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I

6.1

n this chapter, several engineering applications of the Fourier transform are considered. The mathematical basis and several properties of the Fourier transform are presented in Chapter 5. We present additional examples of how the Fourier transform, and the frequency domain in general, can be used to facilitate the analysis and design of signals and systems.

IDEAL FILTERS The concept of the transfer function, which is one of the ways the Fourier transform is applied to the analysis of systems, is discussed in Chapter 5: [eq(5.46)] H(v) = V2(v) . V1(v)

Here, V1(v) is the Fourier transform of the input signal to a system and V2(v) is the Fourier transform of the output signal. Consideration of this concept leads us to the idea of developing transfer functions for special purposes. Filtering is one of those special purposes that is often applied in electronic signal processing. Figure 6.1 shows the frequency-response characteristics of the four basic types of filters: the ideal low-pass filter, the ideal high-pass filter, the ideal bandpass filter, and the ideal bandstop filter. These ideal filters have transfer functions such that the frequency components of the input signal that fall within the passband are passed to the output without modification, whereas the frequency components of the input signal that fall into the stopband are completely eliminated from the output signal. Consider the frequency response shown in Figure 6.1(a). This is the magnitude frequency spectrum of an ideal low-pass filter. As can be seen, this filter has a unity magnitude frequency response for frequency components such that v F vc and zero frequency response for v 7 vc. The range of frequencies v F vc is called the passband of the filter, and the range of frequencies v 7 vc is called the

274

Sec. 6.1

Ideal Filters

H() 1

275

Stopband c

c

Stopband

Passband

c

Passband

StopPassband band 2 1

Passband

1 2

Stopband

PassStopband band 2 1

Passband 0 (d)

Stopband Passband

1 2

stopband. The output of this filter consists only of those frequency components of the input signal that are within the passband. Figure 6.2 illustrates the effect of the ideal low-pass filter on an input signal. Filters are used to eliminate unwanted components of signals. For example, the high-frequency noise shown to be present in V1(v) in Figure 6.2(b) is outside the passband (in the stopband). Therefore, this noise is not passed through the filter to V2(v), and the desired portion of the signal is passed unaltered by the filter. This filtering process is illustrated in Figure 6.2(c) and (d). It should be noted that the filters described previously are called ideal filters. As with most things we call ideal, they are not physically attainable. However, the concept of the ideal filter is very helpful in the analysis of linear system operation, because it greatly simplifies the mathematics necessary to describe the process. That ideal filters are not possible to construct physically can be demonstrated by reconsideration of the frequency response of the ideal low-pass filter. The transfer function of this filter can be written as H(v) = rect(v/2vc).

276

v 1(t) Ideal low-pass filter (a) V1() V0 Noise c (b) H() A

c

v 2(t)

Chap. 6

Information signal

c (d)

Therefore, from Table 5.2, its impulse response is h(t) = F-15H(v)6 = (vc/p) sinc(vct),

as sketched in Figure 6.3. It is seen that the impulse response for this ideal filter begins long before the impulse occurs at t = 0 (theoretically, at t = - q ). Systems such as this, which respond to an input before the input is applied, are called noncausal systems, as discussed in Chapters 2 and 3. Of course, the physical existence of noncausal systems is impossible. However, the concept of noncausal systems, such as ideal filters, can be useful during the initial stages of a design or analysis effort. The following examples illustrate some applications of the ideal filter concept:

Sec. 6.1

Ideal Filters

H() 1

277

c (a) h(t)

c

0 (b)

EXAMPLE 6.1

Application of an ideal high-pass filter Two signals, g1(t) = 2 cos(200pt) and g2(t) = 5 cos(1000pt), have been multiplied together as described in Example 5.9. The product is the signal g3(t) = 5 cos(1200pt) + 5 cos(800pt). For this example, assume that a certain application requires g4(t) = 3 cos(1200pt). This can be obtained from g3(t) by a high-pass filter. The Fourier transform of g4(t) is found, from Table 5.2, to be G4(v) = 3p[d(v - 1200p) + d(v + 1200p)]. Similarly, the Fourier transform of g3(t) is found by Table 5.2 and the linearity property: G3(v) = 5p[d(v - 800p) + d(v + 800p)] + 5p[d(v - 1200p) + d(v + 1200p)]. The frequency spectra of g4(t) and g3(t) are shown in Figure 6.4(a) and (b), respectively. It can be seen that if the frequency components of G3(v) at v = ; 1200p are multiplied by 0.6, and if the frequency components at v = ; 800p are multiplied by zero, the result will be the desired signal, G4(v). An ideal high-pass filter that will accomplish this is shown in Figure 6.4(c). The filtering process can be written mathematically as G4(v) = G3(v)H1(v), where H1(v) = 0.6[1 - rect(v/2vc)], 800p 6 vc 6 1200p. I

