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Applications

Fall 2011

Final Exam Solutions

1. (15 points) Recall that the Fourier transform of cos 2at is (1/2)(

a

+

a

). Find the

Fourier transform of the following modied cosine signals.

1.0 0.5 0.0 0.5 1.0

1.5

1.0

0.5

0.0

0.5

1.0

1.5

f

1

2 1 0 1 2

1.5

1.0

0.5

0.0

0.5

1.0

1.5

t

f

2

4 2 0 2 4

1.5

1.0

0.5

0.0

0.5

1.0

1.5

t

f

3

1

Solution:

(a)

F{sgn(t) cos(4t)} = F{sgn(t)} F{cos(4t)}

=

1

is

1

2

[

2

+

2

]

=

1

2i

_

1

s + 2

+

1

s 2

_

(b)

F{H(t) cos(4t)} = F{H(t)} F{cos(4t)}

=

1

2

[ +

1

is

]

1

2

[

2

+

2

]

=

1

4

[

2

+

2

] +

1

4i

_

1

s + 2

+

1

s 2

_

(c)

F{(t) cos(4t) III

2

} = sinc (s)

1

2

[

2

+

2

]

1

2

III

1/2

=

1

4

[sinc (s + 2) + sinc (s 2)] III

1/2

=

1

4

k=

_

sinc

_

k

2

+ 2

_

+ sinc

_

k

2

2

_

_

_

s

k

2

_

2

2. (15 points) Filtering for interpolation

Suppose you have sampled a signal f(t) at intervals of one unit to obtain f

sampled

(t),

shown below:

The arrows represent -functions of dierent strengths at 0, 1, and 2.

Sketch the following interpolations of f(t):

(a) f

1

(t) = F

1

{sinc (s) Ff

sampled

(s)}

(b) f

2

(t) = F

1

{e

is

sinc (s) Ff

sampled

(s)}

(c) f

3

(t) = F

1

{sinc

2

(s) Ff

sampled

(s)}

Solution:

Each of these formulas can be simplied by the convolution theorem:

(a) f

1

(t) = F

1

{sinc (s) Ff

sampled

(s)} = (t) f

sampled

(t)

(b) f

2

(t) = F

1

{e

is

sinc (s) Ff

sampled

(s)} = (t 1/2) f

sampled

(t)

(c) f

3

(t) = F

1

{sinc

2

(s) Ff

sampled

(s)} = (t) f

sampled

(t)

Here are the plots:

3

(a) Nearest neighbor (b) Zero order hold

(c) Linear interpolation

We nd that these lters correspond to common interpolation methods, nearest neigh-

bor, zero-order hold, and linear interpolation.

4

3. (20 points) The Poisson distribution (in probability) is used to model the probability

of a given number of events occurring over a period of time. For example, the num-

ber of phone calls arriving at a call center per minute is modeled using the Poisson

distribution:

Prob(Number of calls = k) =

k

k!

e

, k = 0, 1, 2, . . . ,

where is the number of phone calls per minute. Now, the rate of arrivals of phone

calls typically varies throughout the day, so itself should be considered to come from

a random variable, , say with probability density function p(). In that case, the

probability Prob(Number of calls = k) is more realistically modeled by the function

P(k) =

_

0

k

k!

e

p() d, k = 0, 1, 2, . . . .

We will call P(k) the Poisson transform of p().

It only makes sense to consider 0, but it is convenient for the discussion below to

allow < 0 by declaring p() = 0 when < 0. Then we can write

P(k) =

_

k

k!

e

p() d, k = 0, 1, 2, . . . .

Note that unlike the Fourier transform, the Poisson transform takes a function of a

real variable and produces a function dened on the natural numbers 0, 1, 2, . . . .

(a) We would like to know that the Poisson transform is invertible, i.e., given the num-

bers P(0), P(1), P(2), . . . it is possible to nd p(). This is a relevant question, as

the values P(k) can be measured experimentally from a large set of observations.

For this, given the values P(k) dene the function

Q(s) =

k=0

(2is)

k

P(k).

Using the denition of P(k) show that Q(s) is the Fourier transform of q() =

e

Hint: Recall the Taylor series for the exponential function

e

a

=

k=0

a

k

k!

.

