EXPERT SYSTEMS AND SOLUTIONS

Email: expertsyssol@gmail.com expertsyssol@yahoo.com Cell: 9952749533 www.researchprojects.info PAIYANOOR, OMR, CHENNAI Call For Research Projects Final year students of B.E in EEE, ECE, EI, M.E (Power Systems), M.E (Applied Electronics), M.E (Power Electronics) Ph.D Electrical and Electronics. Students can assemble their hardware in our Research labs. Experts will be guiding the projects.

IIR Filter Design

Introduction


IIR filter have infinite-duration impulse responses, hence they can be matched to analog filters, all of which generally have infinitely long impulse responses. The basic technique of IIR filter design transforms well-known analog filters into digital filters using complex-valued mappings. The advantage of this technique lies in the fact that both analog filter design (AFD) tables and the mappings are available extensively in the literature.

Introduction


The basic technique is called the A/D filter transformation. However, the AFD tables are available only for lowpass filters. We also want design other frequencyselective filters (highpass, bandpass, bandstop, etc.) To do this, we need to apply frequency-band transformations to lowpass filters. These transformations are also complex-valued mappings, and they are also available in the literature.

Two approaches
Design analog lowpass filter Apply Freq. band transformation s-->s Apply filter transformation s-->z Desired IIR filter

Approach 1, used in Matlab

Design analog lowpass filter Apply filter transformation s-->z Apply Freq. band transformation z-->z Desired IIR filter

Approach 2, study

IIR Filter Design Steps

 

Design analog lowpass filter Study and apply filter transformations to obtain digital lowpass filter Study and apply frequency-band transformations to obtain other digital filters from digital lowpass filter

The main problem



We have no control over the phase characteristics of the IIR filter. Hence IIR filter designs will be treated as magnitude-only designs.

Main Content of This Chapter



Analog filter specifications and the properties of the magnitude-squared response used in specifying analog filters. Characteristics of three widely used analog filters


Butterworth, Chebyshev, and Elliptic filter

Transformation to convert these prototype analog filters into different frequency-selective digital filter

Some Preliminaries


Relative linear scale



The lowpass filter specifications on the magnitude-squared response are given by 1 e| H a ( j;) |e 1, | ; |e ; p 2 1 I 1 0 e| H a ( j;) |e 2 , ; s e| ; | A

Where epsilon is a passband ripple parameter, Omega_p is the passband cutoff frequency in rad/sec, A is a stopband attenuation parameter, and Omega_s is the stopband cutoff in rad/sec.

1 a ( j;) ! 1 I 2
2

at ; ! ; p

1 | H a ( j;) | ! 2 A
2

at ; ! ; s

The relations among epsilon, A, Rp, As, delta1 and delta2

1 R p / 10 R p ! 10 log10 I ! 10 1 2 1 I 1 As / 20 As ! 10 log10 2 A ! 10 A 2 H1 1  H1 1 ! I ! 2 1 H2 1 I 1  H1 1 1  H1 H2 ! A! 1  H1 A H2

Properties of |Ha
H a ( j ; ) ! H a ( s ) | s ! j;

2 (jOmega)|

Analog filter specifications, which are given in terms of the magnitude-squared response, contain no phase information. Now to evaluate the s-domain system function Ha(s), consider
* | H a ( j;) | 2 ! H a ( j;) H a ( j;) ! H a ( j;) H a (  j;) ! H a ( s ) H a (  s ) | s ! j ;

or , H a ( s ) H a ( s ) !| H a ( j;) |2 |; ! s / j
Therefore the poles and zeros of the magnitude-squared function are distributed in a mirror-image symmetry with respect to the jOmega axis. Also for real filters, poles and zeros occur in complex conjugate pairs (or mirror-image symmetry with respect to the real axis). From this pattern we can construct Ha(s), which is the system function of our analog filter.

