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24 Nov'09Comp30291 DMP Section 51 University of Manchester School of Computer Science Comp 30291 : Digital Media Processing Section 5 z-transforms & IIR-type digital filters

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24 Nov'09Comp30291 DMP Section 52 Order is maximum of N and M. Recursive if any b j is non-zero. A 2nd order recursive filter has the difference-equation: y[n] = a 0 x[n] + a 1 x[n-1] + a 2 x[n-2] - b 1 y[n-1] - b 2 y[n-2] Introduction General causal digital filter has difference equation:

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24 Nov'09Comp30291 DMP Section 53 y[n] = a 0 x[n] + a 1 x[n-1] + a 2 x[n-2] - b 1 y[n-1] - b 2 y[n-2] Signal-flow-graph for 2 nd order recursive diff equn z -1 + + + + a0a0 a1a1 a2a2 x[n] -b 2 -b 1 y[n]

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24 Nov'09Comp30291 DMP Section 54 Derivation of ‘z-transform’ If {x[n]} with x[n] = z n is applied to an LTI system with impulse- response {h[n]}, output would be, by convolution :

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24 Nov'09Comp30291 DMP Section 55 ‘z-transform’ of {h[n]} If input is {z n }, output is {H(z).z n } z may be any real or complex number for which series converges. For causal & stable IIR, the series converges when |z| 1. Replacing z by e j gives frequency-response:

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24 Nov'09Comp30291 DMP Section 56 Visualising H(z) On Argand diagram (‘z-plane’), z = e j lies on unit circle. Imaginary part of z Real part of z 1 z = e j

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24 Nov'09Comp30291 DMP Section 57 Example 5.1 Find H(z) for the non-recursive difference equation: y[n] = x[n] + x[n-1] Solution: {h[n]} = {..., 0, 1, 1, 0,... }, therefore H(z) = 1 + z - 1

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24 Nov'09Comp30291 DMP Section 58 Example 5.2 Find H(z) for the recursive difference equation: y[n] = a 0 x[n] + a 1 x[n-1] - b 1 y[n-1] Solution: If x[n] = z n then y[n] = H(z) z n, y[n-1] = H(z) z n - 1 Substitute to obtain: H(z) z n = a 0 z n + a 1 z n - 1 - b 1 H(z) z n - 1 H(z) = a 0 + a 1 z - 1 - b 1 H(z) z - 1 (1 + b 1 z -1 ) H(z) = a 0 + a 1 z - 1 When z = -b 1, H(z) =

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24 Nov'09Comp30291 DMP Section 59 Find H(z) for the 2 nd order difference-equation y[n] = a 0 x[n] + a 1 x[n-1] + a 2 x[n-2] - b 1 y[n-1] - b 2 y[n-2] The same method gives: H(z) z n = a 0 z n + a 1 z n - 1 +a 2 z n - 2 - b 1 H(z) z n - 1 - b 2 H(z)z n-2 a 0 + a 1 z - 1 + a 2 z - 2 H(z) = b 0 + b 1 z - 1 + b 2 z - 2 with b 0 = 1.

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24 Nov'09Comp30291 DMP Section 510 Consider difference-equation for general digital filter N M y[n] = a i x[n i] b j y[n j] i=0 j=1 The same method gives: This is the ‘system function’ of the digital filter. Also referred to as ‘transfer function’

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24 Nov'09Comp30291 DMP Section 511 Ratio of 2 polynomials in z -1 Equal to z-transform of impulse-response when this converges. Easily derived from difference-equation & signal-flow graph. Replacing z by e j gives frequency-response System function

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24 Nov'09Comp30291 DMP Section 512 Example 5.3 Give a signal-flow graph for the system function: Solution: The difference-equation is: y[n] = a 0 x[n] + a 1 x[n-1] + a 2 x[n-2] - b 1 y[n-1] - b 2 y[n-2] Represented by the signal-flow graph below: 2nd order or ‘bi-quadratic’ IIR section in ‘direct form 1’. z -1 + + + + a0a0 a1a1 a2a2 x[n] -b 2 -b 1 y[n]

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24 Nov'09Comp30291 DMP Section 513 Label ‘z -1 ’ box inputs & outputs as shown: To implement ‘direct form 1’ biquad as a program z -1 + + + + a0a0 a1a1 a2a2 X -b 2 -b 1 Y X1 X2 Y1 Y2