(b) Suppose there are two independent factors contributing to the arrival rate of

phone calls, so that =

1

+

2

. Recall for the corresponding probability density

functions we have

p = p

1

p

2

.

We want to nd the relationship between the Poisson transforms, P(k) for p, and

P

1

(k), P

2

(k) for p

1

, p

2

. (Problem continues on next page.)

5

i. As above, let q() = e

1

() = e

p

1

(), q

2

() =

e

p

2

(). Show that

q

1

q

2

= q (use the denition of convolution).

ii. Using Q(s) = Q

1

(s)Q

2

(s) for the Fourier transforms, deduce that

P(k) =

k

m=0

P

1

(m)P

2

(k m),

a discrete convolution of P

1

and P

2

!

Solutions:

For the rst part, we have

Q(s) =

k=0

(2is)

k

P(k)

=

k=0

(2is)

k

_

k

k!

e

p() d

=

_

k=0

(2is)

k

k

k!

_

e

p() d

=

_

e

2is

e

p() d

= F(e

p())

Hence

p() = e

F

1

Q().

For the second part, with q

1

() = e

p

1

(), q

2

() = e

p

2

() we have

(q

1

q

2

)() =

_

q

1

(t)q

2

( t) dt

=

_

e

t

p

1

(t)e

(t)

p

2

( t) dt

=

_

p

1

(t)p

2

( t) dt

= e

p

1

(t)p

2

( t) dt

= e

(p

1

p

2

)()

= q()

6

Next, writing out Q(s) = Q

1

(s)Q

2

(s) we have

k=0

(2is)

k

P(k) =

_

m=0

(2is)

m

P

1

(m)

__

n=0

(2is)

n

P

2

(n)

_

=

m,n=0

(2is)

m+n

P

1

(m)P

2

(n)

Equating like powers of 2is we must have

(2is)

k

P(k) =

m,n=0,m+n=k

(2is)

m+n

P

1

(m)P

2

(n)

=

m,n=0,m+n=k

(2is)

k

P

1

(m)P

2

(n)

= (2is)

k

m,n=0,m+n=k

P

1

(m)P

2

(n),

or

P(k) =

m,n=0,m+n=k

P

1

(m)P

2

(n)

=

k

m=0

P

1

(m)P

2

(k m)

There can be only a nite number of terms in the sum, from 0 to k, since m and n add

to k.

7

4. (20 points) Oversampling

Let f(t) be a bandlimited signal with spectrum contained in the interval 1/2 < s <

1/2. Suppose you sample f(t) at intervals of 1/2 (that is, at twice the Nyquist rate),

to obtain

f

sampled

(t) =

1

2

III

1/2

(t) f(t).

(a) (5 points) Qualitatively explain why the following equation is correct:

f(t) = F

1

{K(s) Ff

sampled

(s)}

where K(s) is dened by

(b) (10 points) Show that you can reconstruct f(t) by

f(t) =

1

2

n=

f

_

n

2

_

k

_

t

n

2

_

where k(y) =

cos(y) cos(2y)

2

y

2

.

You may use the fact that

K(s) = F

_

cos(t) cos(2t)

2

t

2

_

.

(c) (5 points) Describe an advantage the reconstruction formula in part (b) has over

the usual sinc interpolation formula, say in terms of the accuracy of the series if

one only uses a nite number of terms.

Solutions:

(a) We use the properties of the shah distribution to calculate

Ff

sampled

(s) = F(s) III

2

(s).

8

This means the spectrum is periodized with period 2. As the spectrum is ban-

dlimited to the interval (1/2, 1/2), the rst sideband is centered around 2 and

contained in (3/2, 5/2). As nothing exists in (1/2, 3/2), multiplying the periodized

spectrum by K(s) is like multiplying by the rectangle function: it retains only

the central copy.

(b) First, we rewrite f

sampled

(t) as

f

sampled

(t) =

1

2

III

1/2

(t) f(t) =

1

2

n=

f

_

n

2

_

_

t

n

2

_

.

