\TE ACADEMY PUBLICATIONS +The Mentor for Engineers Uc) Unit-wise Coe i tee aceuy According to New Syllabus (May 2008 to May 2017) Copyright © All Rights Reserved by GATE ACADEMY PUBLICATIONS.UNIT-4 LTR Filter Design m CONTEINTIS® Design of Discrete-time IIR filters from Continuous-time Filters Filter design by Impulse invariant and bilinear transformation method Butterworth Chebyshev approximation Filter Elliptic approximation Filter ee 6 HY Frequency transformationnal Processini Digital _—_ 4.1. Analog & Digital Filter = Question 1 Compare analog filter and digital filter. a55| Ans. Comparison of analog filter and digital filter: ea Ti Analog Filter \ Digital Filter ti 7] Analog filter processes analog inputs} 1.| A digital filter processing digital data aver, andgeneratesanalog outputs. ol 2.| Analog filters are constructed from | 2.| A digital filter consists of elements T active or passive electrical components. likeadder, multiplierand delay. Ans: 3.| Analog filter is described by al 3.| Digital filter is described by a differential equation. difference equation. 4.| The frequency response of an analog| 4.| The frequency response can be T filter can bemodlified by changingthe| | changed by changing, the filter! ( components. cvefficients. ( Question 2 How can one design digital filters from analog filters? ( ‘Ans. (a) Map the desired digital filter specification into those for an equivalent analog filter. : (b) Derive the analog transfer function for the analog prototype. ous (0 Transform the transfer function of analog prototype into an equivalent digital os transfer function. Question 3 Mention any two procedures for digitizing the transfer function of an analog filter. eol ‘Ans. The two important procedures for digitizing the transfer function of an analog filter are (a) Impulse invariance method (b) Bilinear transformation method Question 4 5)2- What are the properties that are maintained same in the transfer of analog filter into) digital filter ? [CSVTU Dec 2012, 2011, 2010] ‘Ans. Two properties that are maintained same in the transfer of analog to digital filter: (a) The jO axis in the s-plane should map into the unit circle in the z-plane. Thus thet will bea direct relationship between the two frequency variables in the two domains (b) The left half plane of the s-plane should map into the inside of the unit circle in the z-plane. Thus a stable analog filter will be converted to stable digital filter. 4.2 Filter Design by Approximation of Derivative : Question 5 What do you understand by backward difference? Ans. One of the simplest method for converting an analog filter into a digital fil approximate the differential equation by an equivalent difference equation iter 5 ® »a are ACADEMY PUBLICATIONS ™ IIR Filter Design r The above equation is known as backward difference e question 6 Whatis the mapping procedure between 5-pl -plane i : of differentials? What is its characteristics? Hg ee quation. Ans. The mapping, procedure between s-pl: - f differentials is given by plane and z-plane in the method of mapping of H(2)=H(s)|, ve T The above mapping has the following characteristics : (a) a left half of s-plane maps inside a circle of radius 1/2 centered at z = 1/2 in the z-plane. (b)The right half of s-plane maps into the region outside the circle of radius 1/2 in the z-plane. (Q) The jQ -axis maps on to the perimeter of the circle of radius 1/2 in the z-plane. Question 7 Use the backward difference for the derivative to convert the analog low-pass filter with system function. Sol. The mapping formula for the backward difference for the derivative is given in eq, ie. 1-27 7 4 i The system response of the digital filter is s H(z) =H) IT=1s, H(zZ)= Question 8 2 : ‘Use the backward difference for the derivative and convert the analog filter with system function. 1 He sinaae Sol. Using equation, s4-4 | GATE Digital Signal Proc CADEMY PUBLICATIONS ~ igital § 2e8TONS : mm response of the digital filter's \ The syste! A()=H()|, IfT=1s, Question 9 : ae P An analog filter has the following system function. Convert this filter into a digital filte using backward difference for the derivative. X sy 1 1G a (CSVTUMay 21 oe The system response of the digital filter is HQ) =H), H@)y2————___ O* Faas oie HOFoat Oy r H@= (+027 +9.0177) _y G0.) (1+0.27 +9.0177)z™ IfT=1s, H(@)= 0.0979 1=0.21552"+0,097922= 4.3 Filter Design by Impulse Invariant ; (Question 10 or Cae =a Ans. In this method of digitiz gitized an is asampled version of the impulse th P ‘¢ impulse response of resulting digit First we express the transfer function of Onse of the analog filter. analog filter in partial fraction form -e-4-5 hi HR Fi e — T can be obtained using the transformation Explain the mapping formula for impulse invariance method ‘ans. The Z-transform of an impulse response ks given by EE H@)=S he (i) HO)|, or Samer ro) eee the mapping of points from the s-plane to the z-plane implied by the a ) If we substitute s=o+ /Q and express the complex variable z in polar form as z = re” i) we get, re" e-J00 = get gsr which gives, = r=e% and = o=OF The first term in the product in equation (iii), ¢, has a magnitude of e” and an angle 0-a real number. The second term