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Analogue filtering UNIVERSITY of MANCHESTER School of Computer Science

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1 Analogue filtering UNIVERSITY of MANCHESTER School of Computer Science
11/16/2018 UNIVERSITY of MANCHESTER School of Computer Science Comp30291: Digital Media Processing Section 2 Analogue filtering 4 Oct 2008 Comp30291 Section 2 Comp Oct'08

2 Analog system represented as ‘black box’
x(t) y(t) Inside we could have analogue components, or Digital processor Analog lowpass filter 1 x(t) Analog lowpass filter 2 ADC y(t) DAC Fs Fs 4 Oct 2008 Comp30291 Section 2

3 Analog low-pass filters
Analog Lowpass Filter 1 is ‘antialiasing’ filter: removes any frequency components above Fs/2 before sampling process. Analog Lowpass Filter 2 is ‘reconstruction’ filter: smoothes DAC output to remove all frequency components above Fs/2. Digital processor controls ADC to sample at Fs Hz. Also sends output sample to DAC at Fs samples per second. DAC produces ‘staircase’ waveform: smoothed by ALpF2. DAC output t 4 Oct 2008 Comp30291 Section 2

4 Analogue filters Still needed in the world of DSP
Also, many digital filter designs are based on analog filters. They are ‘linear’ & ‘time-invariant’ (LTI) Analogue filter x(t) y(t) 4 Oct 2008 Comp30291 Section 2

5 Definition of ‘linearity’
System is LINEAR if for any signal x(t), if x(t)  y(t) then a.x(t )  a.y(t) for any constant a. (2) for any signals x1(t) & x2(t), if x1(t)  y1(t) & x2(t)  y2(t) then x1(t) + x2(t)  y1(t) + y2(t) (By x(t)  y(t) we mean that applying x(t) to the input produces the output signal y(t). ) 4 Oct 2008 Comp30291 Section 2

6 Alternative definition of ‘linearity’
System is linear if for any signals x1(t) & x2(t), if x1(t)  y1(t) & x2(t)  y2(t) then a1x1(t) + a2 x2(t)  a1y1(t) +a2 y2(t) for any a1 & a2 4 Oct 2008 Comp30291 Section 2

7 Linearity (illustration)
system If x1(t) y1(t) & x2(t) y2(t) then 3x1(t)+4x2(t)  3y1(t)+4y2(t) t x2(t) y2(t) x1(t) y1(t) + 4 Oct 2008 Comp30291 Section 2

8 Definition of ‘time-invariance’
A time-invariant system must satisfy: For any x(t), if x(t)  y(t) then x(t-)  y1(t-) for any  Delaying input by  seconds delays output by  seconds Not all systems have this property. An LTI system is linear & time invariant. An analogue filter is LTI. 4 Oct 2008 Comp30291 Section 2

9 ‘System function’ for analogue LTI circuits
An analog LTI system has a system (or transfer) function a0 + a1s + a2s aNsN H(s) =  b0 + b1s + b2s bMsM Coeffs a0, a1, ...,aN, b0, ..., bM determine its behaviour. Designer of analog lowpass filters must choose these carefully. H(s) may be evaluated for complex values of s. Setting s = j where  = 2f gives a complex function of f. Modulus |H(j)| is gain at  radians/second (/2 Hz) Argument of H(j) is phase-lead at  radians/s. 4 Oct 2008 Comp30291 Section 2

10 Gain & phase response graphs
Gain: G() = |H(j)| Phase lead: () = Arg[H(j)| -() G(w) Gain Phase-lag f / (2) 4 Oct 2008 Comp30291 Section 2

11 Effect of phase-response
It may be shown that: when input x(t) = A cos(t), output y(t) = A . G() . cos( t +() ) Output is sinusoid of same frequency as input. ‘Sine-wave in  sine-wave out’ Multiplied in amplitude by G() & ‘phase-shifted’ by (). Example: If G() = 3 and () = /2 for all  what is the output? Answer: y(t) = 3.A.cos(t + /2) = 3.A.sin (t) 4 Oct 2008 Comp30291 Section 2

12 Phase-shift expressed as a delay
Express y(t) = A . G() . cos( t +() ) as A . G() . cos ([t + ()/]) = A . G() . cos( [t - ()] ) where () = -()/ Cosine wave is delayed by -()/ seconds. -()/ is ‘phase-delay’ in seconds Easier to understand than ‘phase-shift’ 4 Oct 2008 Comp30291 Section 2

13 Linear phase If -()/ is constant for all , all frequencies delayed by same time. Then system is ‘linear phase’ - this is good. Avoids changes in wave-shape due to ‘phase distortion’; i.e different frequencies being delayed by differently. Not all LTI systems are ‘linear phase’. 4 Oct 2008 Comp30291 Section 2

14 Linear phase response graph
4 Oct 2008 Comp30291 Section 2

15 Low-pass analog filters
Would like ideal ‘brick-wall’ gain response & linear phase response as shown below: () G() 1 C C = cut-off frequency 4 Oct 2008 Comp30291 Section 2

