Prototype filter

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Prototype filters are electronic filter designs that are used as a template to produce a modified filter design for a particular application. They are an example of a nondimensionalised design from which the desired filter can be scaled or transformed. They are most often seen in regard to electronic filters and especially linear analogue passive filters. However, in principle, the method can be applied to any kind of linear filter or signal processing, including mechanical, acoustic and optical filters.

Contents

Filters are required to operate at many different frequencies, impedances and bandwidths. The utility of a prototype filter comes from the property that all these other filters can be derived from it by applying a scaling factor to the components of the prototype. The filter design need thus only be carried out once in full, with other filters being obtained by simply applying a scaling factor.

Especially useful is the ability to transform from one bandform to another. In this case, the transform is more than a simple scale factor. Bandform here is meant to indicate the category of passband that the filter possesses. The usual bandforms are lowpass, highpass, bandpass and bandstop, but others are possible. In particular, it is possible for a filter to have multiple passbands. In fact, in some treatments, the bandstop filter is considered to be a type of multiple passband filter having two passbands. Most commonly, the prototype filter is expressed as a lowpass filter, but other techniques are possible.

A low pass prototype constant k P (pi) filter Constant k prototype.svg
A low pass prototype constant k Π (pi) filter
Parts of this article or section rely on the reader's knowledge of the complex impedance representation of capacitors and inductors and on knowledge of the frequency domain representation of signals.

Low-pass prototype

The prototype is most often a low-pass filter with a 3 dB corner frequency of angular frequency ωc′ = 1  rad/s. Occasionally, frequency f = 1  Hz is used instead of ωc′ = 1. Likewise, the nominal or characteristic impedance of the filter is set to R = 1 Ω.

In principle, any non-zero frequency point on the filter response could be used as a reference for the prototype design. For example, for filters with ripple in the passband, the corner frequency is usually defined as the highest frequency at maximum ripple rather than 3 dB. Another case is in image parameter filters (an older design method than the more modern network synthesis filters) which use the cut-off frequency rather than the 3 dB point since cut-off is a well-defined point in this type of filter.

The prototype filter can only be used to produce other filters of the same class [n 1] and order. [n 2] For instance, a fifth-order Bessel filter prototype can be converted into any other fifth-order Bessel filter, but it cannot be transformed into a third-order Bessel filter or a fifth-order Chebyshev filter.

A passive lumped low-pass prototype filter of fifth order and the T-topology might have the reactance:

+1jΩ -0.64jΩ +2jΩ -0.64jΩ +1jΩ  (exemplary)

To convert them to 50 Ohm multiply the given values by 50. To get the part value convert at the desired cut-off frequency (corner frequency). Example: The resistance shall be 75 Ohm and the corner frequency shall be 2 MHz.

+75jΩ -48jΩ +150jΩ -48jΩ +75jΩ 6μH  1.66nF  12μH  1.66nF  6μH

Filter types with adjustable ripple can not be easily tabulated as such as they depend on more than just the impedance and frequency.

Frequency scaling

The prototype filter is scaled to the frequency required with the following transformation:

where ωc′ is the value of the frequency parameter (e.g. cut-off frequency) for the prototype and ωc is the desired value. So if ωc = 1 then the transfer function of the filter is transformed as:

It can readily be seen that to achieve this, the non-resistive components of the filter must be transformed by:

  and,   

Impedance scaling

Impedance scaling is invariably a scaling to a fixed resistance. This is because the terminations of the filter, at least nominally, are taken to be a fixed resistance. To carry out this scaling to a nominal impedance R, each impedance element of the filter is transformed by:

It may be more convenient on some elements to scale the admittance instead:

The prototype filter above, transformed to a 600 O, 16 kHz lowpass filter Pi filter 600 ohm 16kHz.svg
The prototype filter above, transformed to a 600 Ω, 16 kHz lowpass filter

It can readily be seen that to achieve this, the non-resistive components of the filter must be scaled as:

   and,    

Impedance scaling by itself has no effect on the transfer function of the filter (providing that the terminating impedances have the same scaling applied to them). However, it is usual to combine the frequency and impedance scaling into a single step: [1]

  and,   

Bandform transformation

In general, the bandform of a filter is transformed by replacing where it occurs in the transfer function with a function of . This in turn leads to the transformation of the impedance components of the filter into some other component(s). The frequency scaling above is a trivial case of bandform transformation corresponding to a lowpass to lowpass transformation.

Lowpass to highpass

The frequency transformation required in this case is: [2]

where ωc is the point on the highpass filter corresponding to ωc′ on the prototype. The transfer function then transforms as:

Inductors are transformed into capacitors according to,

and capacitors are transformed into inductors,

the primed quantities being the component value in the prototype.