e®", has unity magnitude and an angle of OT. Therefore, our analog pole is mapped to a place in the z-plane of magnitude e”” and angle OT, The real part of the analog pole determines the radius of the z-plane pole and. the imaginary part of the analog pole dictates the angle of the digital pole. Consider any pole on the /2 axis, where o=0 as shown in figurel. These poles map tothe z-plane at a radius =e" =1- Therefore, the impulse invariant mapping map poles from the s-plane’s j2 axis to the z-plane’s unit circle. \ 4lm(2) JO Ce sphne | ~ efaT 4 z-plane Q Fig. 1: jO- axis mapping to the unit circleigital Signal Processing 4-6 | care acapemy pus Now consider the poles in the left half of ATIONS 14 lane where «<0. ‘These poles map inside the unit circle as shown in figure 2, because r =e <1 for ¢ - Im() \ ja zeplane s-plane Fig, 2: Stable poles mapping inside the unit circle Therefore, all s-plane poles with negative real parts map to z-plane poles inside the unit circle stable analog poles are mapped to stable digital poles. The impulse invariant mapping preserves the stability of the filter. Al poles in the right half of the s-plane map to digital poles outside the unit circle, r=e™>1 for o>0 The mapping is shown in figure 3. Im(z) Fig, 3: Unstable pole mapping outside the unit circle Although the jQ -axis is mapped into the unit circle, it is not one-to-one mapping rather it is many-to-one mapping, whos : single point inthe zplane. The easiest way to relent Plane are mapped toa 'y to explain this is to consider two poles in the § plane with identical real parts, but the imaginary components differing by 2x/1 shown in figure 4, Let the poles be ttt tet tet {as ca These poles map to z- plane poles 2 am - “lz, via impulse invariant mappi = (V) z z it Quest V Ke Ans. | Questsae nit ant Bea IIR Filter Design z-plane | 4: Impulse invariant pole mapping a eq. (¥) and eq. (vi) we find that these poles map to the same location in the z-plane. There are an infinite number of s-plane that map to the same location in the zplane. They must have the same real parts and imaginary parts that differ by some integer multiple of 2/7’. This is main disadvantge of impulse invariant mapping. Question 12 Why impulse invariant method is not preferred in design of IIR filter other than’ low pass filter ? Ans. In impulse invariance method, the mapping from s-plane to z-plane is many to one, i.e. Ck=De ,, Ch+DE T map into the entire z-plane. Thus there is an infinite number of poles that map to the rep location in the 7-plane, producing aliasing effect. Due to the spectrum aliasing the vod is inappropriate for designing highpass filter. That is why ferred in the design of IIR filter other than (for k=0, 1,2, .) all the poles in the s-plane between the intervals impulse invariance meth i the impulse invariant method is not pre low pass filter. Question 13 Sol, For the analog, transfer function 19), determine H(z) using impulse invariant mapping. Assume T=1se¢- Hs) %0 0, otherwiseqf ACADEMY PUBLICATIONS ™ —sampl 2 IIR Filter Design 1g the function produces “TcosnbT, forn>0 0. othenwise an) = 2 eet ala) ole vie) To find the digital filter transformation we take Z-transf -transform H(= Dhearye” = ye" cosnbT 2" Sebo we yz nein) —p Xe a ” “a eft 4 er nit =a (el COP ip orm a + aes Lewmra 1-2e” cosbT 21 +e 2" s+0.2 (+02) +9 Use impulse invariance technique. Assume T= 1 sec. Sol. The system response of the analog filter is of standard form [CSVIUMay 2013] sta H(s)=————> © (stay +b? where a=0.2 and b =3 The system response of the digital filter can be obtained using transformation al a - sta 1-e% cosbT z 2 Gaapabe > Ta2eTeosbT etree Hence, 7 1-e cosh? 27 = em eosbT eee Putting the values of a and b with T= 1 sec. we get, 1-e°02(c083)2 H()= eeerreel = TT eos tes 1+0.81052" =-—_e oe 1@)= Ty gni02" +0.67032GATE ACADEWET soebeer ee Oy 4-10 Ci in er whose system function is : Jog filter into 4 digital filter whose syster co ‘onvert the ana} 36 ae ie HO)= Geo +36 a resonant frequency of ©, = 0.22. Use impulse itvang The digital filter should have pembe elated by Ans. The analog frequency and digital frequency is y o=OT : ae vi tionship we can fi Here o, =0.2n and Q, =6, 80 using the above relationship find samp, period T. 7 202% - % =0,104sec. 6290 ‘The system response of analog filter is of standrad form S H(s)= ——— a (s+a) +5" Here a=0.1, b=6 The system response of the digital filter . be obtained using transformation b - © x Ce" sinar zt “(stay eb 1-26 cosh zt +e" 27 . BOA Putting the values of a and b, we get or CC A810 5.0/6 0.104), 1= 26? cos(6x0.104)z* +e 0.5782" H@)= 1=1.606 27 +0.97927 Question 18 Determine H(z) using impulse invariance method at 5 Hz sampling frequen "| H,(s) given by H,(s) ss abl Gaye [eSVTU Dec 2012, 2000 & Sol. Given sampling frequency =5 Hz ie. T=0.2 00 We. =U, C. Hig. ede (s+I)(8+2) sa sap 2=A(s+2)+A(s-+)) Put s=-1, 25 A(-1+2) > A=2 Put s=-2, 2=B(-241) . H,(s)=—2--2_ stl s42jane ACADEMY PUBLICATIONS 1 ft 4-4 The system function of digital filter is obtain . TIR Filter Design ne 'ed by using transformation, sp, lez {for 7 <1 sec} Hence T=0.2 sec. —o4 04 ace aa2e 4 1-0.81827 1067027 0.4[1-0.672" ~1+0 8187 ‘] os 1-0.672" 0.81827 405az2 iz H@= _ 0.05922"! 