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MITRES 6 008S11 Sol04 PDF

The document discusses the discrete-time Fourier transform (DTFT) and some of its properties. It provides 4 solutions to examples calculating the DTFT of different signals. The key points are: 1) The DTFT relates a discrete-time signal x(n) to its frequency spectrum X(ejw). 2) Solution 4.1 calculates the DTFT of an impulse signal and derives some basic properties. 3) Solution 4.2 calculates the DTFT of the convolution of two signals. 4) Solutions 4.3 and 4.4 provide additional examples calculating the DTFT for shifted and scaled signals.
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0% found this document useful (0 votes)
79 views9 pages

MITRES 6 008S11 Sol04 PDF

The document discusses the discrete-time Fourier transform (DTFT) and some of its properties. It provides 4 solutions to examples calculating the DTFT of different signals. The key points are: 1) The DTFT relates a discrete-time signal x(n) to its frequency spectrum X(ejw). 2) Solution 4.1 calculates the DTFT of an impulse signal and derives some basic properties. 3) Solution 4.2 calculates the DTFT of the convolution of two signals. 4) Solutions 4.3 and 4.4 provide additional examples calculating the DTFT for shifted and scaled signals.
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4

THE DISCRETE-TIME FOURIER TRANSFORM

Solution 4.1

The Fourier transform relation is given by

+C0

X(ejW) = 1

x(n) e-jWn

(a)

thus:

+O0

n=-o

6(n -

X(e") =W

3) e-jwn = e-jw3

n=-o
(b)

1 +

X(e j)

(c) X(e W)

=2

e2 'ejW + 1
2f

an e-jn

=2 (ae jw)n

n=O
+3

n=O

1 - ae

6
n=O

n=-3

7w
e- wn

e-jon = ej3

) =E

(d) X(e

= 1 + cosw

sin(w)

Solution 4.2

(a) In problem 2.4(c) we determined that the convolution of an u(n) and


n u(n) was given by

n+1 _ n+1 u(n)

y(n) =

y(n) =
[n

thus,

(b)

n+l

n+l
a

and

u(n)
n

(n)

From problem 4.1 (c) it follows that

.
H(e 3 ) = 1
1

1
-

a-j

and

X(e j)

-e

S4.1

The Fourier transform of y(n) as obtained in

(a)

00

=2

Y(eS)

n+l _ n+l

e-jwn

n=0

jon

=n

-jwn

n= C,

6 a-

1 -

Be~5"

=(1 -

-a

Se -3w

(1 - ae -

1 -

ae- j

Solution 4.3

+0

+00

Xa(e j)

(a)

kx(n)e-jon

x(n)e-jwn

k X(ej)

x(n

Xb(e)

(b)

k
n=-o

n=-o

n0 ) e-jn

n=-co

Making the substitution of variables

m =

or

n0

Xb (e)

n = m + n0
e-

x(m)

m=-

j(m+n0

e-jon0 x(m)
m=-00

=e-jon 0

X(e jw)

The transform of x(n) is given by

(c)

+0

x(n) e-jwn

X (e w) =
n=-o

thus

dX (ej w)
=
de)
dw

(-jn) x(n) e

n=-o
+C0

or

dX(e j

n x(n)

dw

S4.2

Xc

(e

e-jon

j ton

e-jom

Solution 4.4

(a)

IHa (j)1

-a

e jQ t dt=

h (t)

aeat e-jQ t

0+

|II(jQ)|

Thus

2 + a2

is as sketched below:

IH(jQ) I

Figure

(b)

S4.4-1

hd (n) = cae-anT u(n) = ca(e-aT )n u(n)

thus Hd (e
With

ca
- T e-3
-j
e-aT

jO
) = 1, c

-e-aT
a
a

this choice of c
(1-

IH (eJ ) 12
d

For Hd(e

thus lHd(ejW)I

2e-aT

eaT 2
cos w +

aT

is:

Figure S4.4-2

S4.3

Note in particular that while the frequency response of the


continuous-time filter asymptomatically approaches zero the frequency
response of the digital filter doesn't. However as the sampling period
T decreases, the value of IHd(ejW)| at w = 7 decreases toward zero.
The difference in the minimum values of |Ha(jP)|
of course due to aliasing.

and IHd(e W)I is

Solution 4.5

From the discussion in the lecture, we know that

2:(j
1

A
T

x A (jQ2)

r=-co

and

X(e

+ 2'rr

JW)