Lowpass to bandpass

In this case, the required frequency transformation is: [3]

where Q is the Q-factor and is equal to the inverse of the fractional bandwidth: [4]

If ω1 and ω2 are the lower and upper frequency points (respectively) of the bandpass response corresponding to ωc′ of the prototype, then,

   and    

Δω is the absolute bandwidth, and ω0 is the resonant frequency of the resonators in the filter. Note that frequency scaling the prototype prior to lowpass to bandpass transformation does not affect the resonant frequency, but instead affects the final bandwidth of the filter.

The transfer function of the filter is transformed according to:

The prototype filter above, transformed to a 50 O, 6 MHz bandpass filter with 100 kHz bandwidth Pi filter 50 ohm 6MHz 100kHz.svg
The prototype filter above, transformed to a 50 Ω, 6 MHz bandpass filter with 100 kHz bandwidth

Inductors are transformed into series resonators,

and capacitors are transformed into parallel resonators,

Lowpass to bandstop

The required frequency transformation for lowpass to bandstop is: [5]

Inductors are transformed into parallel resonators,

and capacitors are transformed into series resonators,

Lowpass to multi-band

Filters with multiple passbands may be obtained by applying the general transformation:

The number of resonators in the expression corresponds to the number of passbands required. Lowpass and highpass filters can be viewed as special cases of the resonator expression with one or the other of the terms becoming zero as appropriate. Bandstop filters can be regarded as a combination of a lowpass and a highpass filter. Multiple bandstop filters can always be expressed in terms of a multiple bandpass filter. In this way it, can be seen that this transformation represents the general case for any bandform, and all the other transformations are to be viewed as special cases of it.

The same response can equivalently be obtained, sometimes with a more convenient component topology, by transforming to multiple stopbands instead of multiple passbands. The required transformation in those cases is:

Alternative prototype

In his treatment of image filters, Zobel provided an alternative basis for constructing a prototype which is not based in the frequency domain. [6] The Zobel prototypes do not, therefore, correspond to any particular bandform, but they can be transformed into any of them. Not giving special significance to any one bandform makes the method more mathematically pleasing; however, it is not in common use.

The Zobel prototype considers filter sections, rather than components. That is, the transformation is carried out on a two-port network rather than a two-terminal inductor or capacitor. The transfer function is expressed in terms of the product of the series impedance, Z, and the shunt admittance Y of a filter half-section. See the article Image impedance for a description of half-sections. This quantity is nondimensional, adding to the prototype's generality. Generally, ZY is a complex quantity,

and as U and V are both, in general, functions of ω we should properly write,

With image filters, it is possible to obtain filters of different classes from the constant k filter prototype by means of a different kind of transformation (see composite image filter), constant k being those filters for which Z/Y is a constant. For this reason, filters of all classes are given in terms of U(ω) for a constant k, which is notated as,

In the case of dissipationless networks, i.e. no resistors, the quantity V(ω) is zero and only U(ω) need be considered. Uk(ω) ranges from 0 at the centre of the passband to −1 at the cut-off frequency and then continues to increase negatively into the stopband regardless of the bandform of the filter being designed. To obtain the required bandform, the following transforms are used:

For a lowpass constant k prototype that is scaled:

the independent variable of the response plot is,

The bandform transformations from this prototype are,

for lowpass,

for highpass,

and for bandpass,

See also

Filter bandforms: see, low-pass, high-pass, band-pass, band-stop. Bandform template.svg
Filter bandforms: see, low-pass, high-pass, band-pass, band-stop.

Footnotes

  1. The class of a filter is the mathematical class of the polynomials in the rational function that describe its transfer function. Image parameter filters are not rational and hence do not have a polynomial class. Such filters are classified by type (k-type, m-type etc). Type serves as the class name for image filters and is based on the filter circuit topology.
  2. The order of a filter is the order of the filter's rational function. A rational function is a ratio of two polynomials and the order of the function is the order of the highest order polynomial. Any filter constructed from a finite number of discrete elements will be described by a rational function and in general, the order will be equal to the number of reactive elements that are used.

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<span class="mw-page-title-main">Cutoff frequency</span> Frequency response boundary

In physics and electrical engineering, a cutoff frequency, corner frequency, or break frequency is a boundary in a system's frequency response at which energy flowing through the system begins to be reduced rather than passing through.

