EL 713: Digital Signal Processing Extra Problem Solutions

1.11 Consider the following 9-point signals, 0 n 8.

(a) [3, 2, 1, 0, 0, 0, 0, 2, 1]
(b) [3, 2, 1, 0, 0, 0, 0, 2, 1]
(c) [3, 2, 1, 0, 0, 0, 0, 2, 1]
(d) [0, 2, 1, 0, 0, 0, 0, 2, 1]
(e) [0, 2, 1, 0, 0, 0, 0, 2, 1]
(f) [3, 2, 1, 0, 0, 0, 0, 1, 2]
(g) [3, 2, 1, 0, 0, 0, 0, 1, 2]
(h) [0, 2, 1, 0, 0, 0, 0, 1, 2]
(i) [0, 2, 1, 0, 0, 0, 0, 1, 2]
Which of these signals have a real-valued 9-point DFT? Which of these signals have an imaginary-
valued 9-point DFT? Do not use MATLAB or any computer to solve this problem and do not explicitly
compute the DFT; instead use the properties of the DFT.
Solution:
Signals (f ) and (i) both have purely real-valued DFT. Signal (h) has a purly imaginary-valued DFT.

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1.12 Matching. Match each discrete-time signal with its DFT by filling out the following table. You should
be able to do this problem with out using a computer.

Signal DFT
1
2
3
4
5
6
7
8

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SIGNAL 1 SIGNAL 2
1.5 1.5
1 1
0.5 0.5
0 0
0.5 0.5
1 1
1.5 1.5
0 10 20 30 0 10 20 30

SIGNAL 3 SIGNAL 4
1.5 1.5
1 1
0.5 0.5
0 0
0.5 0.5
1 1
1.5 1.5
0 10 20 30 0 10 20 30

SIGNAL 5 SIGNAL 6
1.5 1.5
1 1
0.5 0.5
0 0
0.5 0.5
1 1
1.5 1.5
0 10 20 30 0 10 20 30

SIGNAL 7 SIGNAL 8
1.5 1.5
1 1
0.5 0.5
0 0
0.5 0.5
1 1
1.5 1.5
0 10 20 30 0 10 20 30

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DFT 1 DFT 2
30 30

20 20

10 10

0 0
0 10 20 30 0 10 20 30

DFT 3 DFT 4
30 30

20 20

10 10

0 0
0 10 20 30 0 10 20 30

DFT 5 DFT 6
30 30

20 20

10 10

0 0
0 10 20 30 0 10 20 30

DFT 7 DFT 8
30 30

20 20

10 10

0 0
0 10 20 30 0 10 20 30

Solution:
Signal 1 has exactly two cycles of a cosine, so you would expect X(2) and X(2) to be nonzero, and
other DFT coefficients to be 0; that gives DFT 4. Note that X(2) is really X(N 2).
Signal 2 has two and a half cycles of a cosine, so you would expect the DFT to have a peak at index
k = 2.5, but that is not an integer there is no DFT coefficient at that index. So the largest DFT
coefficients would be at k = 2 and k = 3 and there would be leakage. There would also be a peak

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EL 713: Digital Signal Processing Extra Problem Solutions

around k = N 2.5. This gives DFT 6.

Similar reasons are used for signals 3 and 4.
The DFT of a constant is an impulse, so signal 6 corresponds to DFT 7. The DFT of an impulse is a
constant, so signal 7 corresponds to DFT 3.
The DTFT of a rectangular pulse is a digital sinc function, so the DFT of a rectangular pulse is samples
of the sinc function. So signal 8 corresponds to DFT 5.
That leaves signal 5 and DFT 8. Signal 5 can be written as a cosine times a rectangular pulse, so the
DFT of signal 5 will be the convolution of a DFT of a cosine with the DFT of rectangular pulse that
is a sum of two shifted digital sinc functions.