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24 Nov'09Comp30291 DMP Section 514 X1=0; X2=0; Y1=0; Y2=0; while 1 X = input(‘X=’); Y = a0*X + a1*X1 + a2*X2 - b1*Y1 - b2*Y2; disp(sprintf(‘Y = %f’, Y) ) ; % output Y X2 = X1; X1 = X ; Y2 = Y1; Y1 = Y ; end; In MATLAB using floating point arithmetic

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24 Nov'09Comp30291 DMP Section 515 y = filter([a0 a1 a2], [b0 b1 b2], x ); In MATLAB using Signal Processing toolbox :

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24 Nov'09Comp30291 DMP Section 516 K=1024; A0=round(a0*K); A1=round(a1*K); A2=round(a2*K); B1=round(b1*K); B2=round(b2*K); X1=0; X2=0; Y1=0; Y2=0; while 1 X = input(‘X=’) ; Y = A0*X + A1*X1 + A2*X2 - A1*Y1 - A2*Y2 ; Y = round(Y/K); % Divide by arith right shift disp(sprint(‘Y=%f’,Y)); % Output Y X2 = X1; X1 = X ; % Prepare for next time Y2 = Y1; Y1 = Y ; end; In MATLAB using fixed point arith & shifting

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24 Nov'09Comp30291 DMP Section 517 Look again at ‘Direct Form 1’ signal-flow-graph It may be thought of as two signal-flow-graphs: z -1 + + + + a0a0 a1a1 a2a2 x[n] -b 2 -b 1 y[n] Non-recursive part Recursive part x[n] y[n]

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24 Nov'09Comp30291 DMP Section 518 Re-ordering LTI systems It may be shown than if we have two LTI systems as shown: L1L1 L2L2 x[n]y[n] L2L2 L1L1 x[n]y[n] then re-ordering L 1 & L 2 does not change the behaviour of the overall system. Only guaranteed to work for LTI systems

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24 Nov'09Comp30291 DMP Section 519 Alternative signal-flow-graph Look again at ‘Direct Form 1’: Re-order the two ‘halves’ & then simplify to ‘direct form 2’: x[n] z -1 + + + + a0a0 a1a1 a2a2 -b 2 -b 1 y[n] a2a2 x[n] z -1 + + a0a0 a1a1 -b 2 z -1 + + -b 1 y[n] a2a2 z -1 + + a0a0 a1a1 y[n] x[n] -b 2 + + -b 1

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24 Nov'09Comp30291 DMP Section 520 ‘Direct Form II’ signal-flow-graph Its difference equation is: y[n] = a 0 x[n] + a 1 x[n-1] + a 2 x[n-2] - b 1 y[n-1] - b 2 y[n-2] i.e. exactly the same as Direct Form 1 It is a 2nd order (bi-quad) section whose system function is: a2a2 z -1 + + a0a0 a1a1 y[n] x[n] -b 2 + + -b 1 W W1W1 W2W2 Direct form II economises on ‘delay boxes’. Notice labels: W,W1 & W2

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24 Nov'09Comp30291 DMP Section 521 W1 = 0; W2 = 0; %For delay boxes while 1 X = input(‘X=’) ; % Input to X W =X - b1*W1 - b2*W2; % Recursive part Y = W*a0 + W1*a1 + W2*a2; % Non-rec. part W2 = W1; W1 = W; % For next time disp(sprintf(‘Y=%f’,Y)); % Output Y using disp end; % Back for next sample Program to implement Direct Form II using normal arithmetic

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24 Nov'09Comp30291 DMP Section 522 K=1024; A0=round(a0*K); A1=round(a1*K); A2=round(a2*K); B1=round(b1*K); B2=round(b2*K); W1 = 0; W2 = 0; %For delay boxes while 1 X = input(‘X=’) ; % Assign X to input W =K*X - B1*W1 - B2*W2; % Recursive part W =round( W / K); % By arith right-shift Y = W*A0+W1*A1+W2*A2; % Non-rec. part W2 = W1; W1 = W; %For next time Y = round(Y/K); %By arith right-shift disp(sprintf( ‘Y=%f’, Y)); %Output Y using disp end; % Back for next sample Direct Form II in fixed point arithmetic with shifting