Properties of |Ha


2 (jOmega)|

We want Ha(s) to represents a causal and stable filter. Then all poles of Ha(s) must lie within the left halfplane. Thus we assign all left-half poles of Ha(s)Ha(-s) to Ha(s). We will choose the zeros of Ha(s)Ha(-s) lying inside or on the jOmega axis as the zeros of Ha(s). The resulting filter is then called a minimum-phase filter.

Characteristics of Prototype Analog Filters



IIR filter design techniques rely on existing analog filter filters to obtain digital filters. We designate these analog filters as prototype filters. Three prototypes are widely used in practice
  

Butterworth lowpass Chebyshev lowpass (Type I and II) Elliptic lowpass

Butterworth Lowpass Filter



This filter is characterized by the property that its magnitude response is flat in both passband and stopband.
The magnitude-squared response of an N-order lowpass filter

| H a ( j; ) | !
2

; 1 ; c

2N

Omega_c is the cutoff frequency in rad/sec.

| H a ( j; ) | !
2

1 ; 1 ; c
2N

1.2

1 N = 100 0.8 N= 1 0.6 N= 2 N = 200

0.4

0.2

0 0 10 20 30 40 50 60 70 80 90 100

Properties of Butterworth Filter



Magnitude Response
  

|Ha(0)|2 =1, |Ha(j c) |2 =0.5, for all N (3dB attenuation at |Ha(j )|2 monotonically decrease for Approaches to ideal filter when N
( j; c ) 2 N ( j; c ) 2 N ! 2N ! 2N 2N s  ( j; c ) ( s  pk )
k !1

c)

To determine the system function Ha(s)

H a ( s) H a ( s) ! 1 s 1 j; c
2N

pk ! ( 1) ( j; c ) ! ; c e

1 2N

j 2TN ( 2 k  N 1)

, k ! 0,1, . ,2 N  1

Poles of |Ha(j )|2



=s/j

=Ha(s) Ha(-s)

Equally distributed on a circle of radius c with angular spacing of pi/N radians

 

For N odd, pk= cej2pik/N For N even, pk= cej(pi/2N+kpi/N)

 

Symmetry respect to the imaginary axis A pole never falls on the imaginary axis, and falls on the real axis only if N is odd A stable and causal filter Ha(s) can now be specified by selecting poles in the left half-plane N
a (s) !

;c

L P poles

( s  pk )

Matlab Implementation


Function [z,p,k] = buttap(N)



To design a normalized ( c=1) Butterworth analog prototype filter of order N Z: zeros; p: poles; k: gain value Provide a direct form (or numerator-denominator) structure. Convert the direct form into the cascade form

Function [b,a] = u_buttap(N,Omegac)



Function [C,B,A] = sdir2cas(b,a)



Design Equations
The analog lowpass filter is specified by the parameters, Omega_p. R_p,Omega_s, and A_s. Therefore the essence of the design in the case of Butterworth filter is to obtain the order N and the cutoff frequency Omega_c.
| H a ( j;) |2 ! 1
; 1 ; c
2N

at ; ! ; p ,  10 log10 at ; ! ; s ,  10 log10

( j; ) 2 ! R p a ( j;) 2 ! As a

log10 [(10 R p /10  1) /(10 As /10  1)] N ! 2 log10 (; p / ; s ) ;p ;s , or ; c ! ;c ! R /10 2N 2N (10 As /10  1) (10 p  1)

Matlab Implementation


Function [b,a] = afd_butt(Wp,Ws,Rp,As)



To design an analog Butterworth lowpass filter, given its specifications. Magnitude response in absolute as well as in relative dB scale and the phase response.

Function [db,mag,pha,w] = freqs_m(b,a,wmax)



Chebyshev Lowpass Filter



Chebyshev-I filters


Have equiripple response in the passband Have equiripple response in stopband

Chebyshev-II filters


Butterworth filters
Have monotonic response in both bands We note that by choosing a filter that has an equripple rather than a monotonic behavior, we can obtain a low-order filter. Therefore Chebyshev filters provide lower order than Buttworth filters for the same specifications.