Signal DFT
1 4
2 6
3 1
4 2
5 8
6 7
7 3
8 5

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EL 713: Digital Signal Processing Extra Problem Solutions

1.25 The analog signal x(t) is band-limited to 40 Hz. Suppose the signal is sampled at the rate of 100
samples per second and that at this rate 200 samples are collected. Then 200 zeros are appended to
the 200 samples to form a 400-point vector. Then the 400-point DFT of this vector is computed to get
X(k) for 0 k 399.
(a) Which DFT coefficients are free of aliasing?
(b) The DFT coefficient X(50) represents the spectrum of the analog signal at what frequency f ?
(Give your answer in Hz).
Solution:

(a) All of the DFT coefficients are free of aliasing. The sampling rate is more that twice the maximum
signal frequency.
(b) The DFT bin width is 100/400 or 0.25 Hz. The 50th DFT coefficient corresponds to the frequency
50 times 0.25 Hz or 12.5 Hz .

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3.3 Matching. The diagrams on the following three pages show the impulse responses, pole-zero diagrams,
and frequency responses magnitudes of 8 discrete-time causal LTI systems. But the diagrams are out
of order. Match each diagram by filling out the following table.

Impulse response Pole-zero Frequency response

1
2
3
4
5
6
7
8

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IMPULSE RESPONSE 1 IMPULSE RESPONSE 2

1 1

0.5 0.5

0 0

0.5 0.5

1 1

0 5 10 15 20 25 0 5 10 15 20 25

IMPULSE RESPONSE 3 IMPULSE RESPONSE 4

1 1

0.5 0.5

0 0

0.5 0.5

1 1

0 5 10 15 20 25 0 5 10 15 20 25

IMPULSE RESPONSE 5 IMPULSE RESPONSE 6

1 1

0.5 0.5

0 0

0.5 0.5

1 1

0 5 10 15 20 25 0 5 10 15 20 25

IMPULSE RESPONSE 7 IMPULSE RESPONSE 8

1 1

0.5 0.5

0 0

0.5 0.5

1 1

0 5 10 15 20 25 0 5 10 15 20 25

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POLEZERO DIAGRAM 1 POLEZERO DIAGRAM 2

1 1

imag part 0.5 0.5

imag part
0
6 0
2

0.5 0.5

1 1
1.5 1 0.5 0 0.5 1 1.5 1.5 1 0.5 0 0.5 1 1.5
real part real part

POLEZERO DIAGRAM 3 POLEZERO DIAGRAM 4

1 1

0.5 0.5
imag part

imag part
0 0

0.5 0.5

1 1
1.5 1 0.5 0 0.5 1 1.5 1.5 1 0.5 0 0.5 1 1.5
real part real part

POLEZERO DIAGRAM 5 POLEZERO DIAGRAM 6

1 1

0.5 0.5
imag part

imag part

0 0
2

0.5 0.5

1 1
1.5 1 0.5 0 0.5 1 1.5 1.5 1 0.5 0 0.5 1 1.5
real part real part

POLEZERO DIAGRAM 7 POLEZERO DIAGRAM 8

1 1

0.5 0.5
imag part

imag part

0
2 0

0.5 0.5

1 1
1.5 1 0.5 0 0.5 1 1.5 1.5 1 0.5 0 0.5 1 1.5
real part real part

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FREQUENCY RESPONSE 1 FREQUENCY RESPONSE 2

1.4 7

1.2 6

1 5

0.8 4

0.6 3

0.4 2

0.2 1

0 0
1 0.5 0 0.5 1 1 0.5 0 0.5 1
/ /
FREQUENCY RESPONSE 3 FREQUENCY RESPONSE 4
6 3.5

5 3

2.5
4
2
3
1.5
2
1

1 0.5

0 0
1 0.5 0 0.5 1 1 0.5 0 0.5 1
/ /
FREQUENCY RESPONSE 5 FREQUENCY RESPONSE 6
3.5 20

3
15
2.5

2 10

1.5
5
1

0.5 0
1 0.5 0 0.5 1 1 0.5 0 0.5 1
/ /
FREQUENCY RESPONSE 7 FREQUENCY RESPONSE 8
3.5 7

3 6

5
2.5
4
2
3
1.5
2

1 1

0.5 0
1 0.5 0 0.5 1 1 0.5 0 0.5 1
/ /

Solution:

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Impulse response Pole-zero Frequency response

1 6 8
2 3 4
3 8 7
4 4 5
5 7 3
6 1 2
7 2 6
8 5 1

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EL 713: Digital Signal Processing Extra Problem Solutions

3.6 An FIR digital filter has the transfer function

H(z) = (1 z 1 )3 (1 + z 1 )3

(a) Sketch the pole-zero diagram of this system.

(b) Sketch |H f ()|.
(c) Would you classify this as a low-pass, high-pass, band-pass, or band-stop filter? Please briefly
explain.
Solution:
Note that because the zero at z = 1 is of third order, not only is H f ( = 0) equal to one, but so is its
first and second derivative, so the frequency response is flat at = 0. The same is true for = .

0.5
Imaginary Part

3 6 3
0

0.5

1
1 0.5 0 0.5 1
Real Part

|H()|
10

0
1 0.5 0 0.5 1
omega/

4 Linear-Phase FIR Digital Filters

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4.5 For the transfer function

H(z) = z 1 + z 6

of an FIR linear-phase filter,

(a) sketch the impulse response
(b) what is the type of the filter (I, II, III, or IV)?
(c) sketch the frequency response magnitude |H f ()|.
(d) sketch the zero diagram
Solution:
This is a Type 2 FIR filter.
To find the zeros of H(z),

z 1 + z 6 = 0 (17)
5
z +1=1 (18)
5
z = 1 (19)
z 5 = ej (20)
5 j +j 2 k
z =e (21)
j /5+j (2 /5) k
z=e (22)

which for different integer values of k gives the values z = ej /5 , z = ej 3 /5 , z = ej 5 /5 = 1,

z = ej 7 /5 , z = ej 9 /5 , and which are shown in the zero diagram.

IMPULSE RESPONSE ZEROS OF H(z)

1.5
1
1
Imaginary Part

0.5
0.5
5
0 0
0.5
0.5
1
1
1.5
0 2 4 6 8 1 0 1
Real Part
|H()|
2

1.5

0.5

0
4/5 3/5 2/5 /5 0 /5 2/5 3/5 4/5

A()
2

2
4/5 3/5 2/5 /5 0 /5 2/5 3/5 4/5

All the zeros lie on the unit circle, with equal spacing between them. From that, we can sketch the
frequency response.

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4.8 Matching. Match each impulse response with its frequency response and zero diagram by filling out
the following table. You should do this problem with out using a computer.

Impulse Response Zero Diagram Frequency Response

A
B
C
D

IMPULSE RESPONSE A IMPULSE RESPONSE B

2 2

1 1

0 0

1 1

2 2
0 2 4 6 0 2 4 6

IMPULSE RESPONSE C IMPULSE RESPONSE D

2 2

1 1

0 0

1 1

2 2
0 2 4 6 0 2 4 6

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ZPLANE A ZPLANE B
1 1

0.5 0.5

Imaginary Part

Imaginary Part
5 4
0 0

0.5 0.5

1 1
1 0.5 0 0.5 1 1 0.5 0 0.5 1
Real Part Real Part

ZPLANE C ZPLANE D
1 1

0.5 0.5
Imaginary Part

Imaginary Part
4 5
0 0

0.5 0.5

1 1
1 0.5 0 0.5 1 1 0.5 0 0.5 1
Real Part Real Part

FREQ RESP A FREQ RESP B

5 5

4 4

3 3

2 2

1 1

0 0
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1
/ /

FREQ RESP C FREQ RESP D

6 6

5 5

4 4

3 3

2 2

1 1

0 0
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1
/ /

Solution:

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Impulse Response Zero Diagram Frequency Response

A C B
B A D
C B A
D D C

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EL 713: Digital Signal Processing Extra Problem Solutions

4.11 FIR Filter Matching Problem.

The following Matlab code fragment defines the impulse responses of four different FIR digital filters.

