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‘ hat od "ni design @ digital filter using Impulse Invariance meth analog filter. jie given specifications, find H, (8 ), the transfer ne 3 the sampling rate of the digital filter, T seconds rial pole filters. the analog filter transfer function as the sum o} N Ck a0) = ee) f using the formula Pute the 2-transform of the digital filter by using He) = Loycat “hig Sampling rates use 5.40 Digital Signal Processing Example 5.11 For the analog transfer function J1(#) = teerftazay determine H(z) using impulse invariance method. Assume T = 1 sec. Solution 2 Sn EO)" GaGa) Using partial fraction we can write A=(4 Verio =A rotten. stl s+2 et » | pean o2 = 6+ erie, w MOS aig aga =-2 4 a ee UE =) Using impulse invariance technique we have, if - Ny Ne H(s)=)>—*— then H(z) = Se ile., (8 — py) is transformed to 1 — e?*T 2-1, ‘There are two poles p, = —1 and p = —2. So EQ) «2 = + l=eTe-l” [erat For T = 1 sec 3 2 40)= [Sale 2 2 “ T=0s07ee1 ~ T—o1egeT ny 0.4652-2 HG) = ope oe Infinite Impulse Response Filters 5.41 Solution : Gives He) = ae a) 1 1 A(t) = L7'[H(s)] = LD [sen] “agar 1 = Vir leatayl = Vie VT in(t/ v2) -} ; aay) of , lata wees - . : / A(nT) = V3. eosin i : UT = 100 i Mo) = Vein Wg gin tf H(z) = 2 ete ten V2 (2) = Zain) “| cae a | oe 0.4532-! ' eT 02s? { Feample $13 pes is ea me | fas A3 Desin third order Buterworth digit ter using impulse invariant Me. Assume sampling period T = 1 sec, Solution : ry it i. » Berghe ef 2 = 3, te fr anton anomie Batrvona { Na, Fs ay (s+) +s41) , wit ce BU Nh Ce . 841 * ayo 470806 * 5E08 — JOBE 5.42 Digital Signal Processing 1 = foaO8 a ee ea = (8+ 05 + 50.865) (5 5015 + 70.866) (9 +0.5 ~ 30.866) 0.570.866 1 ee FOS = 70.866 + 1)(—70.866 — 70.866) 1 Z. * = Tj1-732(0.5 — j0.866) —70.866—1.5 = ee 0.5 + 0.288 C=B = an — j0.288 ct : a eeateT 2 toents Therefore, i) = an -0s + 0.288 =0.5 — 50.288 Toertert + [a g-tngssoseegmt + [—g-08,y0860q-1 Nn s | if H)=S°>*, then He) fei a 1 -1+0.662-1 » ‘ T= 0.36821 * [078621 + 00a , C* Od Example 5.14 Apply impulse lover method and find H(z) for H(s) = wae tf * wk Solution The inverse Laplace transform of given function is A(t) = {; *eos(be) for ¢>0 otherwise | Infinite 1 Impulse Response Fi ‘sampling the function produces mesieniid3 A(nT) = (ealgany for n>o es otherwise H(z) = oe? cos(n)z-n & & Baral 1 2 Ml iMs (0-387 p-1yn 2 {ce (a-jb)T 5. 1 +(e east] me [ 1 1 2[I-e ie + ar ~ 1 e=*T cos(bT)2-1 1 26a cos(bT)2-1 + Maz? i dC 10 Bumple 5.15 An analog filter hi fee ic = 8 filter has ‘a transfer function H(s) 7a, : I eT 4 digital filter equivalent to this using impulse invariant method for 7’ = 0.2 Seltion Given _ | ~ +78 +10 | -333 | 3.33 +2. 8-(-8) | o-(-2) | H(s) Ig +5 “NB Eq. (5.816) we have Bre 253, 8488 ] p{_-3.33 333 JL — ees a] -02| St ee Bigs 0.666 2 [a+ ta] \ 2012! yu —— Nt z i Problem 5.7. An analog filter has a transfer function f | 4 r 5 ‘iy (0) = syeqe tie +6 vere ntact font * tt tte equivalent otis wing impulse - ‘ 5.44 Digital Signal Processing Practice Problem 5.8 An analog filter has ‘transfer function He) = s+3 (8) = By 65+ Design a digital filter equivalent to this using impulse invariant method for T= 1 se, 5.123 Design of HR Filter Using Bilinear Transformation ‘The bilinear transformation js. conformal mapping that transforms the j9 axis into the unit cicle in the 2-plane only once thus avoiding aliasing of frequency compe rents, Furthermore, all points in the LHP of ‘s’ are mapped inside the cunt circle in the z-plane and all points in the RHP of ‘s’ are mapped into corresponding pois cutside the unt circle in the 2-plane. ‘Letus consider an analog linea filter with system function b a voles % W=s5 3) which can be written 3 5 Fe as X() sta so s¥ (0) + a¥(s) = bX(8) os) ‘This canbe characterized by the differential equation oH) a +ay(t) = be(t) u(t) ea be approximated by the trapezoidal formula. 