EE4192 Filters : Examination questions

26/03/03

EE4192: Examination June 2002 Answer THREE questions in TWO hours

1. Deriving the formulae used obtain the transfer function for the first and second order filter sections shown in Figure Q1. State the approximations used in your calculations. [6 marks] Make a separate sketch of the AC magnitude response for each of the filter sections.[4 marks] What is the overall DC gain of the filter shown in the figure? [3 marks] From your knowledge of filter approximations (types of filter) determine the type of filter implemented by the circuit shown in Figure Q1. Justify your answer in terms of the poles and zeros of the circuit.[7 marks]

530pF

vi

39k

100k

47k 1.1nF

47k vo 1.1nF 180k 180k

Figure Q1
[Remember the filter has been designed using standard component values and this leads to slight inaccuracies in the implementation] 2. Compare the properties of Butterworth, Chebyshev and Bessel-Thomson types of filter using the following headings: a). AC magnitude response [6 marks] b). step response [6 marks] c). location of poles on the complex plane [6 marks] Suggest the application areas to which these filter types would be best suited [2 marks]

3. (a) Outline the methodology and the main properties of each of the following three IIR digital filter design techniques. Explain the advantages and disadvantages of each design technique. (i) derivative approximation technique [4 marks] (ii) bilinear transformation method [4 marks] (iii) impulse-response invariant technique [4 marks] (b) Given the following analogue system function, 5 + s Ha(s) = 2 + 3 s + s2 produce a discrete time filter with frequency response approximating H(j), using the impulse-response invariant technique, assuming a sampling frequency of 10 Hz. [4 marks] Give the signal flow graph for the digital filter and give a flow diagram to indicate how it would be implemented on a DSP processor. [4 marks] 4. (a) Explain the following terms as applied to analogue and digital filters: (i) phase delay [2 marks] (ii) group delay [2 marks] (iii) linear phase [1 mark] (iv) minimum phase [1 mark]

EE4192 Filters : Examination questions

26/03/03

Explain why discrete time FIR filters be can made exactly linear phase whereas analogue and digital filters with infinite impulse-responses cannot. [4 marks] (b) A radar system receives an echo of its transmitted signal. Both signals are sampled at 1/T Hz. and must be compared to determine the delay between them to an accuracy of T/2 seconds, i.e. half a sampling interval. Explain how this may be achieved with the use of an FIR discrete time filter acting as a "fractional delay filter". [2 marks] Show that a discrete time filter whose frequency response is H(ej) = e-jM for all for some non-integer value of M would realise any required fractional delay, and calculate the impulse response of this filter. [4 marks] Explain why this filter cannot be realised exactly when M=2.5 and show how a fifth order FIR approximation may be obtained by means of the Fourier series approximation technique. [4 marks] EE4192 June 2002 Solution methods 4 (a) (i) Phase delay: For a discrete time filter, phase delay = -()/ sampling intervals where is phase lead and frequency in radians per sampling interval. This is the amount of time by which a sinusoid of frequency will be delayed. For continuous time, frequency is in radians/second and units of delay are in seconds. (ii) Group delay: -d()/d sampling intervals. Envelope delay i.e. delay to envelope carried by sinusoids in narrow band of frequencies centred on . (iii) Linear phase: If -()/ remains constant for all the filter is linear phase and all sinusoids will suffer same delay regardless of frequency . (iv) Minimum phase: all zeros as well as all poles inside the unit circle (discrete time) or on right hand side of s plane (continuous time). A linear phase analogue system has ( ) = D sec onds for all where D is a constant called the phase delay. It was shown that the impulse response must be symmetric about t = D i.e. h(D + t) = h(D - t) for all t. Now if h(t) is an infinite impulse response it remains non-zero as t . Therefore it must remain non-zero as t -. Therefore it must be non-zero for some values of t < 0. It must be non-causal, whatever the value of D. A non-causal system cannot be realised. Similarly for the impulse response, h[n], if the system is discrete time and the impulse response is IIR. An FIR filter can be made linear phase since the impulse response becomes zero for all n D+P for some finite value of P. It will be zero for all n D-P.