1.48827 +0.54827 Question 19 Determine H(z) using the impulse invariant technique for the analog system function 1 H(s)=—————____ oF p (s+0.5)(s? +0.58+2) a Sol. Using partial fractions H(s) can be writen as H(@)=—__1____.-_4_, Bs+C (s+0.5)(s?+0.5s+2) s+0.5 8? +0.58+2 Therefore, A(s? +0.5s +2) +(Bs +C)(s +0.5) =1 Comparing the coefficients of s*, s and the constants on either side of the above expression, we get A+B=0 0.54+0.5B+C =0 24+0.5C=1 Solving the above simultaneous equations, we get A=0.5, B=-0.5 and C=0. The system, response can be written as, 0.5 05s 05 H(s)=—~=- ay 5542 8405 5405 57 +0.5s+2 § s+0.25 | | Leepenegetetansaae | 5° 919 -05(; 54025) (1.3919 (540 25)? + (1.3919) s -05( am +L an) ao 1.3919 +(1.3919 "(sin 1.3919); 40.0898| 5, 77 (Cos! {castor 0.5| —Jo25y + 1.3919)" (s +0. nt transformation, je (cost.3919)2 * Teo 7 +05) 55 ZT (e931 39192 +e z H()=—— @) Toe 5405 Using impulse invaria O5F (cost. 39197 )zDigital Signal Proce Digital sigh — Letting T= 0.76632" -0.0894{ ———— —_| 1-0.277z | +0.606z" ) 4.4 Filter Design by Bilinear Transformation 3 Question 20 © What is bilinear transformation? ‘Ans. The bilinear transformation is a mappi unit circle in the z-plane only once, thus av The mapping from s-plane to the z-plane in bi ing that transforms the left half of s-plane intothe. oiding aliasing of frequency components ilinear transformation is Question:21 2 What is warping effect ? What is its effect on magnitude and phase response? Ma [CSVFU May 2013, 2010), ‘Ans. 1. Warping effect : The relationship between the analog and digital frequency in bilin ear transformation is given by, o=2tan® Tae For smaller values of o there exist linear relationship between and .. But fer larger values of the relationship is non-linear. This non-linearity introduc’ distortion in frequency axis. This is known as warping effect. 2. Effect of warping effect on magnitude response : The influence of warping} effecto" the magnitude response is shown in figure 1 by considering, an analog filter with? number of passband centered at regular intervals. The derived digital filter will ha\: same number of passband but the center frequency and bandwidths of high frequency passband will tend to reduce this proportionality. : © |e) \e*) V1: Effect on magnitude response due to w aan effect \a. CADE caTE A ATIONS 1% oe 3, Effect of warping effect on o-13 hi ropes a hoe gn Eons: Tc IR Filter Design luence of warping effect on pha sponse, the phase res iB an analog filter ear phase re- P Ponse of derived digital filter vwillbenen neon a | 2 o ZH(e™) Q LH(e*) 7 | | ( Fig. 2: Effect on phase response due to warping “fect Question 22 o9)> - Derive the transformation formula for the bilinear transformation. [CSVTU Dec 2010 & May 2008] Ans. The bilinear transformation is a conformal mapping that transforms the jQ axis into the unit circle in the z-plane only once, thus avoiding aliasing of frequency components. Furthermore, all points in LHP of ’s’ mapped inside the unit circle in the z-plane and all points in RHP of ‘s’ are mapped into corrosponding points outside the unit circle in the z-plane. Let us consider an analog linear filter with the system function ed sitll 6 T Sines H)= voi) sta The differential equation describing the analog filter can be obtained from eq. (i) as shown. below, i) 2-4 X(s) sta SY (s) + a¥ (s) = bX (s) ~~) Taking inverse Laplace transform, aye - (ii MO ap = bx {) Equation (iii) is integrated between the limits (nT) and nr or i a) oat yoar=b | xa sun (iv) nip dt nt ee iver Trapezoidal rule for numeric integration is given by j a()dt= Fann) sant -T)]ignal Proces: pplying Trapezoidal rule in equation (iv), we al yal) =n 1) +L ya + ST = 5 Taking Z-transform, the system function of the digital filter is H(z) td 4 Comparing eq. (i) and eq, (v) we get, ff s=2{! (vi) Ts The general characteristic of the mapping z=e” can be obtained by substituting olar form as s=0+/Q and expressing the complex variance ‘2’ in the p eq. (vi). rc +2rcos ) ) 2rsino ot jo — +2rcosio ) Therefore, 2(_ T\ +r? +2rcoso 2rsino o=2/—7e _ . Tuers+2rcso) tte (viii) 1, When r <1 then o <0 Here r <1, means interior part of circle having unit circle and o <0, mean o * negative which is L.H.S. of s-plane. So this condition indicates that L.H.S. of s-planemay* inside the units circle. 2. When r=I then o=0 Itmean jQ- axis. This condition indicates that the jQ- axis maps on the unit ci 3. When r>1 then o> 0 Here r>1, means exterior part of circle having unit cir Gs He cle and «> 0, mean ¢” positive which is R.H.S. of s-plane. So this condition indicates that R.H.S. of s-plane™? outside the units circle. = 2(_sino )- T\1+c0s0 ce An QuyACADEMY PUBLICATIONS 1 4-15 IR F sone (IX) quencies in the two domain. GAT Equation (ix) gives the relationship between the f e fre question 23 Give the properties of bilinear transformation. ‘ans. Properties : 1. The mapping for the bilinear every point, there is exact tom™ - ation is a one-to-one mapping that is for actly one corrosponding point s, and vice-versa. 