The magnitude-squared response of Chebyshev-I filter

| H a ( j;) | !
2

; 1 I T ; c
2 2 N

N is the order of the filter, Epsilon is the passband ripple factor Nth-order Chebyshev polynomial

N cos 1 ( x )), 0 e x e 1 cos( TN ( x) ! cosh(cosh 1 ( x )), 1 x g

(a) For 0<x<1, TN(x) oscillates between 1 and 1, and (b) For 1<x<infinity, TN(x) increases monotonically to infinity

Observations
At x=0 (or =0); |Ha(j0)|2 = 1; for N odd; = 1/(1+epson^2); for N even At x=1 (or = c); |Ha(j1)|2= 1/(1+epson^2) for all N.

For 0<=x<=1 (or 0<= <= c) |Ha(jx)|2 oscillates between 1 and 1/(1+epson^2) For x>1 (or > c), |Ha(jx)|2 decreases monotonically to 0 .

At x= r, |Ha(jx)|2 = 1/(A^2).

Causal and stable Ha(s)

To determine a causal and stable Ha(s), we must find the poles of Ha(s)Ha(-s) and select the left half-plane poles for Ha(s). The poles of Ha(s)Ha(-s) are obtained by finding the roots of

s 1 I T j; c
2 2 N

It can be shown that if pk ! W k  j; k , k ! 0,1, . , N  1 are the (left half-plane) roots of the above polynomial, then

pk ! W k  j; k , k ! 0,1, . , N  1 T (2k  1)T W k ! (a; c ) cos  2N 2 T (2k  1)T ; k ! (b; c ) sin  2 2N 1 N 1 N N E  1/ E , b ! E  N 1/ E , a! 2 2 1 1 E !  1 2 I I

The poles of Ha(s)Ha(-s)

The poles fall on an ellipse with major axis b c and minor axis a c . Now the system function is

K H a ( s) ! 4 ( s  pk )
k

Left half-plane K is a normalizing factor

Matlab Implementation


Function [z,p,k] = cheb1ap(N,Rp)



To design a normalized Chebyshev-I analog prototype filter of order N and passband ripple Rp. Z array: zeros; poles in p array, gain value k Return Ha(s) in the direct form

Function [b,a] = u_chblap(N,Rp,Omegac)



Design Equations
Given p, s, Rp and As, three parameters are required to determine a Chebyshev-I filter
I ! 10 0.1Rp  1, and A ! 10 As / 20 ;s ; c ! ; p , and ; r ! ;c g ! ( A  1) / I
2 2

Function

log g  g 2  1 10 N ! 2 log10 ; r  ; r  1

? ?

A A

[b,a]=afd_chb1(Wp,Ws,Rp,As)

Chebyshev-II filter
Related to the Chebyshev-I filter through a simple transformation. It has a monotone passband and an equiripple stopband, which implies that this filter has both poles and zeros in the s-plane. Therefore the group delay characteristics are better (and the phase response more linear) in the passband than the Chebyshev-I prototype.

| H a ( j; ) | !
2

1  I T ; c / ;
2 2 N

1

Matlab implementation


Function [z,p,k] = cheb2ap(N,As);



normalized Chebyshev-II Unnormalized Chebyshev-II

Function [b,a] = u_chb2ap(N,As,Omegac)



Function [b,a] = afd_chb2(Wp,Ws,Rp,As)

Elliptic Lowpass Filters

These filters exhibit equiripple behavior in the passband as well as the stopband. They are similar in magnitude response characteristics to the FIR equiripple filters. Therefore elliptic filters are optimum filters in that they achieve the minimum order N for the given specifications These filters, for obvious reasons, are very difficult to analyze and therefore, to design. It is not possible to design them using simple tools, and often programs or tables are needed to design them.

The magnitude-squared response

| H a ( j; ) | !
2

1 ; 1 I U ; c
2 2 N

N: the order; epsilon: passbang ripple; UN() is the Nth order Jacobian elliptic function

Typical responses for odd and even N Computation of filter order N:

 , k ! ; N! ; K (k ) K  1  k
K ( k ) K 1  k12
1 2

p s

, k1 !