>> h1 = [2 7 12.5 12.5 7 2];

>> h2 = h1 .* ((-1).^(0:5));
>> h3 = conv(h2,[1 -1]);
>> h4 = h3 .* ((-1).^(0:6));

Without consulting MATLAB, match each of the two filters, h3 and h4, with their pole-zero diagrams
shown below.

Impulse Response Pole-Zero Diagram

h3
h4

1 1

0.5 0.5
DIAGRAM 1

DIAGRAM 2
4 6
0 0

0.5 0.5

1 1
1 0 1 1 0 1

1 1

0.5 0.5
DIAGRAM 3

DIAGRAM 4

2 6 6 2
0 0

0.5 0.5

1 1
1 0 1 1 0 1

1 1

0.5 0.5
DIAGRAM 5

DIAGRAM 6

6 6 2
0 0

0.5 0.5

1 1
1 0 1 1 0 1

1 1

0.5 0.5
DIAGRAM 7

DIAGRAM 8

4 2 6
0 0

0.5 0.5

1 1
1 0 1 1 0 1

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EL 713: Digital Signal Processing Extra Problem Solutions

Solution:
Note that h1 is a lowpass filter of Type 2. Also, from the code, we see that h2 (n) = (1)n h1 (n), so h2
is a highpass filter of Type 4. From the code we have that H3 (z) = H2 (z) (1 z 1 ) which puts a null
in the frequency response at = 0. Since h2 already has a zero at z = 1, h3 will have a double zero at
z = 1 and will be a Type 1 filter. So h3 will correspond to pole-zero diagram 4 or 6. (This is not such
a good exercise!) From the code we have h4 (n) = (1)n h3 (n) or H4 (z) = H3 (z) which will negate
all the zeros of H3 (z) so h4 will correspond to pole-zero diagram 3 or 8.

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EL 713: Digital Signal Processing Extra Problem Solutions

4.12 Matching Problem.

The following figures show 6 impulse responses, frequency responses, and zero diagrams. Match each
frequency response and zero diagram to the corresponding impulse response.

Impulse Response Frequency Response Zero Diagram

1
2
3
4
5
6

IMPULSE RESPONSE 1 IMPULSE RESPONSE 2

1 1

0.5 0.5

0 0

0.5 0.5

1 1
0 2 4 6 0 2 4 6

IMPULSE RESPONSE 3 IMPULSE RESPONSE 4

1 1

0.5 0.5

0 0

0.5 0.5

1 1
0 2 4 6 0 2 4 6

IMPULSE RESPONSE 5 IMPULSE RESPONSE 6

1 1

0.5 0.5

0 0

0.5 0.5

1 1
0 2 4 6 0 2 4 6

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EL 713: Digital Signal Processing Extra Problem Solutions

FREQUENCY RESPONSE 1 FREQUENCY RESPONSE 2

2 2

1.5 1.5

1 1

0.5 0.5

0 0
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1

FREQUENCY RESPONSE 3 FREQUENCY RESPONSE 4

2 2.5

2
1.5
1.5
1
1
0.5
0.5

0 0
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1

FREQUENCY RESPONSE 5 FREQUENCY RESPONSE 6

2.5 2

2
1.5
1.5
1
1
0.5
0.5

0 0
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1
/ /

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EL 713: Digital Signal Processing Extra Problem Solutions

ZERO DIAGRAM 1 ZERO DIAGRAM 2

1 1

Imaginary Part

Imaginary Part
0.5 0.5

5 5
0 0

0.5 0.5

1 1
1 0 1 1 0 1
Real Part Real Part
ZERO DIAGRAM 3 ZERO DIAGRAM 4
1 1
Imaginary Part