5 5) w(t) = [ y'(r)dr + ylte) d where y/(t) denotes the derivative of y(t). fc “The approximation of the integral it go fol aT and ty = nT —T yi in in Ba (5.85) by the trapezoids , a of vor) =F (y'(or) + u'r -T)] +907 -1) From the eifferential Bq (5.84) we obtain F) y'(a) = ~ay(n) + bo(T) Substituting Eq4(5.87) in Bq. (5.86) we get f a y(aT) = Ft-av(er) 4 be(nP) — ay(n - T) + belt — mn Lt Infinite Impulse Response Filters 5.45 wich implies E i ry = (2 Z) wnt ~T) = Penn) +2(nT ~ Ty) (5.88) vith o(r) = U(T) and 2(n) = 2(nT) we obtain the res i (+F) y(n) — (:-¥) y(n-1) = sme z-transform of this difference equation is aT ii or (1+ z) BAe (: m eal PY) = Tht e-yx(e) Thesystem function of the digital filter is [t(n) + 2(n — 1)} Y(z) _ Pare) H(z) = OES OSs Fabs ers i ae +274) Dividing numerator and denominator by F4 e, we get \ 6 : H(z) = (==). (5.89) ; Bietataee " eocmeat Eq.(5.82) and Eq.(5.89), the mapping from s-plane to the z-plane can 2 (1-29 < Boheme 5.90) mi eee (3 $at } ‘ : i see between s and z is known as bilinear transformation, } Ska s=o4+52 (5.90a) 0) can be expressed as 2(z~ 1) j ss : TFT) pee +! 2 Trev as ie roe 14 seine 7[H]- p [pee te [fSte— 1+ irae rcosw + 1— Al Set iran. reosw +1 —jrsin [tex wm stapes peane) | crear eee =1+r?sin?u + j2rsinw] tage ne 1? cost + Or cosw +r? sin? ‘w | pe W u SW S109 S00 43) 5.46 Digital Signal Processing Separating imaginary and real parts, we have 2 [ at 2rsinw | (6.908) 9= 7 |Pprtearcosw (7141 + Orcoow, Comparing Eq, (5.90a) and Eq. (5.90b), we have (5.91) a1 sy gies Ari P|Tarts treosw)|' T1417 + 2rcos From Eq, (5:91), we find that fr < 1, then o < O-and ifr > 1, then ¢ > 0. Consequently the LHP in ‘s’ maps into the inside of the unit circle in the z-plane and the RHP in the ‘s’ maps into the outside of the unit circle, When r = 1, then o = 0 o and ieee a f= TT + cow T 200? 2, w = pm 5 or or q = 2tan“! w= Btan! The warping effect Let 9 and w represent the frequency variables in the analog filter and the del digital filter respectively. From Eq, (5.92) we have For small value of w w=9r For low frequencies the relationship between {2 and w are linear, is the digital filter have the same amplitude response as the analog filter. frequencies, however, the relationship between w and £2 becomes non- Fig. 5.24) and distortion is introduced in the frequency scale of the digital that of the analog filter. This is known as the warping effect. ‘The influence of the warping effect on the amplitude response is shown! by considering an analog filter with a number of passbands centered at re vals. The derived digital filter will have same number of passbands. But frequencies and bandwidth of higher frequency passband will tend to educ portionately. Infinite Impulse Response Filters 5.47 . Fig. 5.25 The effect on magnitude response due to warping effect. influence of the warping effect on the phase response is shown in Fig. 5.26. dering an analog filter with linear phase response, the phase response of the digital filter will be non-linear. j é effect can be eliminated by prewarping the analog filter. This can be prewarping analog frequencies using the formula w Be pind ad 9) 5.48 Digital Signal Processing ie a Fig.5.26 The effect on phase response due to warping effect Therefore, we have 5 = ata? 6598) = ptm a a = Sten 5.90) Semen ‘ ‘Steps to design digital fiter using bilinear transform technique. eee aS R jes using fiven specifications, find prewarping analog frequencies mula Q = =tan—, 2. Using the analog frequencies find #1(s) of the analog filter. 