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24 Nov'09Comp30291 DMP Section 523 Re-express as: Poles & zeros of H(z) For a discrete time filter: Now factorise numerator & denominator:

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24 Nov'09Comp30291 DMP Section 524 Poles & zeros of H(z) continued z 1, z 2,..., z N, are ‘zeros’. p 1,p 2,..., p N, are ‘poles’. H(z) generally infinite with z equal to a pole. H(z) generally zero with z equal to a zero. For a causal stable system all poles must satisfy p i < 1. i.e. on Argand diagram: poles must lie inside unit circle. No restriction on the positions of zeros.

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24 Nov'09Comp30291 DMP Section 525 Design of IIR ‘notch’ filter Design a 4th order 'notch' filter to eliminate an unwanted sinusoid at 800 Hz without severely affecting rest of signal. The sampling rate is FS = 10 kHz. One way is to use the MATLAB function ‘butter’ as follows: FL = 800 – 25 ; FU = 800+25; [a b] = butter(2, [FL FU]/(FS/2),’stop’); a = [0.98 -3.43 4.96 -3.43 0.98] b= [ 1 -3.47 4.96 -3.39 0.96] freqz(a, b); freqz(a, b, 512, FS); % Better graph axis([0 FS/2 -50 5]); % Scales axes Notch has -3 dB frequency band: 25 + 25 = 50 Hz. MATLAB response

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24 Nov'09Comp30291 DMP Section 526 Gain/phase response of notch filter

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24 Nov'09Comp30291 DMP Section 527 Gain/phase responses of notch filter

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24 Nov'09Comp30291 DMP Section 528 How sharp is the notch? Can answer this question by specifying notch’s -3 dB bandwidth. Have just designed what Barry calls a 4th order band-stop filter. MATLAB calls it a 2nd order band-stop filter. For a sharper notch, decrease -3 dB bandwidth But this will decrease its ‘depth’, i.e. the attenuation at the ‘notch’ frequency. If necessary, increase the order to 6 (3) or 8 (4). Details

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24 Nov'09Comp30291 DMP Section 529 Sketch the same gain-response The -3dB frequencies are at ( 800 + ) and ( 800 - ) Hz. Given (= 25 Hz say) can sketch gain-response: 1 FS/2 F Gain 0.5 0 800 800 + 800 - 0 dB -3 dB

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24 Nov'09Comp30291 DMP Section 530 Implement the 4 th order ‘notch’ filter MATLAB function gave us: a = [0.98 -3.43 4.96 -3.43 0.98] b= [ 1 -3.47 4.96 -3.39 0.96] Transfer (System) Function is:

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24 Nov'09Comp30291 DMP Section 531 z -1 + + + + + + + + 0.98 3.47 -4.96 3.39 -0.96 0.0.98 x[n] y[n] -3.43 4.96 -3.43 A direct Form 2 implementation of the 4th order IIR notch filter

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24 Nov'09Comp30291 DMP Section 532 Problems with ‘direct form’ IIR implementations Implementation on previous slide works fine in MATLAB. But ‘direct form’ IIR implementations of order >2 are rarely used. Sensitivity to round-off error in coeff values will be high. Also range of ‘intermediate’ signals in z -1 boxes is high. High wordlength floating point arithmetic hides this problem But in fixed point arithmetic, great difficulty occurs. Instead we use ‘cascaded biquad sections’

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24 Nov'09Comp30291 DMP Section 533 Using biquad (2 nd order) IIR sections H(z) x[n]y[n] Given 4 th order H(z), instead of: we prefer to arrange two biquad sectns as follows: H 1 (z)H 2 (z) x[n]y[n] G

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24 Nov'09Comp30291 DMP Section 534 Converting to the new implementation Get a & b for 4 th order H(z) as before: [a b] = butter(2, [FL FU]/(FS/2),’stop’); Then execute: [SOS G] = tf2sos(a,b) MATLAB responds with: SOS = 1 -1.753 1 1 -1.722 0.9776 1 -1.753 1 1 -1.744 0.9785 G = 0.978 Transfer function to 2 nd order sectns First sectn 2 nd sectn

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24 Nov'09Comp30291 DMP Section 535 H(z) may now be realised as:

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24 Nov'09Comp30291 DMP Section 536 Example A digital filter with a sampling rate of 200 Hz is required to eliminate an unwanted 50 Hz sinusoidal component of an input signal without affecting the magnitudes of other components too severely. Design a 4th order "notch" filter for this purpose whose 3dB bandwidth is not greater than 3.2 Hz. Solution method: FS=200; FL=50-1.6; FU=50+1.6; [a b]=butter(2,[FL,FU]/(FS/2), ‘stop’); [SOS G] = tf2sos(a,b)

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24 Nov'09Comp30291 DMP Section 537 Many design techniques for IIR digital filters have adopted ideas of analogue filters. Can transform analogue ‘prototype’ transfer function H a (s) into H(z) for an IIR digital filter. Analogue filters have infinite impulse-responses. Many gain-response approximations exist which are realisable by analogue filters e.g. Butterworth low-pass approximation which can be transformed to high-pass, band-pass & band-stop. IIR digital filter design by bilinear transformation

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24 Nov'09Comp30291 DMP Section 538 Butterworth low-pass gain approximation of order n At C, Gain 0.71 i.e -3 dB

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24 Nov'09Comp30291 DMP Section 539 Can transform H a (s), with gain-response G a ( ), to H(z) for an IIR digital filter with similar gain-response G( ). Many ways exist. Most famous is ‘bilinear transformation’. Replaces s by 2(z-1)/(z+1) to transform H a (s) to H(z). Fortunately MATLAB does all this for us. Transformations from H a (s) to H(z) for IIR digital filter

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24 Nov'09Comp30291 DMP Section 540 Properties of bilinear transformation (i) Order of H(z) = order of H a (s) (ii) If H a (s) is causal & stable, so is H(z). (iii) G( ) = G a ( ) where = 2 tan( /2) So gain of analog filter at radians/second becomes gain of digital filter at radians/sample where = 2tan( /2).

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24 Nov'09Comp30291 DMP Section 541 Frequency warping By (iii), from - to mapped to in range - to .

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24 Nov'09Comp30291 DMP Section 542 Shape of G( ) will change under the transformation. If G a ( ) is Butterworth, G( ) will not have exactly the same shape, but we still call it Butterworth. Mapping approx linear for in the range -2 to 2. As increases above 2, a given increase in produces smaller and smaller increases in . Frequency warping (cont)

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24 Nov'09Comp30291 DMP Section 543 G( ) becomes more and more compressed as . Illustrate for an analog gain-response with ripples: (a): Analogue gain response (b): Effect of bilinear transformation Ga()Ga() G( ) Comparing G a ( ) with G( )

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24 Nov'09Comp30291 DMP Section 544 ‘Prototype’ analogue transfer function Although the shape changes, we would like G( ) at its cut off C to the same as G a ( ) at its cut-off frequency. If G a ( ) is Butterworth, it is -3dB at its cut-off freq So we would like G( ) to be -3 dB at its cut-off C. Achieved if analogue prototype is designed to have its cut- off frequency at C = 2 tan( C /2). C is the ‘pre-warped’ cut-off frequency. Designing analog prototype with cut-off freq 2 tan( C /2) guarantees that the digital filter will have its cut-off at C.

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24 Nov'09Comp30291 DMP Section 545 Let required cut-off frequency C = /4 radians/sample. Need prototype transfer fn H a (s) for 2nd order Butt low-pass filter with 3 dB cut-off at 2tan( C /2) = 2 tan( /8) radians/second. C = 2 tan( /8) = 0.828 I happen to remember that the transfer fn for a 2nd order Butt low- pass filter with cut-off C is: If you don’t believe me, check that replacing s by j and taking the modulus gives G( ) = 1/ [1+( / C ) 2n ] with n=2. Set C = 0.828 in this formula, then replace s by 2(z-1)/(z+1). Gives us H(z) for an IIR digital filter. That’s it! Design 2nd order IIR lowpass digital filter by bilinear transfm

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24 Nov'09Comp30291 DMP Section 546 Resulting IIR digital filter x[n] y[n] 0.098 2 0.94 - -0.33