Substituting this formula into the equation from part (a) and simplifying, we nd

f(t) = F

1

{K(s) Ff

sampled

(s)}

= F

1

_

K(s) F

n=

1

2

f

_

n

2

_

_

t

n

2

_

_

= F

1

K(s)

n=

1

2

f

_

n

2

_

_

t

n

2

_

using the convolution theorem

=

_

cos(t) cos(2t)

2

t

2

_

n=

1

2

f

_

n

2

_

_

t

n

2

_

using the given formula

=

1

2

n=

f

_

n

2

_

cos (t n/2) cos 2(t n/2)

2

(t n/2)

2

using the convolution property

(c) Its natural to expect the formula to have a smaller error because it makes use

of twice as many measurements. Moreover, the oversampling formula converges

faster than the sinc interpolation formula because the kernel k(y) goes as 1/y

2

whereas sinc (y) goes as 1/|y| for large N. Put another way, the tail of the series,

after a nite number of terms, is smaller for the oversampling formula than it is

for sinc interpolation.

9

5. (10 points) Sampling a periodic signal: Another interpretation of the DFT

Let x(t) be a periodic signal of period 1. Let N > 1 and sample x(t) by multiplying

by a nite III function of spacing 1/N, forming

g(t) = x(t)

N1

k=0

(t k/N).

Let c

n

be the nth Fourier coecient,

c

n

=

_

1/2

1/2

e

2int

g(t) dt =

_

1

0

e

2int

g(t) dt.

(The integral from 1/2 to 1/2 is more convenient to use for the calculations in this

problem.)

Finally, let f be the discrete signal with values at the sample points,

f[n] = x(n/N), n = 0, 1, . . . , N 1.

What is the relationship between c

n

and the DFT of f[n]?

Figure 1: Sampling of a periodic signal x(t).

10

Solutions:

c

n

=

_

1/2

1/2

g(t)e

2int

dt

=

_

1/2

1/2

_

x(t)

N1

k=0

(t k/N)

_

e

2int

dt

=

_

1/2

1/2

_

N1

k=0

x(t)(t k/N)

_

e

2int

dt

=

N1

k=0

_

1/2

1/2

x(t)e

2int

(t k/N) dt

(its the same as if we integrated from to )

=

N1

k=0

x(k/n)e

2ikn/N

=

N1

k=0

f[k]e

2ikn/N

= Ff[n].

Therefore, c

n

= Ff[n]

11

6. (20 points) Time domain multiplexing

Let f, g, h be N-dimensional vectors with discrete Fourier transforms F, G, H respec-

tively.

(a) (10 points) We combine f and g to form the 2N-dimensional signal w,

w[n] =

_

f[n/2], for n even

g[(n 1)/2], for n odd,

where 0 n 2N 1. Find the discrete Fourier transform W = Fw in terms of

F and G.

Hint: Consider constructing w from upsampled versions of f and g.

(b) (5 points) How does your answer to part (a) simplify if g = 0? Explain why this

makes sense given what you know about replication and upsampling.

(c) (5 points) Suppose we dened w[n] for 0 < n < 3N 1 by

w[n] =

_

_

_

f[n/3], for n = 0 mod 3

g[(n 1)/3], for n = 1 mod 3

h[(n 2)/3], for n = 2 mod 3

Describe how you expect W depends on F, G, and H. You do not need to do any

calculations for this part; you may generalize from part (a). To keep everything

simple and concrete, set N = 4 for this part.

Solution:

(a) First consider an upsampled version of f:

f

up

= (f[0], 0, f[1], 0, . . . , f[N 1], 0)

The DFT of f

up

can be found using calculations similar to those on PS5, Question

3. For 0 m < N, we have

Ff

up

[m] =

2N1

n=0

f

up

[n] e

2i

nm

2N

=

N1

n=0

f

up

[2n] e

2i

(2n)m

2N

(summing over nonzero terms only)

=

N1

n=0

f[n] e

2i

nm

N

= Ff[m]

12

For N m < 2N, we have

Ff

up

[m] =

2N1

n=0

f

up

[n] e

2i

n(mN)

2N

e

2i

nN

2N

=

N1

n=0

f

up

[2n] e

2i

(2n)(mN)

2N

e

2i

2n

2

=

N1

n=0

f[n] e

2i

n(mN)

N

= Ff[mN]

So we have

Ff

up

[m] =

_

F[m], for 0 m < N

F[mN], for N m < 2N 1

Now consider an upsampled version of g:

g

up

= (0, g[0], 0, g[1], 0, . . . , 0, g[N 1])

Using the discrete shift theorem and our result for Ff, we have

Fg

up

[m] =

_

G[m] e

im/2N

, for 0 m < N

G[mN] e

im/2N

, for N m < 2N 1

Now we note that w is the sum

w = f

up

+ g

up

so the DFT of w is

Fw[m] =

_

F[m] + G[m]e

im/2N

, for 0 m < N

F[mN] + G[mN]e

im/2N

, for N m < 2N 1

13

(b) Setting g = 0,

Fw[m] =

_

F[m], for 0 m < N

F[mN], for N m < 2N 1

which is exactly the upsampling-replication duality we saw on Problem Set 7.

