Adsp Ws1415 Solutions

DIGITAL
SIGNAL PROCESSING AND

SYSTEM THEORY
TECHNICAL FACULTY,
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Problem 1
(relationship between continuous and discrete signals)
(a) Keeping in mind that after sampling , = T , the Fourier transform of v(n) is
V (ej )
Va (j)
1
T
1
2
(b) A straight-forward application of the Nyquist criterion would lead to an incorrect

conclusion that the sampling rate is at least twice the maximum frequency of va (t),
or 22 . However, since the spectrum is bandpass, we only need to ensure that the
replications in frequency which occur as a result of sampling do not overlap with
the original. (See figure below of Vs (j).) Therefore, we only need to ensure
2 2/T < 1 T < 2/(2 1 ).
Vs (j)
> 2 2/T
Digital Signal Processing and System Theory, Prof. Dr.-Ing. Gerhard Schmidt, www.dss.tf.uni-kiel.de
Advanced Digital Signal Processing, Exercise Solutions WS 2014/2015
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Problem 2
domain)
(overall system for filtering a continuous-time signal in digital
(a) vi (t) = va (t) p(t)
sVi (j)
Vi (j)
1/T
2/T
2/T
V (ej )
1/T
(b) Since H(ej ) in an ideal lowpass filter with c = /4, we dont care about any
signal aliasing that occurs in the region /4 . We require:
2/T 2 10000 Hz /(4T )
8
1/T
10000 Hz
7
7
104 s
T
8
Also, once all of the signal lies in the range || /4, the filter will be ineffective,
i.e., /4 T (2 104 Hz). So, T 12.5s.
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(c) Sketch of wc as a function of 1/T

c
Slope = /4
8/7 104
8 104
1/T
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Problem 3
(quantization)
(a) Number of quantization levels L

L = 2b ; b = 4
L = 24
= 16
Quantization step
= R/L
= 10/16 V
= 0.625 V
(b) Input-output characteristic of the midtreat quantizer (See lecture slide Slide II-22)
Differences between a midrise and a midtreat quantizer:
(i) A midrise quantizer has no zero level.
(ii) A midrise quantizer is odd symmetric along the y-axis.
(c) For n = 1250
v(n) = 2.925
vq (n) = Q[v(n)]
= 3.125
eq (n) = v(n) vq (n)
= 0.2
Bipolar code (sign and magnitude representation) of the quantized value

Code for 5 = 0101
(d) Real system
v(n)
vq (n)
Q[
Mathematical model
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eq (n)
vq (n)
v(n)
(e) Power of the quantization noise PN

PN = 2 /12
(0.625)2
12
= 0.03255
(f) SNR/dB = 6.02b + 10.8 20 log10 (R/v )

R is the range of the quantizer
v is the RMS amplitude of the input signal
R = 10 V, v = 52 = 3.5355, b = 4
SNR/dB = 25.8477 dB
Linear value = 384.388
(g) New v = 12 = 0.707
New SNR/dB = 11.8683 dB
(h) Compute b from the equation 6.02b + 10.8 20 log10 (R/v ) by equating it to 45
dB.
For 1V input:
b 10 ( Because b has to be an integer )
For 5V input:
b 8 ( Because b has to be an integer )
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Problem 4 (DFT and convolution)

M =8
(a) H8 () =
7
P
k=0
h(n) W8 n = 1 + W8
Y8 () = 1 + W8 + W8 2 + W8 3
(b) y(n) = v(n)
h(n) =
7
P
k=0
v(k)h((k n))M
Y8 () = V8 () H8 ()
V8 () = 1 + W8 2
v(k) = {1, 0, 1, 0, 0, 0, 0, 0}
(c) v(n) has length 3 = M1

h(n) has length 2= M2
M1 + M2 - 1 < M
cyclic convolution equals linear convolution
y(n) = v(n)
h(n)
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v(n)
h((0 n))M
h((1 n))M
h((2 n))M
h((3 n))M
h((4 n))M
h((7 n))M
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Problem 5 (DFT)
(a) Let T be the period of the time-limited signal, v0 (t).
2
T =w
0
4
= 2T (two periods)
t= w
0
vo (t)
1
t
1
T