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24 Nov'09Comp30291 DMP Section 547 Design of 2nd order IIR low-pass digital filter by bilinear transform using MATLAB If required cut-off freq is /4 radians/sample, type: [a b] = butter(2, 0.25) MATLAB gives us: a = [0.098 0.196 0.098] b = [1 -0.94 0.33] The required expression for H(z) is therefore: To save multipliers it is a good idea to re-express this as:

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24 Nov'09Comp30291 DMP Section 548 Realise by ‘direct form 2’ signal-flow graph x[n] y[n] 0.098 2 0.94 -0.33

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24 Nov'09Comp30291 DMP Section 549 Higher order IIR digital filters Example: Design 4th order Butterwth-type IIR low-pass digital filter with 3 dB c/o at f S / 16.. Solution: Relative cut-off frequency is /8. Typing: [a b] = butter(4, 0.125) gives the response: a = 0.0009 0.0037 0.0056 0.0037 0.0009 b = 1 -2.9768 3.4223 -1.7861 0.3556

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24 Nov'09Comp30291 DMP Section 550 z -1 + + + + + + + + 0.00093 2.977 -3.422 1.79 -0.356 0.00093 x[n] y[n] 0.0037 0.0056 0.0037 A direct Form 2 implementation of 4th order IIR filter

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24 Nov'09Comp30291 DMP Section 551 0.00093 x[n] z -1 + + + + + + + + 2.977 -3.422 1.79 -0.356 y[n] 4 6 4 Better ‘direct Form 2’ implementation of 4th order IIR filter

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24 Nov'09Comp30291 DMP Section 552 Higher order IIR digital filters not normally implemented in Direct Form 1 or 2. Instead, implement as cascaded biquad (sos) sections by typing: [a b] = butter(4, 0.125); [sos G] = tf2sos(a,b) MATLAB responds with: sos = 1 2 1 1 -1.365 0.478 1 2 1 1 -1.612 0.745 G = 0.00093 Cascaded bi-quad sections ‘transfer function’ to ‘second order sections’

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24 Nov'09Comp30291 DMP Section 553 H(z) may be realised as: Fourth order IIR Butterworth filter with cut-off fs/16

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24 Nov'09Comp30291 DMP Section 554 Better realisation of H(z) At =0, gain of 1st section is (1+2+1)/(1-1.612+0.0745) =30.12 At =0, gain of 2nd section is (1+2+1)/(1-1.365+0.488) = 35.56 Make gain of each section one at =0 by scaling as follows: Fourth order IIR Butterworth filter with cut-off Fs/16

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24 Nov'09Comp30291 DMP Section 555 Gain-responses for 4th order analog & IIR digital filter

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24 Nov'09Comp30291 DMP Section 556 Compare gain-response of 4th order Butt low-pass transfer function used as a prototype, with that of derived digital filter. Both are 1 (0 dB) at zero frequency. Both are 0.707 (-3 dB) at the cut-off frequency. Analogue gain approaches 0 as whereas digital filter gain becomes exactly zero at = . Shape of Butt gain response is "warped" by bilinear transfn. For digital filter, cut-off rate becomes sharper as because of the compression as .

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24 Nov'09Comp30291 DMP Section 557 High-pass band-pass and band-stop IIR filters Example: 2nd (4th)order bandpass filter with L = /4, u = /2. Solution: [a b] = butter(2,[0.25 0.5]) a = 0.098 0 -0.195 0 0.098 b = 1 -1.219 1.333 -0.667 0.33 freqz(a,b); [sos G] = tf2sos(a,b) sos = 1 2 1 1 -0.1665 0.5348 1 -2 1 1 -1.0524 0.6232 G = 0.098 Many people call this 4th order - MATLAB calls it 2nd order.