I A 1
2

, K ( x) ! 0

T /2

dU 1  x 2 sin 2 U

Matlab Implementation


[z,p,k]=ellipap(N,Rp,As);


normalized elliptic analog prototype Unnormalized elliptic analog prototype Analog Lowpass Filter Design: Elliptic

[b,a] = u_ellipap(N,Rp,As,Omegac)


[b,a] = afd_elip(Wp,Ws,Rp,As)


Phase Responses of Prototype Filters



Elliptic filter provide optimal performance in the magnitudesquared response but have highly nonlinear phase response in the passband (which is undesirable in many applications). Even though we decided not to worry about phase response in our design, phase is still an important issue in the overall system. At other end of the performance scale are the Buttworth filters, which have maximally flat magnitude response and require a higher-order N (more poles) to achieve the same stopband specification. However, they exhibit a fairly linear phase response in their passband.

Phase Responses of Prototype Filters



The Chebyshev filters have phase characteristics that lie somewhere in between. Therefore in practical applications we do consider Butterworth as well as Chebyshev filters, in addition to elliptic filters. The choice depends on both the filter order (which influences processing speed and implementation complexity) and the phase characteristics (which control the distortion).

Analog-to-digital Filter Transformations

 

After discussing different approaches to the design of analog filters, we are now ready to transform them into digital filters. These transformations are derived by preserving different aspects of analog and digital filters.


Impulse invariance transformation



Preserve the shape of the impulse response from A to D filter Convert a differential eq. representation into a corresponding difference eq. Preserve the shape of the step response Preserve the system function representation from A to D domain

Finite difference approximation technique



Step invariance


Bilinear transformation


Impulse Invariance Transformation



The digital filter impulse response to look similar to that of a frequency-selective analog filter. Sample ha(t) at some sampling interval T to obtain h(n): h(n)=ha(nT) jw j; T

w ! ;T or e

!e

Since z=ejw on the unit circle and s=j on the imaginary axis, we have the following transformation from the s-plane to the zplane: z=esT
H ( z) ! 1
g

k ! g

2T k as  j

Frequency-domain aliasing formula

Complex-plane mapping in impulse invariance transformation

Properties:


Sigma = Re(s):
  

Sigma < 0, maps into |z|<1 (inside of the UC) Sigma = 0, maps into |z|=1 (on the UC) Sigma >0, maps into |Z|>1 (outside of the UC) Every semi-infinite left strip (so the whole left plane) maps to inside of unit circle

Many s to one z mapping: many-to-one mapping



 

Causality and Stability are the same without changing; Aliasing occur if filter not exactly band-limited

Given the digital lowpass filter specifications wp,ws,Rp and As, we want to determine H(z) by first designing an equivalent analog filter and then mapping it into the desired digital filter. Design Procedure: 1. Choose T and determine the analog frequencies: p=wp/T, s=ws/T 2. Design an analog filter Ha(s) using the specifications with one of the three prototypes of the previous section. 3. Using partial fraction expansion, expand Ha(s) into Rk N H a ( s ) ! k !1 s  pk 4. Now transform analog poles {pk} into digital poles {epkT} to obtain the digital filter Rk N H ( z ) ! k !1 1  e pk T z 1

Advantages of Impulse Invariance Mapping



It is a stable design and the frequencies linearly related. Disadvantage



and w are

We should expect some aliasing of the analog frequency response, and in some cases this aliasing is intolerable.

Consequently, this design method is useful only when the analog filter is essentially band-limited to a lowpass or bandpass filter in which there are no oscillations in the stopband.

Bilinear Transformation


This mapping is the best transformation method.