2. The j©2—axis maps on to the unit circle Jz , the left half of the s-plane maps to the interior of the unit circle [z|=1 and right half of the splane maps on to the exterior of the unit circle question 24 Compare bilinear transformation method with impulse invariance. [CSVTU Dec 2009] Ans. (1) Impulse invariant filter design neglects aliasing and therefore can be seen only as an approximation of the continuous transfer function. In contrast, the bilinear transformation avoids aliasing problem but therefore there is non-linear relation between continuous and discrete frequency. (2) When designing an impulse invariant system poles and zeros are not being mapping, the same way. Poles are corresponding with 2, T and zeros are dependent on T,A, . Whereas, when using bilinear transformation, the whole s-plane is mapped to the z-plane, there is no difference between zeros and poles. Question 25 What are the advantages and disadvantages of bilinear transformations? Ans. Advantages : 1. The bilinear transformation provides one-to-one mapping. 2. Stable continuous systems can be mapped into realized, stable digital systems. 3. There is no aliasing. Disadvantages : 1. The mapping is highly non-linear producing frequency compression at high frequencies. 2. Neither the impulse response nor the phase response of the analog filter is preserved ina digital filter obtained by bilinear transformation. Question 26 ie A digital filter has following frequency specification : Passband Frequency =®, = 9.2%, Stopband Frequency =«, = What are the corresponding specifications for passband and stopband frequencies in analog domain if, : 1. Impulse invariance technique is used 2. Bilinear transformation is used for designing. Assume sampling time = 1 sec. [CSVTU May 2012 & Dec 2010} 3%Digita Sol. Given data, ©, Impulse invari We know that, © = Hence ance method ar We know that, 2= = 1a Hence, Ang Question 27 Convert the analog filter with system function s+0.1 ¢ a Qa HO)= oad into a digital filter using bilinear transformation. The digital filter should have resonant frequency of o, =x/4. Sol. From the system function, we note that Q, =3. The sampling period T can determined by, U 9, =2 tan ~ a) a 2 ar 2 2 \ T= suns Flan E = 0.276 see 7 ) ‘ 4“ Using bilinear transformation, »* “? a H(2)=H(s)| n >« ACADEMY PUBLICA gat LATION IIR Filter Design 2 z PEDED ONE 4 Jo 140.0272! ~ 0.97322 genie 8572-11842" W842 "4817727 ie H(2)= peo) FON 241)? substituting T= 0.276 sec H(2)= question 28 Apply bilinear transformation to #1(s)=——?_ with T= 0.1 sec. (s+1)(s+3) i [CSVTU Dec 2010] gol. Given that, H(s)=——2__ (s+1)(s+3) For bilinear transformation, H(z) =H(s)|, 2721 (2) =H(s)|,_: (=) 2 “Ty (zl =| [>] 202 e+)? 0.00410 +2)" A _ 2+ es 7 ‘Ans. as i9yfase 1] Ths *+0.6682" ‘Que: stion 29 1 bandwidth of 0.25 is tobe designed from analog filter whose! A digital filter with 3 dl system response is BE i115)= 542, Use bilinear transformation a or Sol. Using the relationship between 2" ; aris given by, The system response of the cigital filter s given by (2) = HO,2(64)4-18 GATE ACADEMY PUBLICATIONg y, oh Digital Signal Processing os Qu 0,828 H(z) 0.828(2 +1) +z 7 828241) 3414-14142" Ans ‘Question 30 - Design a digital low pass filter to appropriate the following T.F. i [CSVTUMay 2009) +2541 Using Bilinear Z-transform (BZT) obtain transfer function H(z) of digital filter, Assuming 3 dB cut off frequency of 150 Hz and sampling frequency of 1.28 kHz. Sol. The analog prototype filter transfer function is given by, where(o, = 300¢/ and 7 =—1_ 128k H(@)=H,(s) [3 (3007)? J + F002 Pl iez (300m)? (+27)? +300) “4 2, == prey + 2x 300m (l-27)+300m)*(+2"4)? 7 0.88(1 +227 +27)x10° 6.55x10°0— 22" +27) +3.412x10° 27) 40.88x10ae 08814221 +27) © 6.55-13.102'+6.552* +3.412—3.41227 +0881 762! +0.8827 = 08804221427) _ 0.081+0.1622" 40,0812 Ans: 4.018227 -11.342'+10.842 = 1.0452" 4037122BUY ogre ACME LEMICATIONS N ies 4. a 4 19 gastion® IIR Filter Design se the bil ——— inear transformati tion to co 04 nvert the analog filter with the system function. IP +9 H(s) = (s into a digital IIR filter. Select T=9.4 [CSVTU May 2009] of zeros rer cor tion obatain mpare the location of the zeros in H(z) with| ed by ap @ yi ; Plying the impulse invariance method in the} sau Using Bilinear Transformation : H(z) = HOS), ,2f Fi H(2)= ; {1 =.1 see} = 0.10421)? food== 5} 90d) 20-2027 + 0.142272) 4OOU —227 +27) +9.01(+ 22" +27) 44-42" f -1 19.9927 \ ance) = 20d n02F = 199 ©)=i501- 781.982" +405.012" Ans. 1 Zeros of H(z) are z, =-0.99*-1 2 Using Impulse Invariant: Hiy=— 20h, a=04,b=3 (+0. +9 1-622" e0s(0.3)2" oar yl tee z' cosbT —. . =a oor 1 H(@)= seat cosble ee [26 cos(0.3)z" +e 1-0,9458z" : 1=1,891627 +0.98022 zeros of H(z) isat z= 0.9458 Oncor i ere transformation are in unit circle and zeros of H(z) using ar tr Zeri ing, biline’ Os of H(z) using bilinear WT cle impulse invariance is insidetal Signal Processing 4-20 GATE ACADEMY PUBLICATIONS 1, Pr. 4.5 Butterworth Filter: Question 32 . Given the magnitude function of Butterworth filt Non magnitude and phase response? Ans. The magnitude function of Butterworth filter is given by er. What i theeffect of varying orderay where Nis the order of the filter and Q, is cut-off frequency: The magnitude response of the Butterworth filter closely approximates the ideal response as the order N increases. The phase becomes more non-linear as N increases, Question 33 Give the properties of Butterworth low pass filter. Ans. The properties of Butterworth low pass filter are : 1. The magnitude response of the Butterworth filter decreases monotonically as the frequency Q increases from 0 to «©. 