16 Butterworth low-pass gain response
Cannot realise ideal ‘brick-wall’ gain response nor linear phase. Can realise Butterworth approximation of order n: Properties (i) G(0) = ( 0 dB gain at =0) (ii) G(C) = 1/(2) ( -3dB gain at  = C) 4 Oct 2008 Comp30291 Section 2

17 Examples of Butterwth low-pass gain responses
Let C = 100 radians/second. G(C) is always 1/(2) Shape gets closer to ideal ‘brick-wall’ response as n increases. 4 Oct 2008 Comp30291 Section 2

18 n=7 n = 2 n=4 LINEAR-LINEAR PLOT G() 1 / (2) radians/second 1 0.9
0.8 1 / (2) 0.7 0.6 0.5 0.4 0.3 0.2 n=7 n=4 n = 2 0.1 50 100 150 200 250 300 350 400 radians/second 4 Oct 2008 Comp30291 Section 2

19 Butterworth gain responses on dB scale
Plot G() in dB, i.e. 20 log10(G()), against . With  on linear or log scale. As 20 log10(1/(2)) = -3, all curves are -3dB when  = C 4 Oct 2008 Comp30291 Section 2

20 dB-LINEAR PLOT dB -10 n=2 -20 -30 n=4 -3dB -40 -50 -60 n=7 -70 -80 -90
dB -10 n=2 -20 -30 n=4 -3dB -40 -50 -60 n=7 -70 -80 -90 50 100 150 200 250 300 350 400 radians/second 4 Oct 2008 Comp30291 Section 2

21 dB-LOG PLOT dB 3 dB n=2 n=4 radians/second -10 -20 -30 -40 -50 -60
dB -10 3 dB -20 -30 n=2 -40 -50 -60 n=4 -70 -80 1 2 3 10 10 10 10 radians/second 4 Oct 2008 Comp30291 Section 2

22 MATLAB program to plot these graphs
clear all; for w = 1 : 400 G2(w) = 1/sqrt(1+(w/100)^4); G4(w) = 1/sqrt(1+(w/100)^8) ; G7(w) = 1/sqrt(1+ (w/100)^14); end; plot([1:400],G2,'r',[1:400],G4,'b',[1:400],G7,'k'); grid on; DG2=20*log10(G2); DG4=20*log10(G4); DG7=20*log10(G7); plot([1:400],DG2,'r',[1:400],DG4,'b',[1:400],DG7,'k'); grid on; semilogx([1:990], DG2,'r', [1:990], DG4, 'b’); 4 Oct 2008 Comp30291 Section 2

23 ‘Cut-off’ rate Best seen on a dB-Log plot
Cut-off rate is 20n dB per decade or 6n dB per octave at frequencies  much greater than C. Decade is a multiplication of frequency by 10. Octave is a multiplication of frequency by 2. So for n=4, gain drops by 80 dB if frequency is multiplied by 10 or by 24 dB if frequency is doubled. 4 Oct 2008 Comp30291 Section 2

24 Filter types - low-pass
G() 1 1 radian/s Ideal Approximatn Low-pass with C = 1 G() C Low-pass 4 Oct 2008 Comp30291 Section 2

25 Filter types - high-pass & band-pass
1 High-pass C G() Band-pass L U 4 Oct 2008 Comp30291 Section 2

26 Filter types - band-stop
G() 1 L U 4 Oct 2008 Comp30291 Section 2

27 Two types of band-pass gain-responses
1 Narrow-band (U < 2 L ) L U G() 1 Broad-band (U > 2 L ) L U 4 Oct 2008 Comp30291 Section 2

28 Three types of ‘band-stop’ gain-responses
1 Narrow-band (U < 2 L ) L U G() 1 Broad-band (U > 2 L ) L U 4 Oct 2008 Comp30291 Section 2

29 Third type of ‘band-stop’ gain-response
1 Notch N Yet another type of gain-response G() 1 All-pass 4 Oct 2008 Comp30291 Section 2

30 Approximatns for high-pass, band-pass etc
Fortunately these can be derived from the formula for a Butterworth LOW-PASS gain response. MATLAB does all the calculations. 4 Oct 2008 Comp30291 Section 2

31 Other approximations A filter which uses a Butterworth gain-response approximation of order n is an ‘nth order Butterworth type filter’. In addition to Butterworth we have other approximations Chebychev (types 1 & 2) Elliptical Bessel, etc. 4 Oct 2008 Comp30291 Section 2

32 Problems 1. An analog filter has: H(s) = 1 / (1 + s)
Give its gain & phase responses & its phase delay at  = 1. 2. Use MATLAB to plot gain response of Butterwth type analog low-pass filter of order 4 with C = 100 radians/second. Solution to (2):- for w = 1 : 400 G(w) = 1/sqrt(1+(w/100)^8) ; end; plot(G); grid on; 4 Oct 2008 Comp30291 Section 2

33 Result obtained:- 4 Oct 2008 Comp30291 Section 2


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