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18 Discrete-Time Processing ofContinuous-Time Signals
Solutions to Recommended Problems S18.1
(a) Since x,(t) = xc(t)p(t), then X,(w) is just a replication of Xe(w) centered at multiples of the sampling frequency, namely 8 kHz or 27r8 X 10' rad/s. The sampling period is T = 1/8000.
X,(W)
8000T
-21r X 8000 21r X 4000 27n X 8000
Figure S18.1-1
(b) X(Q) is just a rescaling of the frequency axis, where 21r8 X 103 becomes 2 1r. X(Q) is shown in Figure S18.1-2.
(c) Y(Q) is the product G(Q)X(Q). Therefore, Y(Q) appears as in Figure S18.1-3.
Y(2)
8000
6000
-2ff --1
4 T
u 48
Figure S18.1-3
2f
S18-1
Signals and Systems S18-2
(d) Y,(w) is a frequency-scaled version of Y(w) but only in the range 0 = --w to 7, as shown in Figure S18.1-4. Also note the gain of T.
Ye(W)
_ - 8000 -8000 r S 4
Figure S18.1-4
S18.2
(a) The maximum nonzero frequency component of H(w) is 5 00w. Therefore, this frequency can correspond to, at most, the maximum digital frequency before folding, i.e., Q = x. From the relation wT = Q,we get
T. = - =2 ms500w
(b) Since w = 500w maps to Q = 7r, the discrete-time filter G(Q) is as shown in FigureS18.2-1.
G
_' .- .04
Figure S18.2-1
(c) The complete system is given by Figure S18.2-2. Note the need for an anti-aliasing filter.
Discrete-Time Processing of Continuous-Time Signals / Solutions S18-3
S18.3
(a) Recall that Xc(w) is as given by Figure S18.3-1.
Xc(w)
7r IT
-10000r 100007T
Figure S18.3-1
X(Q) is given by eq. (S18.3-1) and Figure S18.3-2.
X(Q) 1 E
00 27rn) = 20000 Xc[20000(9 - 27rn)] (S18.3-1)= -
Tn = -oo
Ye(w) is given by eq. (S18.3-2) and Figure S18.3-3.
cI(W)
TX(wT), IwI < T
elsewhere (S18.3-2)
Thus x(t) = y(t) in this case.
Signals and Systems S18-4
(b) Xc(w) is as given in Figure S18.3-4.
Xc(w)
co -54000r 540007r
Figure S18.3-4
We now use eq. (S18.3-1), shown in Figure S18.3-5.
2000OXc(20000(2 - 2n)) n = 0
200007r
54T20
n = - 1
n = + 1
iT 27r
Figure S18.3-5
Thus, in the range ± r, X(Q) = 20000 E",,U X,[20000(9 - 2irn)] is given as in Figure S18.3-6.
X(2 )
200007 -
F41T ISi4gr20 20
Figure S18.3-6
Discrete-Time Processing of Continuous-Time Signals / Solutions S18-5
Using eq. (S18.3-2), we find Ye(w) as in Figure S18.3-7.
YJ(w)
-14000n 14000T Figure S18.3-7
Note aliasing since 27000 Hz is above half the sampling rate of 20000 Hz.
(c) Xe(w) is as given in Figure S18.3-8.
Xc(w)
74 CtT -34000ir 340007r
Figure S18.3-8
Again we use eq. (S18.3-1), shown in Figure S18.3-9.
Signals and Systems S18-6
Thus X(Q) is given as in Figure S18.3-10.
Finally, from eq. (S18.3-2) we have Y,(w) shown in Figure S18.3-11.
Yc(w)
-6000i 600r
Figure S18.3-11
S18.4
It is required that we sample at a rate such that the discrete-time frequency 7r/ 2 will correspond to c. The relation between 9, and cis Q, = ocTo. Thus, we require
= TO W"C
As wc increases, demanding a wider filter, To decreases, and consequently the sampling frequency must be increased. There are two ways to calculate Wa. First, since we are sampling at a rate of
or = 4(c(ir/2)/(Ac '
we need an anti-aliasing filter that will remove power at frequencies higher than half the sampling rate; therefore wa = 2wc. Alternatively, we note that the "folding frequency," or the frequency at which aliasing begins, is 0 = 7r. Since 9 = r/2 corresponds to wc, then r must correspond to 2w,.