Imaginary Part
0.5 0.5

5 2 5
0 0

0.5 0.5

1 1
1 0 1 1 0 1
Real Part Real Part
ZERO DIAGRAM 5 ZERO DIAGRAM 6
1 1
Imaginary Part

Imaginary Part
0.5 0.5

5 2 5
0 0

0.5 0.5

1 1
1 0 1 1 0 1
Real Part Real Part

Solution:
Although each impulse is one of the four types of FIR linear phase filter, in this problem, it is instructive
to look at the number of sign changes in the impulse response. Roughly, the more sign changes an
impulse response have, the higher its frequency content is. Impulse response 3 has no sign changes and
it is a lowpass filter that impulse response has the lowest frequency content. Then impulse response
1 has one sign change. Impulres response 6 has 2 sign changes etc. So you can match them to each of
the frequency responses each frequency response has its passband at a different frequency. From the
frequency response, the zero diagram can be directly found.

Impulse Response Frequency Response Zero Diagram

1 3 2
2 2 4
3 5 1
4 6 5
5 4 6
6 1 3

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EL 713: Digital Signal Processing Extra Problem Solutions

4.15 A student is asked to design a Type I and a Type II low-pass FIR linear-phase filter using DFT-based
interpolation. The student turns in the work shown on the next page, which has two problems.
(a) The student does not provide an explanation.
(b) One of the solutions is correct, the other solution is fake (the student just made up the vector h,
it does not follow from the MATLAB commands).
Identify which solution is correct and provide an explanation for it. Why does the fake solution not
work? You should be able to do this problem with out actually using MATLAB.

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EL 713: Digital Signal Processing Extra Problem Solutions

TYPE I FIR IMPULSE RESPONSE:

>> H = [1 1 1 1 1 0 0 0 0 0 0 1 1 1 1];
>> v = ifft(H);
>> h = [v(9:15); v(1:8)]

h =

0.0394
-0.0667
0
0.0853
-0.0667
-0.0963
0.3050
0.6000
0.3050
-0.0963
-0.0667
0.0853
0
-0.0667
0.0394

TYPE II FIR IMPULSE RESPONSE:

>> H = [1 1 1 1 1 0 0 0 0 0 1 1 1 1];
>> v = ifft(H);
>> h = [v(9:14); v(1:8)]

h =

-0.0318
-0.0494
0.0891
-0.0255
-0.1287
0.2892
0.6429
0.6429
0.2892
-0.1287
-0.0255
0.0891
-0.0494
-0.0318

Solution:
The Type II filter solution is incorrect. The Type I filter solution is right.
For the Type I solution, note that the vector H is of length 15 and that it is real and circularly symmetric.
That means that the inverse DFT, (v in the code) is also real and circularly symmetric, and also of
length 15. The vector v will have the following symmetry pattern:

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EL 713: Digital Signal Processing Extra Problem Solutions

v = [a b c d e f g h h g f e d c b]

So the vector h produced by the 3rd line of MATLAB code in the Type I solution will have the symmetry
pattern:

v = [h g f e d c b a b c d e f g h]

which is indeed a Type I impulse response.

On the other hand, for the Type II solution, note that the vector H is of length 14 and that it is also
real and circularly symmetric. Therefore, the inverse DFT, (v in the MATLAB code) is also real and
circularly symmetric, and also of length 14. The vector v will then have the following symmetry pattern:

v = [a b c d e f g h g f e d c b]

So the vector h produced by the 3rd line of MATLAB code in the Type II solution will have the symmetry
pattern:

v = [g f e d c b a b c d e f g h]

which is not a symmetric impulse response. The actual result of the MATLAB commands for the Type
II filter is:

h =

-0.0318
-0.0494
0.0891
-0.0255
-0.1287
0.2892
0.6429
0.2892
-0.1287
-0.0255
0.0891
-0.0494
-0.0318
0.0714

To design a Type II FIR filter using DFT-based interpolation, we can use a phase-shift of A prior to
using the DFT. In that case, the ones at the end of the A vector should be 1.