EE4192 Filters : Examination questions

26/03/03

If it can be arranged that D-P is positive or zero, the FIR discrete time filter will be causal. We can simply delay the impulse response to make D as large as we like. Therefore for any value of P we can always make D large enough to ensure that D-P 0. Then h[n] = 0 for n < 0. Causal and can be realised by an FIR filter. 4(b) Implement filters delaying original signal x(t) by N samples and the echo e(t) by N+0.5 samples. N is chosen to allow a sufficiently high order fractional delay filter to be made causal. Cross correlate outputs to compare.
x[n] z-1 z-1

z-1

z-1

z-1

z-1

z-1

a0

a1

a5
Crosscorrelatn

To design the FIR filter, set H(e-j) = e-jM By inverse DTFT, impulse response is {h[n]} where h[n] = 1 2 1 = 2 = = 1 2

H (e
j ( n M )

)e jn d

e e

jM

e jn d d = 1 e j ( n M ) 2j (n M )

1 e j ( n M ) e j ( n M ) 2j (n M ) sin( (n M )) = assuming n M (n M ) The gain of this filter will be unity for all and its phase response would be linear phase with a slope of M. The output will be to produce samples of x(t-MT) when the input is obtained by sampling x(t) at 1/T Hz. If M is not an integer this is a "fractional delay" FIR filter. Samples of h[n] are as in the table below:-

EE4192 Filters : Examination questions n 0 1 2 3 -1 -2 h[n] sin(-/2)/(-/2) = 2/() = 0.64 2/ = 0.64 -2/(3) = -0.212 +2/(5) = 0.127 -2/3 = -0.212 -2/5 = 0.127

26/03/03

The filter with impulse response {h[n]} cannot be realized exactly because the impulse response is infinite and non-causal; i.e. t is non-zero for values of n less than zero. To approximate by realizable FIR filter of order 5, using rectangular window for simplicity, truncate to 6 terms and realize as follows:

z-

z-1

z-1

z-1

z-

EE4192 June 2002 Solution methods (cont.) 3(a) (i) derivative approximation technique: The differential equation for an analogue filter with input x(t) and output y(t) may be derived from H(s). a0 + a1 s + a2 s2 + .. If H(s) = 1 + b1 s + b2 s2 + Then y(t) + b1 dy(t)/dt + b2 d2y(t)/dt2 + = a0x(t) + a1dx(t)/dt + a2 d2x(t)/dt2 + If x(t) and y(t) are sampled at 1/T Hz to give {x[n]} and {y[n]} respectively, then we can approximate dy(t)/dt by (y[n] - y[n-1]) / T and dx(t)/dt by (x[n]-x[n-1])/T. We approximate d2y(t)/dt2 by (1/T) (dy(t)/dt) - dy(t-T)/dt) (1/T) ( (y[n] - y[n-1])/T - (y[n-1] - y[n-2])/T) = (1/T2) (y[n] - 2y[n-1]) + y[n-2] ) A similar approach may be used to approximate d2x(t)/dt2 and it may be readily extended to d3y(t)/dt3 etc. Substituting into the differential equation produces a difference equation in the form: y[n] +B1y[n-1] + B2 y[n-2] + = A0x[n] + A1x[n-1] + A2 x[n-2] +

EE4192 Filters : Examination questions

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which may be realised as a digital filter. The derivative approximation technique may be applied directly to H a (s) by simply replacing s by (1 - z 1 )/T to obtain the required function H(z). It is not commonly used. It is quite accurate at lower frequencies and becomes less so as the frequency becomes closer to 1/T since T can no longer be considered small in comparison to the changes that occur from x[n-1] to x[n] and from y[n-1] to y[n]. Advantages: a simple technique to understand and calculate. Preserves order of the analogue filter. Disadvantages: More accurate techniques exist. Stability not guaranteed. (ii) bilinear transformation method: Given Ha(s) replace s by (0.5/T) (z-1) / (z+1) or equivalently (1/(2T))(1-z-1)/(1+z-1) which is equivalent to approximating: dy(t)/dt + dy(t-T)/dt 2 by (y[n] - y[n-1]) / T instead of simply dy(t)/dt

To achieve this for the differential equn: y(t) + b1 dy(t)/dt + = a0x(t) + a1dx(t)/dt + add the delayed version: y(t-T) + b1 dy(t-T)/dt + = a0x(t-T) + a1dx(t-T)/dt + and the substitution is then straightforward. This is the most commonly used IIR digital filter design technique by far. Produces a digital filter with frequency response H(ej) is related to Ha(j) by the "frequency warping" formula: H(exp(j)) = H a (j) where = (2/T) tan(/2) Advantages: Produces a digital filter of the same order as the analogue filter. The digital filter is always stable when Ha(s) has all poles on the left hand side of the s-plane. Effect of frequency warping can be anticipated when designing Ha(s). Disadvantages: The effect of frequency warping changes the shape of the gain and phase response. This must be anticipated when designing Ha(s). The effect of frequency warping on the phase response means that an approximately linear phase analogue filter will not produce an approximately linear phase discrete time filter. (iii) impulse-response invariant technique: The philosophy of this technique is to transform an analogue prototype filter into an IIR discrete time filter whose impulse response {h[n]} is a sampled version of the analogue filters impulse response. Assuming the analogue filters impulse response to be h a (t), then h[n] must be equal to h a (nT) for all n. )It is best to scale ha(t) by T where 1/T is the sampling rate.)