Discrete-Time Processing of Continuous-Time Signals / Solutions S18-7
S18.5
(a) We sketch X(Q) by stretching the frequency axis so that 2 7 corresponds to the sampling frequency with a gain of 1/TO. We then repeat the spectrum, as shown in Figure S18.5-1.
X(O2) 1
7T 7T
Figure S18.5-1
After filtering, Y(Q) is given as in Figure S18.5-2.
Y(E2)
TO
3~ 3
Figure S18.5-2
(b) We see that Y(Q) looks like X(w) filtered and then sampled. The discrete-time frequency is -/3. Again, 2-x corresponds to 27r/To, so 7/3 corresponds to ir/3T. Thus, if x(t) is filtered by G(w) as given in Figure S18.5-3, then y[n] = z[n].
G(c)
3 To 3To
Figure S18.5-3
Signals and Systems S18-8
Solutions to Optional Problems S18.6
(a) Since we are allowing all frequencies less than 1007r through the anti-aliasing filter, we need to sample at least twice 100r, or 200r. Thus, 2 007r = 2-x/T or To = 10 ms. To find K, recall that impulse sampling introduces a gain of 1/TO. To account for this, K must equal To, or K = 0.01.
(b) (i) Since X(w) is bandlimited to 100-r, the anti-aliasing filter has no effect. The Fourier transform of x,(t), the modulated pulse train, is given in Figure S18.6-1.
Xp(2)
1 - 00 To
-4007 -1007T 1007 4007
Figure S18.6-1
Since To = 0.005, the sampling frequency is 4007r. After conversion to a discrete-time signal, X(Q) appears as in Figure S18.6-2.
X(W)
200
-2w7r 2w
2 2
Figure S18.6-2
After filtering, Y(Q) is given by Figure S18.6-3.
Y(n)
200
3 3~
Figure S18.6-3
Discrete-Time Processing of Continuous-Time Signals / Solutions S18-9
(ii) There are three effects to note in D/C conversion: (1) a gain of To, (2) a frequency scaling by a factor of T,, and (3) the removal of repeated spectra. Thus, Y(w) is as shown in Figure S18.6-4.
Y(W)
IT IT
3To 3To
Figure S18.6-4
S18.7
After the initial shock, you should realize that this problem is not as difficult as it seems. If instead of h[n] we had been given the frequency response H(Q), then He(w) would be just a scaled version of H(Q) bandlimited to ir/T.Let us find, then, H(Q). Using properties of the Fourier transform, we have
1 Y(Q) =- ei"YG) + X(Q)
2
Y(Q) 1 H(0) X(Q) 1 - ie-j"
Thus,
1IH(Q)I =
\ - cos 0
( sin g -<H() = -tan' 1 - cos
Therefore, the magnitude and phase of He(c) are as shown in Figure S18.7.
1
He(w)| = \/ -coswT' T '
0, elsewhere
(1 isinwTI,-tan-' - icoswT T
0, elsewhere
Signals and Systems S18-10
S18.8
x [n] --x
IHc(w) I
r iT T T
4Hc(w)
/00000* I /.A
7T T V 7V
Figure S18.7
The system under study is shown in Figure S18.8.