<span class="mw-page-title-main">Resonance</span> Tendency to oscillate at certain frequencies

Resonance is the phenomenon, pertaining to oscillatory dynamical systems, wherein amplitude rises are caused by an external force with time-varying amplitude with the same frequency of variation as the natural frequency of the system. The amplitude rises that occur are a result of the fact that applied external forces at the natural frequency entail a net increase in mechanical energy of the system.

A low-pass filter is a filter that passes signals with a frequency lower than a selected cutoff frequency and attenuates signals with frequencies higher than the cutoff frequency. The exact frequency response of the filter depends on the filter design. The filter is sometimes called a high-cut filter, or treble-cut filter in audio applications. A low-pass filter is the complement of a high-pass filter.

Chebyshev filters are analog or digital filters that have a steeper roll-off than Butterworth filters, and have either passband ripple or stopband ripple. Chebyshev filters have the property that they minimize the error between the idealized and the actual filter characteristic over the operating frequency range of the filter, but they achieve this with ripples in the passband. This type of filter is named after Pafnuty Chebyshev because its mathematical characteristics are derived from Chebyshev polynomials. Type I Chebyshev filters are usually referred to as "Chebyshev filters", while type II filters are usually called "inverse Chebyshev filters". Because of the passband ripple inherent in Chebyshev filters, filters with a smoother response in the passband but a more irregular response in the stopband are preferred for certain applications.

The Sallen–Key topology is an electronic filter topology used to implement second-order active filters that is particularly valued for its simplicity. It is a degenerate form of a voltage-controlled voltage-source (VCVS) filter topology. It was introduced by R. P. Sallen and E. L. Key of MIT Lincoln Laboratory in 1955.

<span class="mw-page-title-main">Butterworth filter</span> Type of signal processing filter

The Butterworth filter is a type of signal processing filter designed to have a frequency response that is as flat as possible in the passband. It is also referred to as a maximally flat magnitude filter. It was first described in 1930 by the British engineer and physicist Stephen Butterworth in his paper entitled "On the Theory of Filter Amplifiers".

<span class="mw-page-title-main">LC circuit</span> Electrical "resonator" circuit, consisting of inductive and capacitive elements with no resistance

An LC circuit, also called a resonant circuit, tank circuit, or tuned circuit, is an electric circuit consisting of an inductor, represented by the letter L, and a capacitor, represented by the letter C, connected together. The circuit can act as an electrical resonator, an electrical analogue of a tuning fork, storing energy oscillating at the circuit's resonant frequency.

An elliptic filter is a signal processing filter with equalized ripple (equiripple) behavior in both the passband and the stopband. The amount of ripple in each band is independently adjustable, and no other filter of equal order can have a faster transition in gain between the passband and the stopband, for the given values of ripple. Alternatively, one may give up the ability to adjust independently the passband and stopband ripple, and instead design a filter which is maximally insensitive to component variations.

<span class="mw-page-title-main">Electronic filter topology</span> Electronic filter circuits defined by component connection

Electronic filter topology defines electronic filter circuits without taking note of the values of the components used but only the manner in which those components are connected.

<span class="mw-page-title-main">Zobel network</span>

Zobel networks are a type of filter section based on the image-impedance design principle. They are named after Otto Zobel of Bell Labs, who published a much-referenced paper on image filters in 1923. The distinguishing feature of Zobel networks is that the input impedance is fixed in the design independently of the transfer function. This characteristic is achieved at the expense of a much higher component count compared to other types of filter sections. The impedance would normally be specified to be constant and purely resistive. For this reason, Zobel networks are also known as constant resistance networks. However, any impedance achievable with discrete components is possible.

<span class="mw-page-title-main">Lattice phase equaliser</span> Type of signal processing filter

A lattice phase equaliser or lattice filter is an example of an all-pass filter. That is, the attenuation of the filter is constant at all frequencies but the relative phase between input and output varies with frequency. The lattice filter topology has the particular property of being a constant-resistance network and for this reason is often used in combination with other constant-resistance filters such as bridge-T equalisers. The topology of a lattice filter, also called an X-section, is identical to bridge topology. The lattice phase equaliser was invented by Otto Zobel using a filter topology proposed by George Campbell.