(b) v(n) = v0 (t)

t=nTA
= v(nTA ) = sin(w0 n
4w0 )
= sin( 4 n)
where TA is the sampling period

v(n) = sin( 4 n) n = 0, 1, 2, . . . , 15
TA = 4w
= T8
wA = 8w0
0
v(n)
1
T A 2T A
8T A
-1
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(c) V () = DF T16 {v(n)} =
15
P
n=0
n
v(n)W16
z = ej

= Z{v(n)}

WM = ej M

V () = V (z)

z=W16
z = WM
here, M = 16
j 2
M ,=1
z=e
W 3
W
2
4
8
+ 16 )
) ( 16 + W16
V () = (1 + W16
) (1 W16
{z
} |
{z
}
|
2
2
I
I)
II)
(1
II
(1 +
4
W16
)
8
W16
)
{z
III
2
0
if is even
if is odd
0
1 (j)
= 0, 4, 8, 12
otherwise
V () equals zero for all except = 2, 6, 10, 14 (III) needs to be computed

only for = 2, 6, 10, 14, however due to the symmetric property of the DFT (III)
is computed only for = 2, 6
W16
W 3
2
+ W16
+ 16
2
2
III)
for = 2:
for = 6:
1 ( 1 j 1 ) + (j)
2
2
2
1 ( 1 j 1 ) + (j)
2
2
2
+
+
1 ( 1 j 1 ) =
2
2
2
1 ( 1 j 1 ) = 0
2
2
2
2j
Now, using the results of (I), (II) and (III) we get V () as

W 3
W
2
4
8
+ 16 )
) ( 16 + W16
V () = (1 + W16
) (1 W16
{z
} |
{z
}
|
2
2
I
II
V (2) = (2)(2)(2j) = (8j)
{z
III
V (6) = (2)(2)(0) = 0
for real valued sequences,
V () = V (M )
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V (10) = V (16 10)

= V (6)
=0
V (14) = V (16 14)
= V (2)
= 8j
V () =
8j
=2
= 14
otherwise
8j
(1)
(d) The discrete sequence v(n) is also time-limited (finite-length sequence)

M=
b length of sequence v(n)

v(n) = sin( 4 n) 1 (n) 1 (n M )
{z
fM (n)
DT F T {fM (n)} =
M
1
X
n=0
1 ejn
M
ej 2 (ej 2 ej 2 )
1 ejM
=
=
1 ej
ej 2 (ej 2 ej 2 )
= ej(
M 1
)
2
sin( M
2 )
sin( 2 )
v(n) = sin( n) fM (n)

4
F {v(n)} = V (ej )
sin M2
1
V (e ) =
DTFT{v(n)} ej 2 (M 1)
2
sin 2
j
(M 1)
sin M2
1
j 0 ( + ) 0 ( ) ej 2
2
4
4
sin 2
M
(M 1)
j j (M 1) (+ ) sin[( + 4 ) M
j 2 ( 4 ) sin[( 4 ) 2 ]
2 ]
2
4
=
e
+

2
sin[ 2 4 ]
sin[ 2 4 ]
(e) V (ej ) =
n=
v(n) fM (n).ejn =
DFT{v(n)} = V

(ej )
15
P
n=0
v(n) ejn
= 0, 1, 2, . . . , 15
= 2
M
10
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(ej )
= V ()
.
= 2
16
j j 15 ( + ) sin[( 8 + 4 )8]
j 15
( 4 ) sin[( 8 4 )8]
2 8
=
e
e 2 8 4
2
sin[ 8 2 4 ]
sin[ 8 2 4 ]
(2)
(1) and (2) should be equal.
11
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Problem 6
(DFT, zero padding, leakage)
Let va (t) be a time-continuous periodic signal

va (t) = 1 + cos(240t) + 3 cos(2120t).
The signal is sampled (s = 2280s1 ) to produce the sequence v(n). For practical
purposes (delay, complexity) the sequence is limited to L samples. M is the length of
the DFT.
1
a) one period L=7, DFTlength M=7, sampling frequency =2 280 s

s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.005
0.01
0.015
Time [s]
0.02
0.025
0.03
DFT V () and Fourier transform V(ej)

M
12
10
Amplitude
8
6
4
2
0
0
0
0
2
pi/2
40
pi
80
120
160
Frequency , , /2
6
3pi/2
200
7
2pi
240
280
12
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b) one period L=7, DFTlength M=14 (zero padding), sampling frequency =2 280 s1
s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.005
0.01
0.015
Zeit [s]
0.02
0.025
0.03
DFT V () and Fourier transform V(e )