278

G4() 3 1200 800 0 (a) G3() 5 5 0 (b) H1() 0.6 5 800 800

Chap. 6

3 1200

5 1200

1200 c 800

1200 c 800

0 (c)

800

1200

EXAMPLE 6.2

Application of an ideal low-pass filter We work with the signals from Example 6.1. Assume this time that an application requires a signal g5(t) = 4 cos(800pt). This can be obtained from g3(t) by a low-pass filter. The Fourier transform of g5(t) is G5(v) = 4p[d(v - 800p) + d(v + 800p)]. The frequency spectrum of g5(t) is shown in Figure 6.5. To pass the frequency components of G3(v) at v = + - 800p with an output amplitude of 4p requires a gain of 0.8, as shown in the ideal low-pass filter in Figure 6.6. Again, the filtering process is written as G5(v) = G3(v)H2(v),

G5() 4 4

800

800

Sec. 6.1

Ideal Filters

H2()

279

1200 c

800

800

1200

EXAMPLE 6.3

Multiplication of two signals The multiplication of two signals, as considered in Examples 6.1 and 6.2, is a simple mathematical concept, and there are some integrated circuit devices that will accomplish this function. One way we generate the product of two signals by using basic electronic components is shown in Figure 6.7. The square-law device shown in Figure 6.7 is an approximation of the effect of passing the signal through a nonlinear device such as an amplifier biased near the saturation level [1]. The output of the square-law device is the square of the input signal. In the system shown, the input to the device is w(t) = C1 cos(v1t) + C2 cos(v2t). Therefore, the output signal is

2 2 2 x(t) = w2(t) = C2 1 cos (v1t) + 2C1C2 cos(v1t) cos(v2t) + C2 cos (v2t).

Using the trigonometric identity of Appendix A, namely, cos2f = 1 2 [1 + cos 2f], we can rewrite the output of the square-law device as

2 x(t) = 1 2 C1 [1 + cos(2v1t)] + 2C1C2cos(v1t)cos(v2t) 2 + 1 2 C2[1 + cos(2v2t)].

As shown in Example 5.9 [or by use of another trigonometric identity, 2 cos a cos b = cos(a - b ) + cos(a + b )], the second term of this expression can be rewritten as 2C1C2 cos(v1t) cos(v2t) = C1C2 cos [(v1 + v2)t] + C1C2 cos [(v1 - v2)t].

w (t)

Square-law device

280

Chap. 6

1 2 1 2 2 2 x(t) = 1 2 [C1 + C2] + 2 C1cos(2v1t) + 2 C2cos(2v2t)

+ C1C2cos [(v1 + v2)t] + C1C2cos [(v1 - v2)t]. The Fourier transform of this signal is found, from Table 5.2 and the linearity property, to be

2 X(v) = p[C2 1 + C2]d(v) +

+ C1C2 p[d(v - v1 + v2) + d(v + v1 - v2)] + C1C2 p[d(v - v1 - v2) + d(v + v1 + v2)]. If the signals considered in Example 6.1 are added together to form the input to the system in Figure 6.7, then w(t) = g1(t) + g2(t) = 2 cos(200pt) + 5 cos(1000pt) and x(t) = 4 cos 2(200pt) + 20 cos(200pt) cos(1000pt) + 25 cos 2(1000pt). The Fourier transform of the signal x(t) is X(v) = 29p(v) + 2p[d(v - 400p) + d(v + 400p)] + 12.5p[d(v - 2000p) + d(v + 2000p)] + 10p[d(v - 800p) + d(v + 800p)] + 10p[d(v - 1200p) + d(v + 1200p)]. The frequency spectrum of x(t) is shown graphically in Figure 6.8(a). If we desire the output of the system to be y(t) = 3 cos(800pt), as in Example 6.1, we require that Y(v) = 3p[d(v - 800p) + d(v + 800p)]. This result can be achieved by multiplying X(v) by a transfer function H(v) such that the frequency components in the ranges 400p 6 v 6 1200p and - 1200p 6 v 6 - 400p are multiplied by 0.3 and the frequency components outside those ranges are multiplied by zero. The frequency response of an ideal bandpass filter that accomplishes this multiplication is shown in Figure 6.8(b). Figure 6.8(c) shows the frequency spectrum of the systems output signal.