Constant k filters, also k-type filters, are a type of electronic filter designed using the image method. They are the original and simplest filters produced by this methodology and consist of a ladder network of identical sections of passive components. Historically, they are the first filters that could approach the ideal filter frequency response to within any prescribed limit with the addition of a sufficient number of sections. However, they are rarely considered for a modern design, the principles behind them having been superseded by other methodologies which are more accurate in their prediction of filter response.

m-derived filters or m-type filters are a type of electronic filter designed using the image method. They were invented by Otto Zobel in the early 1920s. This filter type was originally intended for use with telephone multiplexing and was an improvement on the existing constant k type filter. The main problem being addressed was the need to achieve a better match of the filter into the terminating impedances. In general, all filters designed by the image method fail to give an exact match, but the m-type filter is a big improvement with suitable choice of the parameter m. The m-type filter section has a further advantage in that there is a rapid transition from the cut-off frequency of the passband to a pole of attenuation just inside the stopband. Despite these advantages, there is a drawback with m-type filters; at frequencies past the pole of attenuation, the response starts to rise again, and m-types have poor stopband rejection. For this reason, filters designed using m-type sections are often designed as composite filters with a mixture of k-type and m-type sections and different values of m at different points to get the optimum performance from both types.

mm'-type filters, also called double-m-derived filters, are a type of electronic filter designed using the image method. They were patented by Otto Zobel in 1932. Like the m-type filter from which it is derived, the mm'-type filter type was intended to provide an improved impedance match into the filter termination impedances and originally arose in connection with telephone frequency division multiplexing. The filter has a similar transfer function to the m-type, having the same advantage of rapid cut-off, but the input impedance remains much more nearly constant if suitable parameters are chosen. In fact, the cut-off performance is better for the mm'-type if like-for-like impedance matching are compared rather than like-for-like transfer function. It also has the same drawback of a rising response in the stopband as the m-type. However, its main disadvantage is its much increased complexity which is the chief reason its use never became widespread. It was only ever intended to be used as the end sections of composite filters, the rest of the filter being made up of other sections such as k-type and m-type sections.

General m<sub>n</sub>-type image filter

These filters are electrical wave filters designed using the image method. They are an invention of Otto Zobel at AT&T Corp. They are a generalisation of the m-type filter in that a transform is applied that modifies the transfer function while keeping the image impedance unchanged. For filters that have only one stopband there is no distinction with the m-type filter. However, for a filter that has multiple stopbands, there is the possibility that the form of the transfer function in each stopband can be different. For instance, it may be required to filter one band with the sharpest possible cut-off, but in another to minimise phase distortion while still achieving some attenuation. If the form is identical at each transition from passband to stopband the filter will be the same as an m-type filter. If they are different, then the general case described here pertains.

<span class="mw-page-title-main">Commensurate line circuit</span>

Commensurate line circuits are electrical circuits composed of transmission lines that are all the same length; commonly one-eighth of a wavelength. Lumped element circuits can be directly converted to distributed-element circuits of this form by the use of Richards' transformation. This transformation has a particularly simple result; inductors are replaced with transmission lines terminated in short-circuits and capacitors are replaced with lines terminated in open-circuits. Commensurate line theory is particularly useful for designing distributed-element filters for use at microwave frequencies.

<span class="mw-page-title-main">RLC circuit</span> Resistor Inductor Capacitor Circuit

An RLC circuit is an electrical circuit consisting of a resistor (R), an inductor (L), and a capacitor (C), connected in series or in parallel. The name of the circuit is derived from the letters that are used to denote the constituent components of this circuit, where the sequence of the components may vary from RLC.

<span class="mw-page-title-main">Frequency selective surface</span> Optical filter

A frequency-selective surface (FSS) is any thin, repetitive surface designed to reflect, transmit or absorb electromagnetic fields based on the frequency of the field. In this sense, an FSS is a type of optical filter or metal-mesh optical filters in which the filtering is accomplished by virtue of the regular, periodic pattern on the surface of the FSS. Though not explicitly mentioned in the name, FSS's also have properties which vary with incidence angle and polarization as well - these are unavoidable consequences of the way in which FSS's are constructed. Frequency-selective surfaces have been most commonly used in the radio frequency region of the electromagnetic spectrum and find use in applications as diverse as the aforementioned microwave oven, antenna radomes and modern metamaterials. Sometimes frequency selective surfaces are referred to simply as periodic surfaces and are a 2-dimensional analog of the new periodic volumes known as photonic crystals.

Staggered tuning is a technique used in the design of multi-stage tuned amplifiers whereby each stage is tuned to a slightly different frequency. In comparison to synchronous tuning it produces a wider bandwidth at the expense of reduced gain. It also produces a sharper transition from the passband to the stopband. Both staggered tuning and synchronous tuning circuits are easier to tune and manufacture than many other filter types.

References

  1. Matthaei et al., pp. 96–97.
  2. Matthaei et al., pp. 412–413.
  3. Matthaei et al., pp. 438–440.
  4. Farago, p. 69.
  5. Matthaei et al., pp. 727–729.
  6. Zobel, 1930, p. 3.

Bibliography