M
12
10
Amplitude
8
6
4
2
0
0
0
Problem 7
4
pi/2
40
10
pi
80
120
160
Frequency , , /2
12
3pi/2
200
14
2pi
240
280
(FFT)
(a) Even indexed sequence: ve (n) = [v(0), v(2), v(4), v(6)],

Odd indexed sequency: vo (n) = [v(1), v(3), v(5), v(7)]
13
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c) four periods L=28, DFTlength M=28, sampling frequency =2 280 s1

s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.01
0.02
0.03
0.04
0.05
0.06
Zeit [s]
0.07
0.08
0.09
0.1

M
50
Amplitude
40
30
20
10
0
10
15
pi/2
40
20
pi
80
25
3pi/2
120
160
Frequency , , /2
200
2pi
240
280
DFT expressions of the sequences:

Ve, () = DF T {ve (n)}
ve W4n
n=0
=
Vo, () = DF T {vo (n)}
4
X
4
X
ve ej
n=0
4
X
2
n
4
vo W4n
n=0
4
X
vo ej
2
n
4
n=0
(b) DFT of v(n) is given by

V8 () =
M/21
P
n=0
v(2n) ej
2
2n
8
M/21
P
n=0
v(2n + 1) ej
2
(2n+1)
8
14
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d) four periods L=28, DFTlength M=56 (zero padding), sampling frequency =2 280 s1
s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.01
0.02
0.03
0.04
0.05
0.06
Zeit [s]
0.07
0.08
0.09
0.1

M
50
Amplitude
40
30
20
10
0
10
20
30
pi/2
40
ej
2
(2n)
8
80
= ej
2
(n)
4
(c)
n=0
50
3pi/2
120
160
Frequency , , /2
200
2pi
240
280
= W4n
M/21
M/21
V8 () =
40
pi
v1 (n) W4n + W8
{z
Ve, ()
n=0
v2 (n) W4n
{z
Vo, ()
Direct DFT:
Complexity: M 2 = 64
Modified method:
Complexity: 2 (M/2)2 + M = 40
(d) Yes, the complextiy can be further reduced by applying the same principle. We
get,
15
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e) two periods L=14, DFTlength M=15 (zero padding), sampling frequency =2 280 s
s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.01
0.02
0.03
Time [s]
0.04
0.05

M
25
Amplitude
20
15
10
0
0
10
pi/2
40
V8 () =
1
P
n=0
15
pi
80
v1,1 (n)W2n +W82
3pi/2
120
160
Frequency , , /2
1
P
n=0
200
v1,2 (n)W2n +
1
P
n=0
2pi
240
v2,1 (n)W2n +W82
280
1
P
n=0
v2,2 (n)W2n
16
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f) two periods L=14, DFTlength M=21 (zero padding), sampling frequency =2 280 s
s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.01
0.02
0.03
Time [s]
0.04
0.05

M
25
Amplitude
20
15
10
0
0
0
0
Problem 8
pi/2
40
10
12
14
16
pi
80
120
160
Frequency , , /2
18
3pi/2
200
20
2pi
240
280
(FFT)
Given x(n) = ej(/M )n
When M is even, X() =
M ej/4 ej(/M )
17
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g) L=30, DFTlength M=30, sampling frequency =2 280 s

s
5
4
3
Amplitude
2
1
0
1
2
3
0
0.02
0.04
0.06
Time [s]
0.08
0.1

M
50
Amplitude
40
30
20
10
0
0
10
15
pi/2
40
20
25
pi
80
3pi/2
120
160
Frequency , , /2
200
30
2pi
240
280
y(n) = ej(/M )n
Y () =
=
=
=
=
2M
1
X
n=0
M
1
X
n=0
M
1
X
n=0
M
1
X
n=0
M
1
X
y(n)ej(2/2M )n
2
ej(/M )n ej(2/2M ) +
2
ej(/M )n ej(2/2M ) +
2
ej(/M )n ej(2/2M ) +
2
ej(/M )n ej(2/2M ) +
n=0
M
1
X
ej(/M )n ej(2/2M )n
n=M
N
1
X
l=0
N
1
X
l=0
N
1
X
ej(/M )(l+M ) ej(2/2M )(l+M ) |n = l + M

2
ej(/M )(l+M ) ej(2/2M )(l+M )