Sec. 6.2

Real Filters

X() 29 12.5 10 10 10 10 12.5

281

2.0

0.4 (a)

0.8

1.2

2.0 (krad/s)

H() 0.3

0.6

(krad/s)

0.8

(krad/s)

Examples 6.1 and 6.2 illustrate the concept of modifying signals by the application of filters. The filters considered in this section are ideal filters. It was shown that ideal filters are not physically realizable. Therefore, the results shown in these examples are not achievable with physical systems. However, the results can be approximated by physical systems, and the concept of the ideal filter is useful for simplifying the analysis and design processes. 6.2 REAL FILTERS The ideal filters described in Section 6.1 are not physically realizable. This was shown by examining the inverse Fourier transform of the frequency-domain functions that describe the ideal low-pass filter frequency response. The impulse response of the ideal low-pass filter implies a noncausal system. Similar analyses could be used to show that the ideal bandpass, ideal high-pass, and ideal bandstop filters are also physically unrealizable.

282

R v i (t) C

Chap. 6

RC Low-Pass Filter Figure 6.9 shows the schematic diagram of an RC low-pass filter. We now find the frequency response function of this electrical network and show that it is an approximation of the ideal low-pass filter. To find the frequency-response function, we begin by writing the differential equations that describe the voltages and current in the circuit: vi(t) = Ri(t) + 1 i(t)dt; CL -q

t

vo(t) =

1 i(t)dt. CL -q

After finding the Fourier transform of each equation, term by term, we have Vi(v) = RI(v) + 1 1 I(v),Vo(v) = I(v). jvC jvC

Therefore, the frequency-response function that describes the relationship between the input voltage vi(t) and the output voltage vo(t) in the frequency domain is H(v) = Vo(v) 1 = . Vi(v) 1 + jvRC

the frequency response function can be rewritten as H(v) = 1 = H(v) ej(v). 1 + jv/vc (6.1)

The magnitude and phase frequency spectra of the filter are described by the equations H(v) = 1 21 + (v/vc)2 and (v) = - arctan(v/vc),

respectively. The magnitude frequency spectrum of the filter is shown in Figure 6.10.

Sec. 6.2

Real Filters

H() 1.2

283

H() 1

0.4

0.2

0 10

1 0

10 /c

It should be noted that at the frequency v = vc, the magnitude ratio between the input and output signals is H(vc) = Vo(vc) Vi(vc) = 1 22 .

The ratio of the normalized power (normalized average power is defined in Section 5.1) of the input and output signals is given by H(vc) 2 = Vo(vc) 2 Vi(vc) 2 = 1 . 2

Because of this relationship, the cutoff frequency of this type of filter is often called the half-power frequency. Lets compare the polar form of H(v) as given in (6.1) with the form of the Fourier transform of a time-shifted function [eq(5.13)]

f f(t - t0) F(v)e-jvt0.

We might suspect that the phase angle, (v) is somehow related to a time shift caused by this circuit as a signal is processed through it. This is indeed true. The

284

Chap. 6

time shift involved is called the phase delay, and it is a function of the signal frequency. This time delay (or phase shift if we consider the frequency-domain manifestation) is a characteristic of all physically realizable filters. Generally, the more closely a physical filter approximates an ideal filter, the more time delay (negative phase shift) is present in the output signal. It is apparent from a comparison of the magnitude frequency spectrum of the RC low-pass filter, shown in Figure 6.10, with those of the ideal low-pass filter, shown in Figure 6.11, that the RC low-pass filter is a relatively crude approximation of the ideal low-pass filter. The magnitude ratio of the RC low-pass filter decreases gradually as the frequency increases, instead of remaining flat, for 0 F v F vc. Also, the magnitude ratio is finite instead of zero for v 7 vc. Engineers who design analog electronic filters try to achieve a good-enough approximation to the ideal filter by designing for a flat-enough frequency response in the passband and a steep-enough roll-off at the cutoff frequency. One of the filter designs that is commonly used to satisfy these criteria is the Butterworth filter. Butterworth Filter The general form of the magnitude frequency-response function for the Butterworth filter is H(v) = 1 21 + (v/vc)2N , (6.2)

where N is called the order of the filter. In other words, N is the order of the differential equation needed to describe the dynamic behavior of the filter in the time domain. By comparing (6.1) and (6.2), we can see that the RC low-pass filter is a first-order Butterworth filter. Figure 6.12(a) and (b) show RLC realizations of second- and third-order Butterworth filters, respectively. The values of the electrical components are determined by the desired cutoff frequency, vc, as indicated in the figure. It is left as an

H()

1

/c

Sec. 6.2

Real Filters

L R0 c 2 2 C c R0 (a) vo(t) L 4R0 3c 3 2R0c

285

R0 vi (t)

R0 1 vi (t) C1 2R

vo(t)

0 c

C2

(b)

/c

exercise for the student to confirm that the circuits shown in Figure 6.12 have magnitude spectra described by (6.2). Magnitude spectra for low-pass Butterworth filters of various orders are shown in Figure 6.13. It can be seen from the figure that for each value of N, the magnitude spectrum has the same cutoff frequency, vc.

