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Bilinear transform

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Thebilinear transform(also known asTustin's method,afterArnold Tustin) is used indigital signal processingand discrete-timecontrol theoryto transform continuous-time system representations to discrete-time and vice versa.

The bilinear transform is a special case of aconformal mapping(namely, aMöbius transformation), often used for converting atransfer functionof alinear,time-invariant(LTI) filter in thecontinuous-time domain (often named ananalog filter) to a transfer functionof a linear, shift-invariant filter in thediscrete-time domain (often named adigital filteralthough there are analog filters constructed withswitched capacitorsthat are discrete-time filters). It maps positions on theaxis,,in thes-planeto theunit circle,,in thez-plane.Other bilinear transforms can be used for warping thefrequency responseof any discrete-time linear system (for example to approximate the non-linear frequency resolution of the human auditory system) and are implementable in the discrete domain by replacing a system's unit delayswith first orderall-pass filters.

The transform preservesstabilityand maps every point of thefrequency responseof the continuous-time filter,to a corresponding point in the frequency response of the discrete-time filter,although to a somewhat different frequency, as shown in theFrequency warpingsection below. This means that for every feature that one sees in the frequency response of the analog filter, there is a corresponding feature, with identical gain and phase shift, in the frequency response of the digital filter but, perhaps, at a somewhat different frequency. The change in frequency is barely noticeable at low frequencies but is quite evident at frequencies close to theNyquist frequency.

Discrete-time approximation[edit]

The bilinear transform is a first-orderPadé approximantof the natural logarithm function that is an exact mapping of thez-plane to thes-plane. When theLaplace transformis performed on a discrete-time signal (with each element of the discrete-time sequence attached to a correspondingly delayedunit impulse), the result is precisely theZ transformof the discrete-time sequence with the substitution of

whereis thenumerical integrationstep size of thetrapezoidal ruleused in the bilinear transform derivation;[1]or, in other words, the sampling period. The above bilinear approximation can be solved foror a similar approximation forcan be performed.

The inverse of this mapping (and its first-order bilinearapproximation) is

The bilinear transform essentially uses this first order approximation and substitutes into the continuous-time transfer function,

That is

Stability and minimum-phase property preserved[edit]

A continuous-time causal filter isstableif thepolesof its transfer function fall in the left half of thecomplexs-plane.A discrete-time causal filter is stable if the poles of its transfer function fall inside theunit circlein thecomplex z-plane.The bilinear transform maps the left half of the complex s-plane to the interior of the unit circle in the z-plane. Thus, filters designed in the continuous-time domain that are stable are converted to filters in the discrete-time domain that preserve that stability.

Likewise, a continuous-time filter isminimum-phaseif thezerosof its transfer function fall in the left half of the complex s-plane. A discrete-time filter is minimum-phase if the zeros of its transfer function fall inside the unit circle in the complex z-plane. Then the same mapping property assures that continuous-time filters that are minimum-phase are converted to discrete-time filters that preserve that property of being minimum-phase.

Transformation of a General LTI System[edit]

A generalLTI systemhas the transfer function The order of the transfer functionNis the greater ofPandQ(in practice this is most likelyPas the transfer function must beproperfor the system to be stable). Applying the bilinear transform whereKis defined as either2/Tor otherwise if usingfrequency warping,gives Multiplying the numerator and denominator by the largest power of(z+ 1)−1present,(z+ 1)-N,gives It can be seen here that after the transformation, the degree of the numerator and denominator are bothN.

Consider then the pole-zero form of the continuous-time transfer function The roots of the numerator and denominator polynomials,ξiandpi,are thezeros and polesof the system. The bilinear transform is aone-to-one mapping,hence these can be transformed to the z-domain using yielding some of the discretized transfer function's zeros and polesξ'iandp'i As described above, the degree of the numerator and denominator are now bothN,in other words there is now an equal number of zeros and poles. The multiplication by(z+ 1)-Nmeans the additional zeros or poles are [2] Given the full set of zeros and poles, the z-domain transfer function is then

Example[edit]

As an example take a simplelow-passRC filter.This continuous-time filter has a transfer function

If we wish to implement this filter as a digital filter, we can apply the bilinear transform by substituting forthe formula above; after some reworking, we get the following filter representation:

The coefficients of the denominator are the 'feed-backward' coefficients and the coefficients of the numerator are the 'feed-forward' coefficients used for implementing a real-timedigital filter.