2. The magnitude response of the Butterworth filter closely approximates the ideal response as the order N increases. 3. The poles of the Butterworth filter lie on a circle. Question 34 Se Give the equation for the order of N and cut-off frequency 9, of Butterworth filter. Ans. The order of the filter where &, = Stopband attenuation in dB at stopband frequency ©, @, = Passband attenuation in dB at passband frequency f ° Cut-off frequency ©, = —® lo" ye a (2 Order of the filter is also given by, N > 7 log 2.) Q, \a >a en TT| =f AE NATIONS ™ 4 Oe 224 is i whe! —____IIR Filter Design 1 1 esti oe " (qu! ive the expression for location of Poles of norm, iz sof the Billeracs alized Butterworth filter. i The poles 0 tutterworth filter is piven by, _— Sect Hitek ea ty m Qk-1 where be 4 Gk-Na ina Nv and N is the order of the filter, (question 36 Design transfer funetion for: Mand 2" order of Butterworth filter. [CSVTU May 2011] gol. Poles of the Butterworth filter is given by sce $ -34(4)s where, %& =5*|- For 1* order N =1 m (2-1 =—+|——|n=n w-E(iler 5 Hence the filter transfer function for 1* order Butterworth filter is given by, attest HO= yal For 2 order filter N =2 s =e, where ,=34(—y 4ACADEMY PUBLIC ? [a ATION, iven PY, Any 1 ed ww pass Butterworth filter that has a3» (Question 37 Determine attenuation at 500 Hz ind the poles of lor the order and the Po in of do dB at 1000 Fe (CSVTUMay and an attenual Sol. Given data a, =34B, Q, = 2xnx500 =1000 rad/see P , Pp x rad/sec o, = 40 dB, 2, =2 701000 = 2000 The order of the filter 0 wa og | si log Jas V\i0™ Yio" wl 5 ON 10" =1 6.6 vee i> lo lo. (2) : 8 000% ” Rounding N to the nearest higher value we get N=7 ‘The poles of the Butterworth filter are given by, 5, = 2.0" =1000Ke* =I, 2.7 where as Question 38 Prove that ©, Go ay ae ye Ans. The magnitude squa i 8 391 jog| 2 Q, Approximating the nearest higher value, we have N=4. Butterworth Transfer function : : i Now write the 4® order transfer function of normalized Butterworth filter Fo HO) = ayo 765354 Ie +184778 +0) Now to obtain (2) using impulse invariant method H(s) should be expanded in partial fraction form, As+B Cs+D H()=3=—— tees F4076S38+1 $° 418477541 1=(As+ BY(s? +1.8477s +1) +(Cs + D\(s? +0.7653s +1) ‘Comparing the coefficients we get, A+C=0 1.84774 +B +0.7653C +D =0 A+18471B+C +0.7653D =0 Ac? B+D=1 Solving these equations we get, A=-0,9238, B=—0.707, C =0.9238, f 9238, D= Now we have, ‘ae Hg) = 0523884 (0.707) 8 +0.76535 +1 Let H(s)=H,(s)+H,(s) 8 4+18477541 Rearranging, //(s) into the standrad form, 1H (s) = 22385-0707. 8? +0.7653541 (wv) osase| 240.2659 8° +0.7653541rf if 4-25 cy — y Now. 1,69) =11@)],., + (low pass t oa 0 low pass transformation) prom equation (wv), 2s - 0.92381 +0.107) H0)= 7 ~ (0.58815 40,707) (2) 40765552 4 0.40525? +0.48725-41 = OSSBUS41.202) ___=1451(5+4.202) 0.4052(s? +1.2025-3.4679) 2(+0.601)° + (145)? H(o) = 21451 +0.60) 1451 ° (s+0.601" +0451" (s+0.601)? +(1.451)* 1 Similarly, Rearranging H,(s) into standard form 0.92385 +1.707 HAS) = Fears 5? +1.8477s-41 0.9238n25 41.707 1.451(s+2.902) a(S) = z ~ (er 145ly +(0.602) (s +1451) + (0.602) (2) 4184772241 7 © 1.451 Lasys+14sy_,__@.450_ H, ? , = Fay ase Om | HAD’ +0602) 3.4973 0.602 soe (Vii) (5 +1451 oom LAS (s+ Dt LAs + (0.600) H(s) = (8) (s+1.451)* + (0.602) Determination of H(2) : Transforming eq.(vi) into digit 1-2" cos(l Ase H(z)= (1.451) = eH eos 4502 The" i ilter TF. using impulse invariant method we get al fi ost 45 D3 = 0.601 ma go ASD | +e" ..(witi) -1.451-0. 23212 “T0317 0.30062"4-26 GATE ACADEMY PUBLICATION ® Digital Signal Processing Transforming equation (vii) in th 1-e7! * cos(0.602 H,(z)=1451 e' sin(0.602)z | Je (c0s0.602)z | +e™ 13.4973; 1.451+0.18482 sone(ix) ~ 1=0,38622' +0.055z Combining equation (viii) and equation (ix) we get, H(z) = H,(2)+H,(z) ie. 1451-0.2321z" | _1451+0.1848; a 1-0.131027 +0,300627 1- 0.38622" +0.055z* a po Question 40 Design a Butterworth filter using the impulse invariance method for the following specifications H(z)= 08s|He\<1 0<@<0.20 \He”™)| $0.2 06x e¢=0.75 and But the relationship between analog and digital edge frequency ®,=OA7 and o, Let T=1 sec. Q, is given by, ,=06r and O,=022 Determination of the order of the filter : Ne lela) 2171 log(Q,/2,) Approximating to the nearest higher value, N= 2 Determination of -34B cut-off frequency;— ie UR Filter Design Transforming Normalized low pass fio, H=HO)), “tt low pass filter, _ 0.5266 trata oat s* +1.035+0.5266 =—O516j 0.516; S+0514/051 SF051—j081 5, _ 0516; 0.516; = 16j SI-J0.5) s-0514j03n Determination of H(z) ; Obtain H(z) by transforming 77 ,(s) using impulse invariance method we get, 0.516; 0.516; 0.30192" H(z) ah 5 — Ost gs05H os 1.0482"! +0.3627 sie Question 44 Design a digital Butterworth filter that satisfies the following constraint using Bilinear transformation. Assume T= 1 sec 09s|H(e™)|<1 ene ee |e|<02 3n/45o50 [CSVTU May 2013 & Dec 2011, 2008] Sol. Given that: §, =0.9, 6. We know that, = 6=0483 = 1 = 4.8989 o cies are related by, 'n case of bilinear transformation anal08 and digital frequen 4 ¢ of bilinear tr Prad/sec 2 9, =2 an 2 TS 4,828 rad/see stan. SyGATE ACADEMY PUBLICATIgy, ' \ 8 4A 4-2! cA al Signal Processing : Determination of order of the filte _—_tog(2/2)_5 9.695 Toe (2, 72, 0 10,) N=3 Jue we have, mating nearest higher 9 Seeeaten ‘of -3dB cut-off frequency * 2 2.5467 rad/see 04843)" Rea Butterworth filtertransferfunction: ss sunction is given by, For 3" order normalized Butterworth filter tr 1 HO" ered ‘Transforming normalized low pass filter to low pass, 1 (2.5467) “Ts ga) (6425467) (07 + 2.54675 +6.4857)< 2.5467 Q2. 