2
ej(/M )(l+M ) ej(2/2M )(l+M )
l=0
= (After simplification)
=
2M
1
X
ej(/M )n ej(2/2M ) + (1)
M
1
X
ej(/M )n ej(2/2M )
Digital Signal
Processing and System Theory, Prof. Dr.-Ing.
Gerhard Schmidt, www.dss.tf.uni-kiel.de
n=0
n=0
Advanced Digital Signal Processing,
Exercise
Solutions
WS
2014/2015
M 1
= (1 + (1) )
ej(/M )n ej(2/M )/2
n=0
18
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h) L=15, DFTlength M=30 (zero padding), sampling frequency =2 280 s

s
5
4
Amplitude
3
2
1
0
1
2
3
0
0.01
0.02
0.03
Time [s]
0.04
0.05
0.06

M
30
25
Amplitude
20
15
10
5
0
0
0
0
10
15
pi/2
40
20
25
pi
80
Y () =
120
160
Frequency , , /2
2X(/4)
0
3pi/2
200
30
2pi
240
280
is even
is odd
19
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v(0)
v(1)
v(2)
v(3)
v(4)
v(5)
v(6)
v(7)
DFT
Order 8
v(0)
v(2)
v(4)
v(6)
W80
W81
W82
W83
DFT
Order 4
v(1)
v(3)
v(5)
v(7)
Problem 9
V8 (0)
V8 (1)
V8 (2)
V8 (3)
V8 (4)
V8 (5)
V8 (6)
V8 (7)
W84
W85
W86
W87
DFT
Order 4
V8 (0)
V8 (1)
V8 (2)
V8 (3)
V8 (4)
V8 (5)
V8 (6)
V8 (7)
(FFT)
(a) Since x(n) is real valued x(n) = x (n)

Compute the M -point DFT X()
X() =
M
1
X
x (n)ej2/M n
n=0
1
M
X
x( n)ej2/M n
n=0
1
M
X
x( n)ej2/M n ej2/M M
n=0
= X (M )
XR () = XR (M )
& XI () = XI (M )
(3)
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(b) Given real valued sequences x1 (n)
sX1 () and x2 (n)
sX2 ()
Complex sequence g(n) = x1 (n) + jx2 (n)

G() = GR () + jGI ()
G() = X1 () + X2 ()
= (X1ER () + jX1OI ()) + j(X2ER () + jX2OI ())
= (X1ER () X2OI ()) + j(X1OI () + X2ER ())
|
{z
Real part GR()
{z
Imaginary part GI()
Even and odd parts of G()

GER () = 1/2{GR () + GR(M ) }
= X1 ()
Similarly,
GOR () = X2OI ()
GEI () = X2ER ()
GOI () = X1OI ()
And finally,
X1 () = GER () + jGOI ()
X2 () = GEI () jGOR ()
21
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Problem 10 (signal flow graph)

The signal flow graph in figure of the problem describes the input-output relationship
of v(k) and y(k).
X(z) = V (z) + 0.5 X(z)z 1

V (z)
X(z) =
1 0.5z 1
Y (z) = 2 (3V (z) + X(z)) + 2X(z)z 1
2V (z)z 1
2V (z)
+
Y (z) = 6V (z) +
1 0.5z 1 1 0.5z 1
6V (z) 3V (z)z 1 + 2V (z) + 2V (z)z 1
Y (z) =
1 0.5z 1
8 z 1
V (z)
Y (z) =
1 0.5z 1
h(k) = 8 (0.5)k 1 (k) (0.5)k1 1 (k 1)
22
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Problem 11 (signal flow graph)

Bottom figure in the question:
Y (z) = V (z)z 1 + 2rcos(0 )Y (z)z 1 r2 Y (z)z 2
H(z) =
Y (z)
z 1
=
V (z)
1 2rcos(0 )z 1 + r2 z 2
Top figure in the question:

X(z) = V (z) r2 sin2 (0 )Y (z)z 1
W (z) = X(z) + rcos(0 )W (z)z 1

X(z)
V (z) r2 sin2 (0 )Y (z)z 1
W (z) =
=
1 rcos(0 )z 1
1 rcos(0 )z 1
W (z)z 1
Y (z) = W (z)z 1 + rcos(0 )Y (z)z 1 =
1 rcos(0 )z 1
2
2
1
1
(V (z) r sin (0 )Y (z)z )z
Y (z) =
(1 rcos(0 )z 1 )2
Y (z)((1 rcos(0 )z 1 )2 + r2 sin2 (0 )z 2 ) = V (z)z 1
H(z) =
z 1
Y (z)
=
V (z)
1 2rcos(0 )z 1 + r2 z 2
23
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Problem 12 (round-off effects in digital filters)

a)
V (z)z
1
V (z)
=
Y (z) = V (z) + Y (z)z 1 =
4
z 1/4
1 41 z 1
z
V (z) = .5
z1
5
1 2
1 2
z
2z
8 z 1/8
= 1/2 +
Y (z) =
= 2 52
(z 1/4)(z 1)
(z 1/4)(z 1)
z 4 z + 1/4
2/3
1/24
+
z 1/4 z 1
invers transform
= 1/2
y(n) = 1/2 o (n) 1/24 (1/4)n1 1 (n 1) + 2/3 (1)n1 1 (n 1)
y(k)|n = 2/3
b)
unquantized case (working from the difference equation):
y(n) = v(n) + 1/4 y(n 1)
y(0) = 1/2,
y(1) = 1/2 + 1/4 1/2 = 5/8,
y(3) = 85/128,
y(4) = 341/512,
y(2) = 1/2 + 1/4 5/8 = 21/32,
y(5) = 1364/2048
quantized case (working from the difference equation):

y(n) = v(n) + Q[1/4 y(n 1)]
y(0) = 1/2,
y(1) = 1/2 + Q[1/4 1/2] = 1/2 + Q[1/8] = 5/8

y(2) = 1/2 + Q[1/4 5/8] = 1/2 + Q[5/32]
y(3) = 5/8
truncation!
1/2 + 1/8 = 5/8
...
c)
direct form II:
H(z) =
V (z) =
1 + z 1
1 41 z 1
1/2
1 + z 1
1/2
1 41 z 1
invers transform
Y (z) = H(z) V (z) =

y(n) = 1/2 (1/4)k
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unquantized case:
y(n)|n = 0
quantized case (working from the difference equation):
y(n) = v(n) + v(n 1) + Q[1/4 y(n 1)]
y(0) = 1/2 + 0 + 0 = 1/2,
y(1) = 1/2 + 1/2 + Q[1/4 1/2] = 1/8,
y(2) = 1/2 1/2 + Q[1/4 1/8] = 0
...
25
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 13
(round-off effects in digital filters)
Let h(n), h1 (n), and h2 (n) represent the unit sample responses corresponding to the
system functions H(z), H1 (z), and H2 (z), respectively. It follows that
h1 (n)
h2 (n)
H(z)
=
=
polynom division
partial fraction expansion
H(z)
(1/2)n 1 (n)
(1/4)n 1 (n)
H1 (z) H2 (z)
1
1
z2
=
1 0.5z 1 1 0.25z 1
z 2 34 z + 1/8
3
1/8
4 z 1/8
=
1
+
3
(z 1/2)(z 1/4)
z 2 4 z + 1/8
B
A
+
1+
(z 1/2) (z 1/4)
3
z 1/8
3/8 1/8
lim 4
=
=1
1/4
z1/2 (z 1/4)
1+
3
4z
3
4z
1/8
3/16 1/8
=
= 1/4
1/4
z1/4 (z 1/2)
1/4
1
1+
z 1/2 z 1/4
lim
inverse transform :
h(n)
=
=
=
0 (n) + (1/2)n1 1 (n 1) 1/4 (1/4)n1 1 (n 1)
0 (n) + 2 (1/2)n 1 (n 1) (1/4)n 1 (n)

(2 (1/2)n (1/4)n )1 (n)
first cascade realization:

y(n)
v(n)
z 1
z 1
1
4
1
2
e1 (n)
e2 (n)
Figure 1: Cascade system realization 1
26
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Qa1 (z)
Qa2 (z)
E1 (z)
1 21 z 1
E1 (z)
1/4 Qa2 (z)z 1 + E2 (z) +
1 21 z 1
E2 (z)
E1 (z)
+
1 41 z 1
(1 21 z 1 )(1 41 z 1 )
lecture eq.: (4.39)
1/2 Qa1 (z)z 1 + E1 (z) =
lecture eq.: (4.39)
{z
corresponds to H2 (z)
{z
corresponds to H(z)
2 at the output of the first cascade realization can