(c) First set e

i/3N

= . Next, we deduce from part (a) that G will be modulated by

m

and that H will be modulated by

2m

. Putting these ideas together, we

conclude

Fw[m] =

_

_

_

F[m] +

m

G[m] +

2m

H[m], for 0 m < 4

F[m4] +

m

G[m4] +

2m

H[m4], for 5 m < 8

F[m8] +

m

G[m8] +

2m

H[m8], for 9 m < 12

14

7. (20 points) LTI input/output pairs

Assuming that L is an LTI system, answer the following questions. Each part is inde-

pendent of the others, i.e. the system L in (a) is dierent from the system L in (b).

(a) If L{cos(2t)} = y(t), nd L{sin(2t)} in terms of y(t).

(b) If L{(t)} = y(t), nd L{(t/2)} in terms of y(t).

(c) If L{sinc (t)} = sin(

2

3

t), nd L{sinc (t/2)}.

(d) If L{(t)} = cos(t), nd L{(t)}.

Hint: For part (a), think rst how to relate sin(2t) to cos(2t), and then use that

relationship to determine the unknown output. In fact, most of these problems can be

solved by nding an appropriate connection between the two given inputs.

15

Solution:

(a) The two inputs to the system are related by a shift in time:

sin(2t) = cos(2(t 1/4))

By time-invariance, the response to the sine must be y(t 1/4) or y(t1/4+n),

where n is an integer. Other answers are also possible:

L{sin(2t)} =

1

2

L{

d

dt

cos(2t)} =

1

2

d

dt

L{cos(2t)} =

y

(t)

2

(b) The two inputs are related by

(t/2) =

2

(t) = (t 1/2) + (t + 1/2).

Using linearity and time invariance, the output is y(t + 1/2) + y(t 1/2) .

(c) In the frequency domain, the input (s) produces responses at s = 1/3. So we

can say that the only frequencies in the range |s| < 1/2 that produce a nonzero

response are at s = 1/3; the other frequencies in that range are zeroed out. The

input (2s) only has frequencies in the interval |s| < 1/4, so the output must be

zero .

(d) First note that = , so the question is asking us to nd

L{(t)} = L{ } = h ( )

where h is the impulse response of the system. Since convolution is associative,

this is the same as

h ( ) = (h ) = L{} = cos(t) (t)

We also accept any equivalent answer, such as

(t)cos(t) = F

1

_

sinc(s)

1

2

[(s 1/2) + (s + 1/2)]

_

(t) = sinc(1/2) cos(t)

16

8. (20 points) The Fourier transform of a 2D function

The pictures below show a two dimensional function f(x) and the magnitude of its

spectrum, |F(s)|. The spectrum was obtained by Matlabs two dimensional DFT

routine. The images are normalized so that white pixels correspond to one and black

pixels to zero.

f(x)

|F(s)|

On the following page are magnitude plots of modied versions of f(x). For each signal

(1-4), indicate the corresponding Fourier transform magnitude plot (a-e), shown on the

subsequent page. Provide a brief explanation as to your reasoning.

17

(1)

(2)

(3) (4)

18

(a) (b)

(c) (d)

(e)

19

Solution:

(1) (d): Periodizing in the x

2

direction corresponds to sampling in s

2

direction.

Or, downsampling in s

2

direction results in aliases along the x

2

direction.

(2) (c): Periodizing in the x

1

direction corresponds to sampling in s

1

direction.

Or, downsampling in s

1

direction results in aliases along the x

1

direction.

(3) (e): Downsampling along the x

2

direction results in aliases in the s

2

direction.

(4) (a): Rotating in space results in an identical rotation in frequency.

20

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