EE4192 Filters : Examination questions

26/03/03

The impulse response can be obtained from Ha(s) be means of an inverse Laplace transform. It is convenient to express Ha(s) in partial fraction form to achieve this easily. Thus: H a (s) = A1 s-p1 + A2 s - p2 + A3 s + p3 + ...

The corresponding impulse response is then: 0 : t <0 h a (t) = A 1 exp(p 1 t) + A 2 exp(p 2 t) + ... : t 0 Multiplying by the constant T, replacing t by nT and taking z transforms gives: TA1 1 - exp(p 1 T)z - 1 TA2 1 - exp(p 1 T)z - 1

H(z) =

+ ...

Complex conjugate pairs of perms may be combined to obtain 2nd order sections. Advantage: Produces reasonably accurate responses without frequency warping. This is useful where the phase response must be preserved, especially an approximately linear phase response. Always stable and preserves order. Disadvantage: Aliassing occurs in the sampling of the analogue impulse response. If it is not bandlimited, this will be serious. Hence this technique works reasonably for low-pass and band-pass filters but not for high-pass & band-stop filters 3 (b) Ha(s) = By Partial fractions, Ha(s) = 4 (1+s) 3 (2+s) 5 + s 2 + 3 s + s2 and 1/T = 10 making T = 0.1.

H(z)

4 x 0.1 (1 - e-0.1z-1)

3 x 0.1 (1 - e -0.2z-1) 0.3 (1 - 0.82 z-1)

0.4 (1 - 0.90 z-1)

EE4192 Filters : Examination questions

26/03/03

0.1 - 0.06 z-1 (1 - 1.72 z-1 + 0.74 z-2)

Signal flow-graph:-

x[n] z-1

0.1 y[n] 1.72

z-1 -0.74

-0.06

Flow diagram:Set variables W1 & W2 to zero ; Begin: Input next sample into X ; Set W = X + 1.72 * W1 - 0.74 * W2 ; Set output Y = 0.1 * W - 0.06 * W1 ; Set W2 = value of W1 for next loop ; Set W1 = value of W for next loop ; Go back to Begin

EE4192 Filters : Examination questions EE3192 Exam May 2001

26/03/03

3. An ideal quadrature phase band-pass discrete time filter has frequency response:

0 : /4 j : /4 H ( e j ) = + j : 3 / 4 0 : 3 / 4

/4 < < 3 / 4 < < /4 ||

[2 marks]

Sketch its gain and phase response. Show that the response of such a filter to the discrete time input signal {A cos (0 n) } is the output signal

[4 marks] { A sin (0 n) } for any frequency 0 in the range /4 < 0 < 3/4. Calculate the impulse-response {h[n]} of this ideal filter, and hence design a sixth order causal FIR digital filter whose impulse-response is a rectangularly windowed and suitably delayed version of {h[n]}. [6 marks] Give its signal flow-graph in a form which minimises the number of multipliers required. [3 marks] By means of a flow-diagram, or otherwise, indicate how this signal flow graph would be implemented on a DSP processor with integer arithmetic only. [3 marks] How will the gain and phase response of this sixth order digital filter differ from those of H(ej) as specified in the formula above? [ 2 marks] 4.(a) How may an analogue low-pass transfer function Ha(s) with cut-off frequency 1 radian/second be transformed to a highpass transfer function with cut-off frequency C ? If an nth order Butterworth low-pass transfer function were transformed in this way, what would be the resulting gain-response? [2 marks]

Explain how an analogue band-pass transfer function with cut-off frequencies L and U be obtained, in some cases, by cascading a low-pass and a high-pass transfer function. [2 marks]
(b) An IIR discrete time band-pass filter is required with 3dB cut-off frequencies at 400 Hz and 2 kHz. The sampling frequency is 8 kHz. Assuming a pass-band gain of 0 dB, the gain must be less than -20 dB at frequencies below 50 Hz and at frequencies between 3 kHz and 4 kHz. Show that it is reasonable to design the filter by applying the bilinear transformation method to a Butterworth low-pass transfer function and a separate high-pass transfer function to be arranged in cascade. [2 marks] Show that to meet the specification, the minimum order of low-pass and high-pass filter is 3 and 1 respectively. [4 marks] Design the IIR discrete time band-pass filter and give its signal flow graph. [10 marks] The general formula for an nth order Butterworth low-pass analogue transfer function with cut-off frequency C: 1 Ha(s) = [n/2] P (1+s/c) {1 + 2sin[(2k-1)/2n]s/c + (s/c)2} k=1 where [n/2] is the integer part of n/2 and P=0 or 1 depending on whether n is even or odd. Solutions: Handwritten solutions are available