So the quantization noise variance qa
be calculated with help of the impulse responses h2 (n) and h(n) as
2
qa
= e2
n=0
h2 (n) + e2
{z
h22 (n),
n=0
{z
e2 denoting the variance of e1/2 (n)
2 is calculated in the same way and

For the second cascade realization, the variance qb
gives
#
"
2
qb
= e2
h2 (n) +
n=0
n=0
n=0
h21 (n)
n=0
n=0
h21 (n) =
1
= 4/3
1 1/4
h22 (n) =
1
= 16/15
1 1/16
h2 (n) =
4
1
4
+
= 1.83
1 1/4 1 1/8 1 1/16
Therefore,
2
qa
= 2.90 e2
2
qb
= 3.16 e2
and the ratio of noise variances is

2
qb
= 1.09
2
qa
Consequently, the noise power in the second cascade realization is 9% larger than in the
first realization.
27
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 14
(digital filter design)
Determine the unit sample response hi of a linear-phase FIR filter of length L = 4 for
which the amplitude frequency response H0 () at = 0 and = /2 is specified as
H0 (0) = 1,
H0 (/2) = 1/2.
Even length L type-2 or type-4 linear phase system.

H0 (0) = 1 no type-4 linear phase system (H0 (0) = 0!).
Type-2 linear phase system:
L/21
H0 () = 2
= 2
hi cos
i=0
1
X
hi cos
i=0

L1
i
2

3
i .
2
At = 0,
1 = 2
1
X
hi cos(0)
i=0
1/2 = h0 + h1 ,
at = /2,
1/2 = 2
1
X
i=0
1/4 =
hi cos

3
3
i /2 = 2 (h0 cos( ) + h1 cos( ))

2
4
4
1
1
h1 h0 .
2
2
Solving these equations, we get

h0 = 0.073
h1 = 0.427
and with symmetry
h2 = h1
h3 = h0
28
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 15
(a)

Hd (ej ) = (1 21 ())ej(/2 )
(
arg(Hd (ej )) =
for < <
/2 ,
< < 0
/2 , 0 < <
(b) A Hilbert transformer of this nature requires the filter to have a zero at z = 0
which introduces the radians phase difference at that point. Thus, only Types
III and IV fulfill the requirements (see lecture).
(c)
Hd (ej ) = (1 21 ())ej(/2 )
Z0
ej/2
2
Z0
hdi =
1
2
j(/2 ) ji
1
d
2
ej(i ) d
1cos((k ))
,
(i )
0,
2 sin2 ((i )/2)
,
(i )
0,
for < <
ej/2
2
ej(/2 ) eji d
ej(i ) d
i 6=
i=
i 6=
i=
For the windowed FIR system to be linear phase it must be antisymmetric about
L1
2 . Since the ideal Hilbert transformer hdi is antisymmetric about i = we
should choose = L1
2 .
29
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
(d) The delay is
L1
2
211
2
= 10 samples. L is odd, it is therefore a type III system.

|H(ej )|
hi
0.5
0
0.5
0.5
0
0
15
10
i
20
Figure 2: Scetches to problem 15 part d)

(e) The delay is
L1
2
201
2
= 9.5 samples. L is even, it is therefore a type IV system.

|H(ej )|
hi
0.5
0
0.5
0.5
0
0
10
i
15
Figure 3: Scetches to problem 15 part e)
30
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 16
From the given equations we have c(k) = 2h(S k), 1 k S. For type III linear-phase
filters h(S) = 0 and L is odd.
H03 ( ) =
S
X
i=1
ci sin(( )i)
S
X
ci sin(k) cos(i) =
S
X
(1)i+1 ci sin(i).
i=1
i=1
Thus H03 () = H03 ( ) implies

S
X
i=1
or equivalently,
i = 2, 4, 6, . . ..
S
P
i=1
ci sin(i) =
S
X
(1)i+1 ci sin(i),
i=1
(1 (1)i+1 )ci sin(i) = 0 which in turn implies that ci = 0 for
ci = 2 h(S i), 1 i S can be rewritten as hi = 1/2 cSi . As S is even hi = 0 for i

even.
31
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 17
(a) From the given equations, we get