EXAMPLE 6.4

MATLAB program to show frequency response of Butterworth filters The frequency response of normalized Butterworth filters of various orders can be generated by the following MATLAB program:

% This MATLAB program generates a Butterworth filter of % specified order and displays the Bode plot of the % magnitude frequency response.

286

Chap. 6

N=input... ('Specify the order of the filter:') z_p_k='The zeros, poles and multiplying constant.' [z,p,k]=buttap(N), pause num_den='The numerator and denominator coefficients.' [num,den]=zp2tf(z,p,k),pause [mag,phase,w]=bode(num,den); plot(w,mag) title([Magnitude Bode plot ',num2str(N),... 'th order Butterworth filter']) xlabel('omega') ylabel('Magnitude') EXAMPLE 6.5

Design of a second-order Butterworth filter The input signal, v1(t), to the filter network shown in Figure 6.14 is a rectified cosine voltage signal with a peak amplitude of 33.94 V and a frequency of 377 rad/s. Using the results of Example 5.18, we find the half-wave rectified cosine signal to have the frequency spectrum V1(v) = 53.31 a sinc(np/2)[d(v - (n + 1)377) + d(v - (n - 1)377)]. q

q n=-

The frequency spectrum of the rectifier output signal is plotted in Figure 6.15(a). We now design a physically realizable filter to minimize all frequency components except the dc component at v = 0. The filter and load circuit shown in Figure 6.14 will be designed as a second-order Butterworth filter with a cutoff frequency of 100 rad/s. Because we are dealing with a filter that is implemented with a physically realizable electrical circuit, the impedance of the filter will distort the rectified cosine signal if the filter is connected directly to the rectifier circuit. In order to simplify the discussion that follows, we will assume that the rectified cosine signal at the input to the filter circuit is the output of an isolation amplifier, as discussed in Section 2.6 and shown in Figure 2.36. A second-order Butterworth filter has a magnitude frequency response described by H(v) = 1 21 + (v/vc)4 = v2 c 2v4 + v4 c . (6.3)

L v i (t)

Filter

Load

RL

Sec. 6.2

V1() 67.88

Real Filters

287

53.31

22.63

1508 754 377 1 0.9 0.8 0.7 Magnitude 0.6 0.5 0.4 0.3 0.2 0.1 0 0 100 200 300 mag 0.070 at 377 (rad/s) 400 500 377 754

1508 4.53

1.94

(a)

1.08

Frequency (rad/s)

(b)

V0() 67.88

(c)

288

Chap. 6

The frequency response function of the filter circuit is H(v) = 1 1 - v2LC + jvL/R ,

which has the magnitude frequency response H(v) = 1/LC 2 v + (1/LC) + v B (1/RC) R C LC

4 2 2 2

(6.4)

By comparing (6.3) and (6.4), we can see that the RLC filter will match the Butterworth form if we choose vc = If we assume a load resistance 1 A LC and L = 2R2C.

R = 1 k and calculate the inductor and capacitor values to give a cutoff frequency of 100 rad/s, we find that L = 14.14 H and C = 7.07 mF. The frequency response of this filter is plotted in Figure 6.15(b). Figure 6.15(c) shows the magnitude frequency response of the filters output signal, Vo(v) = H(v) V1(v) . It can be seen that non-dc components have been reduced in magnitude by the Butterworth filter, but not completely eliminated as they would be if an ideal filter were available. (See the results in Example 6.1.) I

The following example of an application of a Butterworth filter shows the effect of filtering in both the time and frequency domains:

EXAMPLE 6.6

Butterworth filter simulation Figure 6.16(a) shows a SIMULINK simulation of a simple system. The signal generator block is set to generate a square wave with magnitude of 1 V and fundamental frequency of 100 rad/s. The analog Butterworth filter block is set to simulate a fourth-order low-pass Butterworth filter with a cutoff frequency of 150 rad/s. The filter transfer function is derived by the MATLAB command, butter(N,Wn,s) with N = 4 and Wn = 150.

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