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24 Nov'09Comp30291 DMP Section 558 2nd (4th) order IIR bandpass

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24 Nov'09Comp30291 DMP Section 559 Arranged as 2 biquad sections x[n] 0.1 -2.17 -0.54 1 y[n] 2 1.05 -0.62

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24 Nov'09Comp30291 DMP Section 560 Higher order band-stop IIR filter Example: 4th (8th)order bandpass filter with L = /4, u = /2. Solution: [a b] = butter(4,[0.25 0.5],’stop’) a = 0.35 -1.15 2.815 -4.24 5.1 -4.24 2.815 -1.15 0.35 b = 1 -2.472 4.309 -4.886 4.477 -2.914 1.519 -0.5 0.12 freqz(a,b); [sos G] = tf2sos(a,b) sos =1 -.828 1 1 -0.351 0.428 1 -.828 1 1 -0.832 0.49 1 -.828 1 1 -0.046 0.724 1 -0.828 1 1 -1.244 0.793 G = 0.347

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24 Nov'09Comp30291 DMP Section 561 Gain & phase resp of IIR band-stop filter

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24 Nov'09Comp30291 DMP Section 562 Band-pass as arrangement of 4 ‘sos’ biquads H 1 (z)H 2 (z)H 3 (z)H 4 (z) Careful with scaling for band-pass & high-pass sections Can make gain =1 in pass-band for each section. Not at =0 this time !!!

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24 Nov'09Comp30291 DMP Section 563. Must use [sos G] = tf2sos ([a 4 a 3 a 2 a 1 a 0 ], [b 4 … b 0 ]) ‘help tf2sos’ to find out abt this function Transfer-function to second order sections: ‘tf2sos’

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24 Nov'09Comp30291 DMP Section 564 Option of cascading high pass & low-pass digital filters to give band-pass or band-stop filters must be used with care. It is much simpler & avoids the factorisation problem. Then make sure that analogue prototype is wide-band High-pass IIR filters are designed by: [a b] = butter(4,0.125,’high’); Wide-band band-pass & band-stop filters

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24 Nov'09Comp30291 DMP Section 565 Comparison of IIR and FIR digital filters Advantage of IIR type digital filters: Economical in use of delays, multipliers and adders. Disadvantages: (1) Sensitive to coefficient round-off inaccuracies & effects of overflow in fixed point arith. These effects can lead to instability or serious distortion. (2) An IIR filter cannot be exactly linear phase.

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24 Nov'09Comp30291 DMP Section 566 Advantages of FIR filters: (1) may be realised by non-recursive structures which are simpler and more convenient for programming especially on devices specifically designed for DSP. (2) FIR structures are always stable. (3) Because there is no recursion, round-off and overflow errors are easily controlled. (4) An FIR filter can be exactly linear phase. Disadvantage of FIR filters: Large orders can be required to perform fairly simple filtering tasks.

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24 Nov'09Comp30291 DMP Section 567 Problems 1 Find H(z) for the following difference equations (a) y[n] = 2x[n] - 3x[n-1] + 6x[n-4] (b) y[n] = x[n-1] - y[n-1] - 0.5y[n-2] 2 Show that passing {x[n]} thro’ H(z) = z - 1 produces {x[n-1]}. 3 Calculate the impulse-response of the digital filter with 1 H(z) = 1 - 2 z - 1 Draw its signal flow graph, plot its poles and zeros and comment on them.

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24 Nov'09Comp30291 DMP Section 568 4. If LTI systems L1 & L2, with imp-responses {h1[n]} & {h2[n]} are arranged as below, calculate overall impulse- response. Show that this is affected by interchanging L1 & L2. 5. Low-pass IIR digital filter required with cut-off at f s /4 & stop-band attenuation at least 20 dB for all frequencies above 3f s /8 & below f s /2. Design by bilinear transfn using MATLAB.

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24 Nov'09Comp30291 DMP Section 569 6. Design a 4th order band-pass IIR digital filter with lower & upper cut-off frequencies at 300 Hz & 3400 Hz when f S = 8 kHz. 7. Design a 4th order band-pass IIR digital filter with lower & upper cut-off frequencies at 2000 Hz & 3000 Hz when f S = 8 kHz. 8. What limits how good a notch filter we can implement on a fixed point DSP processor? In theory we can make notch sharper & sharper by moving the -3dB points closer & closer. What limits us in practice? How sharp a notch could we could get in 16-bit fixed pt arithmetic? 9. What order of FIR low-pass filter would be required to be approx as good as the 2nd order IIR low-pass filter ( /4 cut-off) designed in these notes?

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Unit 9 IIR Filter Design 1. Introduction The ideal filter Constant gain of at least unity in the pass band Constant gain of zero in the stop band The.

Unit 9 IIR Filter Design 1. Introduction The ideal filter Constant gain of at least unity in the pass band Constant gain of zero in the stop band The.

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