3. Select the sampling rate of the digital filter, call it T seconds per sa™pI€ . ai-zt se 8 = FT oa ito the transfer function found in step 2 5.124 The matched z-transfo ‘Another method for convert 3 as ill tap the poles and err of HE) di to an equivalent digi diel it poles and aeros in he #98 M H(s) = are) ie “m) Infinite Impulse Response Filters. 5.49 nore {34} are the zeros and {7} are the poles ofthe filter, then the system function rine digital filter is M TI (1 - "7s H(z) = eh T] (= em?'z-1) ket 65,97) vbere T is the sampling interval. Thus each factor of the form (s ~ a) in H(s) is mapped into the factor 1 ~ e”'2-1, This mapping is called the matched z-transform, 2 Example 5.16 Apply bilinear transformation to H(s) = Grnery secand find H(z). with T = 1 Solution 2 Given HO) = eT Subsiute « = 2 [=] in (5) to get #7(2) T +277 H(z) = Hella 2 * OF AYGF 2) long (4354) ae x : Beample 5.17, ing the bilinear transform, design a highpass filter, monotonic in Passband wit frequency of 1000 Hz and down 104B at 350 Hz. The sampling luency is 5.50 Digital Signal Processing Solution Given ap = 34B; we = Wp = 2 x m x 1000 = 20007 rad/sec Qs =10dB; w= 2 x x 350 = 700n rad/sec 1 1 = = = 2x 10-4sec 5000 T= HOMan ol 3 a Sons Sines Fig. 5.27 ‘The characteristics are monotonic in both passband and stopband. Therefore, th filter is Butterworth filter, Prewarping the digital frequencies we have 28 ag (200m x 2 x 10-4) op =F 2 = 104 tan(0.27) = 7265 rad/sec 2 tan (7008 ee x 1074) 2x10 a = 10* tan(0.07n) = 2235rad/sec First we design a lowpass filter for the given specifications and use suitable formation to obtain transfer function of highpass filter. ‘The order of the filter {9°200) 4 é 9 we RV a 8) ono oe leg 7385 a ‘Therefore, we take N = 1, : ‘The frs-order Buterworth filter for Q= = 1 rads is H(s) = 7+ Infinite Impulse Response Filters 5.51 spe ighpass filter for Qe = vsfomation = 7265 rad/sec can be obtained by using the The transfer function of highpass filter 1 Le cea ae — 8+ 7265 Using bilinear transformation H@)=HO)|_, (2) = ae! 8+ 1265 leat _ 0.5792(1 — 21) © -1=0.15842-1 A ee Example 5.18 Determine H(z) that results when the bilinear transformation is ap- to (0) = 2 ae 8? + 0.6928 + 0.504 Slution ‘nbiinear transformation i H@) =H) ai Assume 7 : = 1 sec. Then =21)7? P = 7] + 4.525 H(z) = 1 Syl a «fi 2 | +0002 [F =| +0504 T4271 1+ 1.4479 + 0.178327} + 1.447927? T= 11875221 + 0.52992-? 45.52 Digital Signal Processing Practice Problem 59 An analog filter has 2 transfer function 1 H(s) = Fy 60+9 n method. Design a digital filter using bilinear transformation Practice Problem 5.10 Repeat practice problem 5.7 using bilinear transforig method. 5.13 Frequency ‘Transformation in Digital Domain ‘A digital lowpass filter can be converted into a digital highpass, bandstop, bes ‘or another digital filter. These transformations are given below. 5.13.1 Lowpass to Lowpass aa zt-a l-az = Salley = p)/21 o> Sin ap + o4)/2] a tip = passband frequency of lowpass filter 4 = passband frequency of new lowpass filter 54132 Lowpass to highpass : ) : ae zita e | where = ~ 28 + 9)/2) conl(uy =u) /2] Wp = passband frequency of lowpass filter W, = passband frequency of highpass filter 5133 Lowpass to Bandpass Infinite Impulse Response Filters 5.53 3 ~ se wy, k= oth [ he “| tan (5.100) wy, = upper cutoff frequency wy = lower cutoff frequency uM Lowpass to Bandstop cos|(ws +1)/2] wre allow —1)/21 % 101) ke = tan|(wa — «1)/2} tan eS - — s —— ample 5.19 Convert the single pole lowpass filter with system function (2) i ay and ot SEE int bandpass iter with upper and lover ct ese eet ae = twa ly. The lowpass filter has 3 4B bandwidth Wp = & and wy = J A i. fier is Ye Ggitatso-digital transformation from lowpass filter 10 8 bandpass ket

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