5467)" 2.5467 Determination of H(): H(@)=H,(s)|,_2/ . He) = ————____ 2 or ea PES asi a a +2.54672 H@= 0.23320 +21) we 1+0.43942"' +0. ue +0.04162> Question 42 Design and realize a digital LP requirements: ys Using bilinear transformation to st the fll (a) monotonic stopband and passband (&)-34B cutoff frequency at 0.6m radians, and (© magnitude down at 1648 at'0.25, radians, - [esvTUDec 200. Sol. Given that, «, =34B, a, =164B, @ — =0. AssumeT=1sec. + 15R, ©, = 06x Using frequency relationship, Q, = Ftan Se ve = tan OTe 4.828 rad/seo\CADEMY PUBLICATIONS ™% we 4-29 IIR Filter Design 2.7527 rad/see petermination of order of the filter: imating to nearest high, approximating igher value we h; = Determination of —3 dB cut-off frequency : aaa Q, 2.7527 Q,= Gia, 1, Sara 2.7527 rad/sec Butterworth transfer function : for4* order Butterworth filter transfer function is given by, 1 © (CF +0.7653s +)(F +L. 847154) Transforming normalized low pass filter to low pass, H)=HO) H(s) ay ore 2] peter] S41] a1 87x 2.7527)° 401653457597 = 2.7527 (2.7527) : (s? +2.1066s +7. 577)(s? + 5.0865 +7.571) Determination of H(z) : A@= Ha()),.2(27) Hay= z 2 =) {Fl es} call Ti+z ast & 1042) er A(z) = 0.167 os 18262 0.03012 140.7822 | +0.67992 1 drawn. ructul °° Realization direct form Ii struct re can BeGATE ACADEME. + vemivariong CS i s! 4-30 Digital Signal Processing 8 ion 43 a oar transformation to meet # oa orth filter using Bilinear transtorn he const Design a digital Butterw oss|H(e*)|s1 0S <0. [CSVTUDec 2m ay |ne)|s02 0.6nS0S7 Sol. Given that, 8, =0.8 5, =0.2, ©, =0.2% We know that, = 2 =4,8989 => €=0.75 viee® But relationship between analog ai tion is given by, T=1 sec. nd digital frequencies in case of bilinear transfom a, =2tan®* = 2 tan 22% = 0.6498 rad/sec eT 2 0, = 2 tan = 2tan 2% = 2.7527 radisee Te 2 2 Determination of order of the filter : log@./e)___log(4.8989/0.75) “Yog(Q,/2,) log(2.7527/0.6498) Approximating N to the nearest higher value, we have N = 2 Determination of -3 dB cut-off frequency : Q, _ 0.6498 "O75 Butterworth low pass filter transfer function : For 2 order Butterworth LPF transfer function is given by, 1 =0.75 rad/see H(s)= s4V2s41 Transforming normalized low pass filter to low pa SS, H(s)= Hs), P 4g a8 46 Ry 0.5625 * +1061 15 40.5625 8 (0.75) a 1 0.75* Determination of H(z) :eee IIR Filter Di ign : 0.08410 +274) eee 1-1.028427 0363157 Ans (question 44 A digital low pass IIR filter is bilinear transformation techni to be designed with Butterworth approximation using} 1, Pass band magnitude is co “ue. Find order the filter to meet following specifications nstant within 1 dB for frequency below 0.2m. 2 Stop band magnitude is greater than 15 dB for frequency below 0.32 to x. [CSVTU Dec 2009] | Sol Given that, 4, =14B, a, =154B, 0, =035, @, =0.2n | Assume T= 1 sec. Using bilinear transformation, 0.31 =2 tan—* = 1.019 rad/sec tan 22% = 2ta 2 nea Determination of order of the filter: = 0.6498 rad/see 10° 1 | 10°" log V2—1 = log(Q, /Q,) eee 10m Tog(1.019/ 0.6498) Ane. 2531 N= Approximating to nearest higher value we have - Hence order of the filter is 6. 46 Chebyshev Filter : \Questio, mn 45 i i )perties. Tap Gite the Chebyshev filter transfer function and“ properties, The transfer function of Chebyshev filter is g Ye | GQ)Signal Proc! 4-32 GATE ACADEMY PUBLICATIONS ~, igo] re yt parameter of the filter related to the ripple in the pas \* order Chebyshev pol band mial and is defined as Cy(x) = c0s(N cos), xis Cy (x) = cosh(N cosh’ ¥), Jxj>1 Pi ties : 2 1 The magnitude response of the Chebyshev filter exhibits ripple either in passband o, in stopband according to type. : 2. The poles of the Chebyshev filter lie on ellipse. Question 46 E Give the expression for location of poles of a Chebyshev filter. Ans. The poles of the Chebyshev filter can be found by using the formula aN where, | = minor axis of ellipse |- major axis of ellipse b=Q, |e tH 7 2 Q, = passband frequency use, gai c, = passband attenuation N = order of the filter. Question 47 Distinguish between Butterworth and Chebyshev filter. [CSVTUMay 2003] Ans. 1. The magnitude response of Butterworth filter decreases monotonically as the fe quency Q increases from 0 to e, whereas the magnitude response of the Chebyshe" filter exhibits ripple in the pass band and monotonically decreasing in the stopban4 . The transition band is more in Butterworth filter compared to Chebyshev filte: 3. The poles of the Butterworth filter lie on a circle whereas the poles of the Chebysh filter lie on an ellipse. 