Transformation for a general first-order continuous-time filter[edit]

It is possible to relate the coefficients of a continuous-time, analog filter with those of a similar discrete-time digital filter created through the bilinear transform process. Transforming a general, first-order continuous-time filter with the given transfer function

using the bilinear transform (without prewarping any frequency specification) requires the substitution of

where

.

However, if the frequency warping compensation as described below is used in the bilinear transform, so that both analog and digital filter gain and phase agree at frequency,then

.

This results in a discrete-time digital filter with coefficients expressed in terms of the coefficients of the original continuous time filter:

Normally the constant term in the denominator must be normalized to 1 before deriving the correspondingdifference equation.This results in

The difference equation (using theDirect form I) is

General second-order biquad transformation[edit]

A similar process can be used for a general second-order filter with the given transfer function

This results in a discrete-timedigital biquad filterwith coefficients expressed in terms of the coefficients of the original continuous time filter:

Again, the constant term in the denominator is generally normalized to 1 before deriving the correspondingdifference equation.This results in

The difference equation (using theDirect form I) is

Frequency warping[edit]

To determine the frequency response of a continuous-time filter, thetransfer functionis evaluated atwhich is on theaxis. Likewise, to determine the frequency response of a discrete-time filter, the transfer functionis evaluated atwhich is on the unit circle,.The bilinear transform maps theaxis of thes-plane (which is the domain of) to the unit circle of thez-plane,(which is the domain of), but it isnotthe same mappingwhich also maps theaxis to the unit circle. When the actual frequency ofis input to the discrete-time filter designed by use of the bilinear transform, then it is desired to know at what frequency,,for the continuous-time filter that thisis mapped to.

This shows that every point on the unit circle in the discrete-time filter z-plane,is mapped to a point on theaxis on the continuous-time filter s-plane,.That is, the discrete-time to continuous-time frequency mapping of the bilinear transform is

and the inverse mapping is

The discrete-time filter behaves at frequencythe same way that the continuous-time filter behaves at frequency.Specifically, the gain and phase shift that the discrete-time filter has at frequencyis the same gain and phase shift that the continuous-time filter has at frequency.This means that every feature, every "bump" that is visible in the frequency response of the continuous-time filter is also visible in the discrete-time filter, but at a different frequency. For low frequencies (that is, whenor), then the features are mapped to aslightlydifferent frequency;.

One can see that the entire continuous frequency range

is mapped onto the fundamental frequency interval

The continuous-time filter frequencycorresponds to the discrete-time filter frequencyand the continuous-time filter frequencycorrespond to the discrete-time filter frequency

One can also see that there is a nonlinear relationship betweenandThis effect of the bilinear transform is calledfrequency warping.The continuous-time filter can be designed to compensate for this frequency warping by settingfor every frequency specification that the designer has control over (such as corner frequency or center frequency). This is calledpre-warpingthe filter design.

It is possible, however, to compensate for the frequency warping by pre-warping a frequency specification(usually a resonant frequency or the frequency of the most significant feature of the frequency response) of the continuous-time system. These pre-warped specifications may then be used in the bilinear transform to obtain the desired discrete-time system. When designing a digital filter as an approximation of a continuous time filter, the frequency response (both amplitude and phase) of the digital filter can be made to match the frequency response of the continuous filter at a specified frequency,as well as matching at DC, if the following transform is substituted into the continuous filter transfer function.[3]This is a modified version of Tustin's transform shown above.

However, note that this transform becomes the original transform

as.

The main advantage of the warping phenomenon is the absence of aliasing distortion of the frequency response characteristic, such as observed withImpulse invariance.

See also[edit]

References[edit]

  1. ^Oppenheim, Alan (2010).Discrete Time Signal Processing Third Edition.Upper Saddle River, NJ: Pearson Higher Education, Inc. p. 504.ISBN978-0-13-198842-2.
  2. ^ Bhandari, Ayush."DSP and Digital Filters Lecture Notes"(PDF).Archived fromthe original(PDF)on 3 March 2022.Retrieved16 August2022.
  3. ^Astrom, Karl J. (1990).Computer Controlled Systems, Theory and Design(Second ed.). Prentice-Hall. p. 212.ISBN0-13-168600-3.

External links[edit]