal way, but it's like saying that
a shunt resistor on an attenuator isn't in
the signal path.

Still, a very interesting and flexible topology.
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Old 8th April 2003, 04:37 PM   #4
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Lightbulb FDNR

Quote:
Originally posted by BrianL
Check out Burr-Brown (TI) ap note AB-026A
by Rick Downs, circa 1991:
http://www-s.ti.com/sc/psheets/sbaa001/sbaa001.pdf



Also, don't believe any of the B.S. about the
"op-amps not being in the signal path". One
can make the argument is that this is true
in some literal way, but it's like saying that
a shunt resistor on an attenuator isn't in
the signal path.

Hi bel, I concur.
Sony has these FDNR's in there top of line models.
But you can very clearly hear the difference if you replaced the NE5532's by AD712's or AD827's !


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Old 8th April 2003, 05:01 PM   #5
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Maybe the truth lies inbetween.

If you build a notch filter for instance, the OPAMP will definitely have as much influence as if it was IN the signal path (if not even more, due to the high Q circuit).
But there is also the possibility that the influence is reduced (I don't say zero) compared to "ordinary" filters at frequencies where the FDNR circuit is having high impedance.

Regards

Charles

/having used simple non FDNR shunt circuits for VCFs
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Old 8th April 2003, 05:18 PM   #6
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I have also read somewhere that it's neccessary to have matched opamps (duals).
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Old 8th April 2003, 08:53 PM   #7
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Quote:
In regards to your question about FDNR filters, yes, they do have some interesting features which make them attractive, especially in audio applications.

One thing I like about them is that the active elements (op amps) are not directly in the signal path. They are in a shunt arrangement. I have made this statement in print before and have generated some discussion on what in the signal path really means. However the signal, as it appears at the output of the filter, never passes through an op amp. This is probably not a big deal, one way or another, since modern op amps are very good.

I haven't done an exhaustive study on the dynamic range, SNR, and the like on the FDNR. Neither has anyone else that I have been able to discover. At any rate, since the active elements are in a shunt arrangement, it is intuitively obvious, at least to me, that they would have much less effect on dynamic properties than if they were directly in series. Again, this is just gut feel. Maximum Q may be something that should be investigated. Some filter typologies have a practical limit to the achievable Qs. I have never run into this issue with the FDNR, but, there again, I have never designed one with a high Q section. Again, from inspection, since the Q is typically dependant on component matching, the FDNR should perform well.

Component value spread is very low in the FDNR, which is a plus. Section interaction is also low.

One disadvantage is the design is a bit more complicated. You start with a passive LC prototype and transform it. This is fine if you have a catalog of passive filters, such as in the Williams book (see the references after the filter chapter), but another level of complexity if not. Because of the interaction between stages, passive filters are somewhat difficult to design, especially if the order gets high.

Another potential disadvantage is the number of components. More than the Sallen-Key and Multiple feedback, but less than the state variable and Biquad. Since we are just transforming components, the Q and Fo of the filter is not as independent as the state variable and the biquad, but neither are the Sallen -Key and the MFB.

Another question that comes up is the bandwidth requirements of the op amps. Again, I haven't done a study on this but can offer the following as a guide. The 2 op amps in the filter are used as a gain block and an integrator. The integrator will be the most demanding application. Rule of thumb is that you would like the integrator to have 20 dB of loop gain at the center frequency. So a 10 kHz filter would require a 100 kHz or better op amp.
I got this answer from Mr Hank Zumbahlen at Analog Devices
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Old 8th April 2003, 08:55 PM   #8
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Quote:
Originally posted by BrianL
Check out Burr-Brown (TI) ap note AB-026A
by Rick Downs, circa 1991:
http://www-s.ti.com/sc/psheets/sbaa001/sbaa001.pdf
This was a very good document. Thanks!
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Old 9th April 2003, 03:05 AM   #9
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Default analog filter humor

from Maxim's website:<em>
Analog Filter Design Demystified

This article shows the reader how to design analog filters. It starts by covering the fundamentals of filters, it then goes on to introduce the basic types like Butterworth, Chebyshev, and Bessel, and then guides the reader through the design process for lowpass and highpass filters. Includes the derivation of the equations and the circuit implementation.


It's a jungle out there.

A small tribe, in the dense wilderness, is much sought after by head hunters from the surrounding plains. The tribe knows it is threatened, because its numbers—killed off by the accelerating advance of modern technology—are dwindling at an alarming rate. This is the tribe of the Analog Engineers.

<b>The guru of Analog Engineers is the Analog Filter Designer, who sits on the throne of his kingdom and imparts wisdom while reminiscing of better days. You never get to see him even with an appointment, and you call him "Sir."

The countless pages of equations found in most books on filter design can frighten small dogs and children. This article unravels the mystery of filter design, enabling you to design continuous-time analog filters quickly and with a minimum of mathematics. The throne will soon be vacant.

The Theory of Analog Electronics

Analog electronics has two distinct sides: the theory taught by academic institutions (equations of stability, phase-shift calculations, etc.), and the practical side familiar to most engineers (avoid oscillation by tweaking the gain with a capacitor, etc.). Unfortunately, filter design is based firmly on long-established equations and tables of theoretical results. Filter design from theoretical equations can prove arduous. Consequently, this discussion employs a minimum of math—either in translating the theoretical tables into practical component values, or in deriving the response of a general-purpose filter.
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Old 9th April 2003, 08:38 AM   #10
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Quote:
Originally posted by BrianL
Also, don't believe any of the B.S. about the
&quot;op-amps not being in the signal path&quot;. One
can make the argument is that this is true
in some literal way, but it's like saying that
a shunt resistor on an attenuator isn't in
the signal path.

Still, a very interesting and flexible topology.
I think there is some truth in "not in the signal path" becuase at low frequencys (LP-filter) the signal passes simply through. Disadvantage of the filter is the high output impedance.
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