4. For the same specification the number of poles in Butterworth filter are more wh" compared to Chebyshev filter i.e. the order of the Chebyshev filter is less than that! xv Butterworth. Question 48 Write true or false. [csvruMay 2!) 1. Poles of Butterworth filter lie on an ellipse. False aN| ie ee a art oi Chebyshev fiterliews wt 4-23 : * true Man ellipse, IIR Filter Design 1, Chebyshev filter poles are close , 3. fo JQ-, = axis than those in But tt : 4, Acausal and stable IIR filter can not “_ nes have linear phase, son 49 Given the specification a, =3 4B, «, = 16 ap, S,=1 | order of the filter using Chebyshey approxi a mation. Find H(s). [CSV nt c ). _ [CSVTU May 2009] 2,= 2xx 1000 = 2000x rad/sec Q, = 2nx 2000 = 4000x rad/sec a, =36B, a, =16dB Determination of order of the filter: 10% —] cosh"! (os = 10' 1 40007 20007 cosh”! 10° N2 Sere er eee eet cosh" (2, /Q,) cosh” Rounding N to the next higher value we get N=2. Determination of minor axis and major axis: = (10% —1)° = 0% 1)" =1 usela vite? =2414 ; 2-414)?” ] (yw [2419 -CM9T o1on a=Q, = 2000 7 [a ee [es +@4udy"?] 22197" : [(eaOnea ey b=a, oe 2000r- 2 Poles of the Chebyshev filter: ‘= acos$, + jbsind, kL? 6, =% 4 Qk-Im ko12 2° oN 42% 2205" rer eae? 6 = 643.46n+ J1554" , 55 acosd, + jbsi + jbsin 0 2643.46 f1554% 2 he : 2 =acos$, + jbsinDigital re 7 4-34 GATE CADEMY PUBLICATIONg ‘igital Signal Processi Finding denominator of H(s) Denominator = (s +643.467) +(1554R)" Finding numerator of H(s) : For N even substitute s = 0 in t Vive . This value is equal to the numerator. (643.465)? +5549)" _ (4414.38)? he denominator polynominal and divide the resuyy, Numerator = hee Hence, a? Ho= (1414.38) ms (5 +643.46n)" +(1554x)" Question 50 : ; Determine the order and the poles of low pass Chebyshev filter that has a1 dB ripplein the passband and passband frequency ©, =1000x,astopband frequency of 2000x and an attenuation of 40 dB or more. [CSVTU May 2010] Sol. Given that, a, =1dB, 2, =1000z rad/sec a, = 40 dB, Q, = 2000n rad/sec Determination of order of the filter: 2——_e cosh” ( 000 10001: Rounding N to the next higher value, we getN=5 Determination of minor axis and major axis : = V0" -1=0,508) w=e" + Vise? = 417 ww eu on, gs Sm [-we"] b=0,>—>—== z 1041n Determination of poles : a 2%, Qk=-Un & 2N F105 , = 180°, $) =144°, $, = 180", 4, = 2169 $ 's = 252° sol.DEMY PUBLICATIONS 14 cosd, + jbsing, -89.5+ /989n, 2897 234.2n- j612n, s, = 89.Sr— jo89q a Ans, ne the system function 7 peter Pope) 4 nN A(z) of the | mane specication using btincar ce order Chebyshev filter with fo A seat 3a ripple in passband 0<@<0.2_ " 2 dB attenuation in stopband 0.452<@ < a © given that, , =34B, &, =254B, 0, =0.25, @, = : . 2, @ 0.2 st =2tan—— =0.65 9, => 3 = 0.65 rad/sec 0.452 2 @, =< tan— = 2tan 1.71 rad Q, T 2 rad/sec Determination of the order of the filter: ‘pproximating to the nearest higher value, we have N=3 Determination of -3 dB cut-off frequency: 2+ _ 0,65 rad/sec (o""** -1) ; Determination of minor axis and major 215 Vo? = =1 | bsetavi-e? =2.414 ma] 5 (2.414)? | 0.678 UN gl aay? +2: | beg oa a" 241") -0 193s | oa Pay ofthe filter: 21,2, Cos, + jbsind, k 2, Qk-Ne 4 21,2,3 2 IN _ alll-36 Digital Signal Processing 4 4, =120°, 4; =180° 6, = 240° GATE ACADEMY ee lly oe ee NN s, = acosd, + jbsind, 7 675 + j0.587 =0.1935c0s120° + j0. 678sin 120° = 0.09675 + J 2 s, = acoso, + josing, ae ¢ = 0.1935 =0.1935c08180° + j0.678sin 180° = 0.1935 of s, = acos®, + jbsind, oO = ~0,09675— j0.587 Finding denominator of H(s) : Denominator = (s+0.1935) (s? +0.1935s +0.354) Finding numerator of H(s) : For N odd substitute s = 0 in the denominator polynominal and find value. The valu equal to the numerator of the transfer function. Numerator = (0.1935) (0.354) = 0.0685 Hae i) (8 +0.1935)(s? +0.1935s + 0.354) Determination of the system function H(z) : H(2)=H(s)|, , H@= 0.006871 +2"')* (10.8232) (1-1.627 40.9152) f Question 52 a Design a Chebyshev filler forthe flowing - wee ey we ) |-03s07 U8 gel al ew 6 ee 0.7255 roles of the Chebyshev filter: cosd, + jbsing, (Qk-I)n 2N k=12. 4, = 225° 0.2564 + 70.513 0.2564 ~ j0.513 Finding denominator of H(s): Denominator = (s +0.2564 = j0.513)(s +0.2564 + # 513) = (s +0.2564)' + (0.513)° s* + eo) + YPROWYIS + Finding numerator of H(s) : @s (0.2564) +0513)" _ 9.263 Numerator To = 0.2631 Chebyshev filter transfer function: 0.2631 ; (40.2564) + (0.513) o convert H(s) int H(s)= ] tandrad form Toobtain (2), we have t ips 0. 5128) (0 Hs) = (540.2564) +0 (0513) impulse invariant method, Transforming, H(s) into H(z) usin a2setF gin(0 5131) fe P=twe} 05128 gos(0. 513102 N@) 0.19542! H(z) =~ = eT 134832 | +0. 59872GATE AC! -38 Digital Signal Processing 4-3 Question 53 n _ Design a Chebyshev lowpass filter with the specification %, \4B tipple in W esi , ios iB! =154B ripple in the stopband 0.3250 ST. Using inp {CSVTUMay 99 passband 050 50.2% invariant method. (Assume T=1sec:) a, =14B, a, =154B, @, =0.3% Op lationship, =0.2n T=1sec. Sol. Given data, Using frequency rel 3x rad/sec ©» orn radisee 9,= 7 T T Value of N: ata, cosh” tom Nz = =3.2 cosh (2, /2,) Rounding the value of N toa next higher value, we getN=4 Axis of ellipse: io™ =1 = JLo -1 = 0.508 4 Vite? 24.17 0-0, ag ee | pas wm gym peg ee | cog] GUD aay | 2 | ery |=067 H Poles of the filter : k=1,2,3,4 = 112.5%, 6, =157.5°, 6, = 4, =157.5°, $, = 202.5%, 6, = 247.59 Ne 5, = @c0s6, + jbsing, = -0.0876+ /0.619, 0.2115 + 70.2564 ote ) : 5 =-02115~ j0.2564, 5, = ~0.0876~ j0,6 Denominator of the H(s) : LE Denominator = (s? +0.1755 +0, 391) (5? Numerator of the H(s) : = — 39D O11 Numerator === merator =F = 003834NS IM ev filter transfer fancy ‘ e a 0.03834 HO ST 0st0 391) (5? $0. 