2 1  z 1 1 s / 2 s! z! 1 1 s / 2 T 1 z

T T sz  s  z  1 ! 0 2 2

Linear fractional transformation

Complex-plane mapping in bilinear transformation

Observations


Sigma < 0 |z| < 1, Sigma = 0 |z| = 1, Sigma > 0 |z| > 1

The entire left half-plane maps into the inside of the UC. This is a stable transformation.  The imaginary axis maps onto the UC in a one-to-one fashion. Hence there is no aliasing in the frequency domain. Relation of to is nonlinear = 2tan-1( T/2) =2tan( /2)/T;


Function [b,a] = bilinear(c,d,Fs)

Given the digital filter specifications wp,ws,Rp and As, we want to determine H(z). The design steps in this procedure are the following: 1. Choose a value for T. this is arbitrary, and we may set T = 1. 2. Prewarp the cutoff freq.s wp and ws; that is calculate p and s using (8.28) : p=2/T*tan(wp/2), s=2/T*tan(ws/2) 3. Design an analog filter Ha(s) to meet the specifications. 4. Finally set 2 1  z 1 H ( z) ! H a T 1  z 1 And simplify to obtain H(z) as a rational function in z-1

Advantage of the bilinear Transformation

  

It is a stable design There is no aliasing There is no restriction on the type of filter that can be transformed.

Lowpass Design Using Matlab

     

Matlab Function: [b, a] = butter(N, wn) [b, a] = cheby1(N, Rp, wn) [b, a] = cheby2(N, As, wn) [b, a] = ellip(N, Rp, As, wn) Buttord, cheb1ord, cheb2ord, ellipord can provide filter order N and filter cutoff frequency wn, given the specification.

Lowpass filter design

    

Digital filter specification Analog prototype specifications Analog prototype order calculation,Omega_c,wn Digital Filter design (four types) dir2cas

Digital filter design examples

For the same specifications, do the following:  Ex8.21: Butterworth LP filter design; w p ! 0.2T , R p ! 1dB  Ex8.22: Chebyshev-I LP filter design; ws ! 0.3T , As ! 15dB  Ex8.23: Chebyshev-II LP filter design;  Ex8.24: Elliptic LP filter design; Prototype Butterworth Chebyshev-I Elliptic Order N 6 4 3 Stopband Att. 15 25 27

Frequency-band Transformation


Design other kinds of filters:

  

High-pass filters Band-pass filters Band-stop filters

Using the results of Low-pass filter and Frequency Band Transformation

Frequency-band Transformation
Let HLP(Z) be the given prototype lowpass digital filter, and let H(z) be the desired frequency-selective digital filter. Define a mapping of the form 1 1
Z ! G ( z ) such that H ( z ) ! H LP ( Z )
Z 1 !G ( z 1 )

Given that HLP(Z) is a stable and causal filter, we also want H(z) to be stable and causal. This imposes the following requirements: 1. G( ) must be a rational function in 1/z so that H(z) is implementable. 2. The unit circle of the Z-plane must map onto the unit circle of the z-plane. 3. For stable filters, the inside of the unit circle of the Z-plane must also map onto the inside of the unit circle of the z-plane.

Frequency Transformation for Digital Filters

The general form of the function G() that satisfies the above requirements is a rational function of the all-pass type given by

z 1  E k Z 1 ! G ( z 1 ) ! s4 n !1 , k 1 1Ek z

Ek

Now by choosing an appropriate order n and the coefficients {alpah_k}, we can obtain a variety of mappings. The most widely used transformations are given in Table

Use zmapping function to perform the LP-to-HP transformation

      

Digital lowpass filter specifications Analog prototype specification Analogy Chebyshev prototype filter calculation Bilinear transformation Digital highpass filter cutoff frequency Lp-to-HP frequency-band transformation dir2cas

Comparison of FIR vs. IIR Filters



 

In the case of FIR filters these optimal filters are the equiripple filters designed via the parks-McClellan algorithm, while in the case of IIR filters these are the elliptic filters. For FIR filter the standard realization is the linear-phase direct form, while for elliptic filters cascade forms are widely used. Comparisons: Multiplications per output sample For most applications IIR elliptic filter are desirable from the computational point of view. If we take into account the phase equalizers, then FIR filter designs look good because of their exact linear-phase characteristics.