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EL 713: Digital Signal Processing Extra Problem Solutions

4.16 Optional : Mitra 4.19. (But use the frequencies 0.2, 0.4, 0.9)
Solution:
This is an interpolation problem:

A() = 0 for = 0.2, 0.9, and A(0.4) = 1.

We can use the general interpolation approach described in the notes. The impulse response we get as
a solution is

h(n) = [0.3968, 0.1127, 0.4276, 0.1127, 0.3968]

The frequency response we obtain is shown in the figure.

IMPULSE RESPONSE h(n)
1

0.8

0.6

0.4

0.2

0.2

0.4

0.6

0.8

1
1 0 1 2 3 4 5
n

|A()|
1.4

1.2

0.8

0.6

0.4

0.2

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
/

N = 5;
M = (N-1)/2;
wk = [0.2 0.4 0.9]*pi;
Ak = [0 1 0];
C = cos(wk*[0:M]);
a = C\Ak;
h = (1/2)*[a([M:-1:1]+1); 2*a([0]+1); a([1:M]+1)];

figure(1)
clf
subplot(2,1,1)
stem(0:N-1,h,.)
xlabel(n)
title(IMPULSE RESPONSE h(n))
axis([-1 5 -1 1])
[A,w] = firamp(h,1);
subplot(2,1,2)

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EL 713: Digital Signal Processing Extra Problem Solutions

5.3 (Porat 6.1) We saw that the height of the largest side-lobe of the rectangular window is about -13.5
dB relative to the main-lobe. What is the relative height of the smallest side-lobe? You may assume
the window length N is odd. (Hint: At what frequency will it be located?).
Solution:
The smallest side lobe of the Dirichlet kernel is at = . The formula is

sin N2


f
S () =
.
sin 21
so
N


sin
S f () = 2
1

=1
sin 2

when N is odd. Recall that S f (0) = N . The relative height of the smallest side-lobe is therefore 1/N .

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EL 713: Digital Signal Processing Extra Problem Solutions

5.15 The following spectrogram is taken from the HW submitted by an EL 713 student.

Which of the following three parameters would you suggest the student modify to best improve the ap-
pearance of this spectrogram? What change would you make to that parameter? The three parameters
are:
R = block length.
L = time lapse between blocks.
N = FFT length. (Each block is zero-padded to length N .)
Explain your answer.
Solution:
The student should reduce L the time-lapse between blocks. (Equivalently, the overlap fraction should
be increased.) Making that change will reduce the step like appearance of the spectrogram.

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5.17 The following figures show a signal and its spectrogram, computed with different sets of parameters,
R {30, 60}, N {64, 256}
where
R = block length
N = FFT length (nfft in the Matlab specgram function). (Each block is zero-padded to length N .)
(a) For each of the spectrograms, indicate what you think R and N are, and explain your choices.
(b) Describe how the spectrogram would change if the time-skip (L in the notes), is increased.

0.5
SIGNAL

0.5

1
0 20 40 60 80 100 120 140 160 180 200

SPECTROGRAM A SPECTROGRAM B SPECTROGRAM C

0.5 0.5 0.5

0.4 0.4 0.4

frequency

frequency

0.3 0.3 frequency 0.3

0.2 0.2 0.2

0.1 0.1 0.1

0 0 0
0 50 100 150 0 50 100 150 0 50 100 150
time time time

Solution:
For spectrogram B, the vertical width of the bar with respect to the frequency axis is less than for
spectrogram A, so spectorgram B uses a longer block length than spectrogram A. In addition, the point
in the middle where the frequency starts to decrease is blurred, which also suggests that spectrogram B
uses a longer block length.
The horizontal stripe effect in spectrogram C is due to discontinuities along the frequency axis so for
spectrogram C there is less zero padding that for A and B. Except for the stripe artifact, spectrogram
C appears to be similar to spectram A, so it appears that they use the same block length.

Spectrogram R N
A 30 256
B 60 256
C 30 64

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