423540.1) A -—__ ‘ At s (0.0876 + 70.619) S~0.0876—jo.619) ; B s~ (021154 j0D5Gy s 2115 + 70.2568) © sacg5 s-(@nis— salving for A, A*, B, BY, we got (92115-02565) A=-0.0413+ 0.0814, B=0,0413- jo.2166 Using impulse invariant method, obtain 11(2) j ot & Fat S— Py gat le Hence, (2) = ~0.083-0.02452"" 0,083 +0.023827 7 Ans. 11.4927 +0.83927 "1-1.562" 40,6552" (question 54 pee Design a digital Chebyshev filter to satisfy the constraints using Bilinear transforma-' tion and T = 1 sec. {CSVTU Dec 2011 & May 2012, 2011] ead s|ie|s1_ 0<0<02R opi! Loe ene Sol. Given that, 8, =0.707, 5, =0.1, ©, =0.2n, 0, =05e We know that, Lig, = 2= 9.9498 Vd+2?) 1 0.707 = 8=! Vite" i oy the bilinear transformation : we intend to employ P A Prewarped frequency values : SiMe) . The prewarped frequencies are given by method, we must prewarp the 0.2% _ 9.6498 rad/see Z 2 © = 0,6498 ra 0, = 2 tan 22 = 2 tan re? 2 0 22 tan tan OSE = 2 radsee Q, = 2 tan 2+ = “T filter: Determination of order of the 7 Np cosh G18) — 31.669 cosh (2, /2,) Rounding N to the neare higher value, have N=2 i fi cians.Digital s Value of ellipse , = acosd, + jbsing, -0.209- 0.5048 5, = -0.209 + 0.5048, Denominator of H(s) Denominator =[s~(-0.209+ j0.5048)][s ~(-0.209) - 70 504] = (5 + 0.209)? +0.2548 Numerator of H(s) : Numerator = Chebyshev filter transfer function 0.211 H(sy= 220 __ = F0.209? 0.2548 To obtain H(z) using bilinear transformation, 4.7 Elliptic Approximation Filter Question 55 Describe elliptic filters. Ans. The elliptic filters is sometimes calk led a and stopband. Among the filter types @ oUt filter. Ths i 4 pat or ter has equiripple P®* xd deviations, elliptic filters have the minima or fltet Order, passband and 30F ii response of an odd ordered elliptic filter et TaNsition bandwidth. The MB" ye! i fi ilter is shown in figure. The magnitude 54\CADEMY_ PUBLICATIONS 1 {ie ponse iS given by Yosh __HR Filter Design | wa = lon | Te'U ray igo Up) iS the Jacobian ellipt; were Ue) AN elliptic functioy band ripple. Nof order Nana y ua A constant related to the Fig. Magnitude response of low. pass elliptic filter 43 Frequency Transformation : fastion 56 What is meant by 3-dB cut-off frequency of LPF 2 E u 2 Why is frequency transformation 2 needed ? we os the different types of frequency transformation ? Explain digital frequency transformation for LP, HP, BP and BS filters. [CSVTU May 2011] ie. 3B cut-off frequency : The edge passband ; also known as the corner frequency of 3 dB point since it is the ‘corner’ in the asymptotic Bode plot and is generally 3 dB lower than the peak passband symp! 8 ly. peak p. gain. Need of frequency transformation : Low pass filter can be considered as a prototype filter and its system function can be cbiained. Then, if a high pass or bandpass or bandstop filter is to be designed, it can be easily obtained by using frequency transformation. The different type of frequency transformations are: 1. Analog frequency transformation 1. Digital frequency transformation Digital Frequency Transformation : ve liacily te arta The frequency romsformation is done in the digital domain by r placing, the varial (2). This mapping must take into init circle must map onto itself and the unit 0 map onto itself, the implication is bya function of #7, Le. fl account the stability criterion. All the poles lying within the aa eset circle must also map onto itself- For the u! that for r= 4 ju) =|sere ) em eee te pass | forall frequencies. So the mapping is that of an all pass =1 fora Hence we must have |/(¢ Miller & of the form Se") = 2 i Ai-az42 GATE ACADEMY FUBLICATIO\¢ y e filter, we must To get suitable stable filter from the stable prototype filter st have . od for converting, the prototype low be obtaines pass or bandstop filter. transformation formula can : filter into a digital low pass, high pass Low pass to low pass : PAS dp, 1,274 l-az where sin[@, -o,)/2] ase sin[(@, +0,)/2] = passband frequency of low pass filter ©', = passband frequency of new low pass filter Low pass to high pass : no>|— ltaz where _cos[(@, +0,)/2] cos (@, -0,)/2] ©, = passband frequency of low pass filter ©, = passband frequency of high pass filter Low pass to bandpass : 2ak yk -1 OL Trek? fie PORT 2a 73 --, kt k+l +1 where eos[(, -0,)/2] k =e! ©, = upper cut-off frequency ©, = lower cut-off frequeney ty = passband frqeuency of low pass filt ass filterTIR Filter Design ee ge tan[(o, -0,)/2]tan Oe 2 0, = upper cut-off frqeuency 0, = lower cut-off frequency 0, = passband frequency of low pass filter Question 57 design a digital filter An analog filter has a transfer function. H@=— tes s+7st10 [CSVTU Dec 2013] equivalent to this by using impulse invariant method. 10 Sa, Given, H(s) For Impulse Invariant method, x TC, H@)= Dicer 3.33 285 ]=04|5 Wehave H()=T ete ee 666 0.661 »2o| -| aon 10.672 0201s Ans. = 7-1.03782 40.2472 | goo dit.