covrsot p(t) T [ ) x:c(t)W yc~tW ypW ot) sont en annipleConversionimpulse -- +ahst)uXnca to T[n e -qr y [n]
T T
( 8(t-nT) n=
Figure S18.8
From our previous study, we know that Xe(o) in the range ±ir/Tlooks just like X(Q) in the range ±ir. Similarly, Y,(w) between -- r/T and +r/T looks like Y(Q) in the range -ir to r. Although there is a factor of T, we can disregard it in analyzing this system because it is accounted for in the H(w) filter. The transformation of xc(t) to yc(t) will correspond to filtering x[n], yielding y[n]. In fact, the equivalent system will have a system function H(Q) given by
H(Q) = He , |u| < r,
where He(w) is the Fourier transform of h(t). Thus, we need to find Hc(w). The relation between yc(t) and xc(t) is governed by the following differential equation:
d td +y4t+ dyc +~ 3yc(tt)=t)) =4'( dyt xc(t )
Discrete-Time Processing of Continuous-Time Signals / Solutions S18-11
Using the properties of the Fourier transform, we have
(j) 2 Ye(w) + 4(j)Ye(w) + 3Y0(w) = Xco), 1
He(w) =1 (jco)2 + 4jw + 3
Therefore,
H(Q) 2 + 4j + Jul <K + 4j- + 3
S18.9
(a) It is instructive to sketch a typical y,(t), which we have done in Figure S18.9-1.
Let us suppose that T is changed by being reduced. Then the envelope of y,(t) seems to correspond to a higher-frequency cosine. At time kt,
2wk 2w(kT) 2wty,(t) = cos N- b(t NT NT- kT) = cos NT 6(t - kT) = cos - 6(t - kT),
where we use the sampling property of the impulse function. Thus,
y,(t) = Y cos (t - kT) = cos wot ( ot - kT), k= -w0 k=-w0
where wo = 21/NT. If the minimum wo is wi, and since T = 21/N(o,
2T Tmax=
A) I
Similarly,
= T.in NW2
Signals and Systems S18-12
(b) Recall that sampling with an impulse train repeats the spectrum with a period of 2r/T and a gain factor of 1/T. Since 5([cos(2t/NT)J is as given by FigureS18.9-2, Y,(w) is then given by Figure S18.9-3.
7T 7T
2,n 27r NT NT
Figure S18.9-2
T - NT T T
7rr 2n Nr i
2| r 27r 27/N - 1 ~-WO WO= 2rN - 1\ 21r
T T N T N T
Figure S18.9-3
(c) The minimum value of N is 2, corresponding to the impulses at o and (27r/T wo) being superimposed at w/T. The lowpass filter cutoff frequency must be such that the (superimposed) impulses at 7r/T are in the passband and those at 37r/T are outside the passband. Consequently,
ir 3T T
(d) Comparing Y(w) and Y,(w) in Figures S18.9-2 and S18.9-3 respectively, we see that for N > 2 the cosine output will have an amplitude of 1/T = w/21r. If N = 2, then the output amplitude will be 2/T = w/7r.
S18.10
(a) By sampling sc(t), we get
s[n] = setnT) = x(nT) + ax(nT - TO) = x(nT) + ax[(n - 1)T]
since T = To. Let x[n] = x(nT). Then
s[n] = x[n] + ax[n - 1]
Therefore
x[n] = -ax[n - 11 + s[n]
Discrete-Time Processing of Continuous-Time Signals / Solutions S18-13
This is a first-order difference equation, so given s[n], we can find x[n]. Since
x(t) is appropriately bandlimited, we can then set
y[n] = -ay[n - 1] + s[n]
which will make
A yc(t) = -x(t)
T
(b) From part (a) we see that T = A will make y(t) = x(t).
(c) Since we do not want to alias, we still need T < 7r/wM. Now
s(t) = x(t) + ax(t - TO)
Taking the continuousFourier transform, we see that
S(w) = X(w) + ae -iwTox(w)
Thus, the continuous-time inverse system has frequency response
1 1 + ae ~-T0
We want to implement this in discrete time. Therefore, using the relation, we
obtain
-
H(Q) = H, (Q) a 1 1 <M <A =i1 + aef jQ(T 0 /T)' ileM s bM
Again, the filter should be A = T.
MIT OpenCourseWare http://ocw.mit.edu
Resource: Signals and Systems Professor Alan V. Oppenheim
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