W e
X

XA()
XA (j)

IT

Let us assume that XA(j) has some arbitrary shape as indicated below.
Since we are assuming that T is sufficiently small to prevent aliasing,
.
Then XA(jQ) and X(e3W) are as
XA(jQ) must be zero for J|I >
Y(ejW) corresponding to the output of the

filter and YA(jQ) and YA(jQ) follow in a straightforward way and are
as indicated in figure S4.5-1. Thus yA(n) could be obtained directly

by passing x(n) through an ideal lowpass filter with unity gain in

shown in figure S4.5-1.

the passband and a cutoff frequency of 7T rad/sec. For the case in

part (a) the cutoff frequency of the overall continuous-time filter

rad/sec and for the case in part (b) the cutoff frequency
x 10
is

is

S4.4

4
x 10 rad/sec.

XA (j 0)
1

XA
1

-10

7T

7T

XA (e)

Y(e

-2 r

100000

Tr

YA

K>
27T

27T

4
T

K>

K
7T

Tr

2
4 T 4T

-p

YA (j )

7T

7T

Figure S4.5-1
S4.5

Solution 4.6

(1) and (2) can be verified by direct substitution into the inverse

Fourier transform relation.

(3) and (4) follow from (1) since

[x(n))

(5) and (6) follow from (2) since Re[X(ew)]

Re

[x(n)]

(n)j and jIm

[x(n) + x

and jIm [X(e")]

(X (e)

Ix(n) - x

(n).

X(elw) + X (eW)]

X (e~)3

Solution 4.7*

If X(ejW) denotes the Fourier transform of x(n), then

x(O)

X(e j)

dw

-Jr
Thus, with y(n) denoting the convolution of f(n) and g(n) and since
Y(e

) = F(e W) G(e j), we wish to show that


y(O) = f(O) g(Q)
+CO

But

yfn)

f(k)

g(n -

k)

k=-co

so

y(O) =

f(k) g(-k)
k=-o

Since f(k) is zero for k < 0 and g(-k) is zero for k > 0,
g(O)

f(k) g(-k) = f(0)

k=-o
Solution 4.8*

(a)

Method A:

Then

Consider x(n) as a unit-sample 6(n).


g(n)=h(n) and

r(n) = h(n) * g(-n) =


+0

Finally,

k=-o

s(n) = r(-n) =

h(k) h(k + n)

k=-o

h(k) h(k + n)

Consequently, h1 (n) =

k=- 0

S4.6

h(k) h(-n+k)

To show that this corresponds to zero phase, we wish to show that


h (n) = h1 (-n) since from Table 2.1 of the text with h (n), if
h (n) = h1 (-n) then
H (e )W = H
h

(ejW) and hence the frequency response is real.

h(k) h(k -

(-n) =

n)

k=-o

letting k - n = r,

h(n + r) h(r)

h1 (-n) =
k=-o

which is -identical to h (n).

Alternatively we can show that h1 (n) corresponds to a zero-phase

filter by arguing in the frequency domain.

Specifically,

Let

g(n) = g(-n).

Also,

R(e

and

S(ejW)

) = X (e
= R (e

Then

H (e

) = IH(e

) = X (e

G, (e

) H (e W) H(e)

= X (ejW)

) H (e)

IH(e j)12

|H(ejW )12

= X(e W)
Thus,

Ge)

2.
)|

Since H (e )W is real, it has a zero phase

characteristic.
(b)

Method B:
)

G (e

X(e W) H(e )W

R(e W)

X (ej)

H(e )W

Y(e

G (ej)

+ R(e )W

)-=)
-

X(e j)

= X(e j)

[H(ejW) + H*(e

)]

[2 Re H(ejW)]

Therefore H 2 (ejW) = 2 Re H(e

) = 2

IH(e

)I

cos[(arg H(ejW)]) and

consequently is also zero phase.


(c)

H (ejW) and H2(e

) are sketched below.

Clearly method A is the

preferable method.
S4.7

H (eJ])

Figure S4.8-1

(e"')

7
3-7

Figure S4.8-2

S4.8

MIT OpenCourseWare
http://ocw.mit.edu

Resource: Digital Signal Processing


Prof. Alan V. Oppenheim

The following may not correspond to a particular course on MIT OpenCourseWare, but has been
provided by the author as an individual learning resource.

For information about citing these materials or our Terms of Use, visit: http://ocw.mit.edu/terms.

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