1 = 1 101 /20 = 1 100.005 = 0.0114460
2 = 102 /20 = 101.75 = 0.01778279

10log10 (0.00020355796) 13
10log10 (1 2 ) 13
=
N
= 98.2730
=
2.324
)
2.324 2( 2kHz1.8kHz
12kHz
Since the number must be an integer, we round up the value yielding N = 99.
As the length L = N + 1 is even, a type II FIR filter can be designed to meet
the specifications. A type I filter can be designed by increasing the order by 1 to
N = 100.
(b) Note that the width of the transition bands are not equal. We therefore use the
width of the smallest transition band to compute the order N.
1 =
2 =
N
FP1 FS1
= 0, 031416
2
FT

FS2 FP2
2
= 0, 062832
FT
10log10 (0.002 0.001) 13
= 602.51
2.324 2( 0.35kHz0.3kHz
)
10kHz
The order of the required FIR filter is N = 603. As the length L = N + 1 is even,
a type II FIR filter can be designed to meet the specifications.
32
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 18
(a) In the impulse invariance design, the poles transform as zi = esi T and we have the
relationship
1
1
s
s si
1 e i T z 1
Therefore,
Ha (s) =
2
1
s + 0.1 s + 0.2
In this case the solution is unique, since ha (t) is real, and the poles are both on the
-axis in the s-plane. Due to the periodicity of z = ej a more general answer for a
complex impulse response ha (t) would be
Ha (s) =
2
1
2n
s + (0.1 + j T ) s + (0.2 + j 2m
T )
where n and m are integers.

(b) Using the inverse relationship for the bilinear transform
z=
1 + (T /2)s
1 (T /2)s
we get
Ha (s)
2
1
e0.2
e0.2
T =2
=
=
1(T /2)s
1+(T /2)s
1s
1+s
1
1
e0.4
e0.4
1(T /2)s
1+(T /2)s
1s
1+s
2(s + 1)
s+1
0.2
0.4
s(1 +
+ (1 e
) s(1 + e
) + (1 e0.4 )
1
s+1
s+1
2
0.2
0.4
0.2
0.4
1e
1+e
1+e
s + 1+e0.2
s + 1e
1+e0.4
e0.2 )
Since the bilinear transform does not introduce any ambiguity, the representation is
unique.
33
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 19
(a) Recall that = T , T denoting the sampling period. So the specifications for the
continuous-time signal are
0.89125 |H(ej )| 1,
0 || 0.2/T,
|H(e )| 0.17783,
0:89125
0.3/T || /T.
H (j! )j
0:17783
0:2=T
=T !
0:3=T
(b) At the passband edge, we have

|H(j0.2/T )|2 =
1
1+
2N
( 0.2
c T )
1+
2N
( 0.3
c T )
= 0.891252 .
(4)
= 0.177832 .
(5)
At the stopband edge, we have

|H(j0.3/T |2 =
We can solve these two equations for N and c T and we get

N
c T
= 5.8858
= 0.70474
to solve the equations above exactly. Rounding up to the next integer N = 6 and
inserting N in (1) we get c T = 0.7032 to meet the specifications of the continuoustime filter for the passband edge exactly. Then the stopband edge specifications
of the continuous-time filter are exceeded.
34
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 20
Given:
- Filter C: Continous-time IIR-Filter with System fincrion H(s)
- Filter B: Stable discrete-time filter H(z) derived through bilinear transform from
Filter C
Question: Can Filter B be an FIR-Filter?
Excurs: IIR-Filter(System function):
H(z)IIR
Y (z)
=0 b z
P
=
= n
X(z)
=0 a z
relation to the Laplace-domain: zi = es,i/0,i contains poles and zeros

FIR-Filter(System function):
H(z)FIR =
m
X
b z
=0
contains m zeros and m-th order pole at z = 0

Filter is stable, if the poles are located within the unit circle (z-domain) or within the
left Laplace-(s)-plane.
In this case, filter C is an IIR-Filter, this means, it will have poles, that are nonzero,
because otherwise it woukd be an FIR-Filter. Therefore, and due to the fact, that the
bilinear transform is unique in converting from s-domain to z-domain, also filter B will
habe poles. Thus, filter B cannot be a FIR-Filter.
35
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 21
First we determine the values for 1 and 2 . They are used in the magnitude representation of filter specifications.
1 = 1 101/20 = .108749061866
2 = 1040/20 = .01
1 is related to by
1
1 + 2
1
1
2 =
(1 1 )2
= 0.508847139905
(1 1 )2 =
and is given by
=
1
1 = 99, 995
22
The normalized frequencies for the passband edge and stopband edge in the digital
domain are given by
p = 2 40/240
s = 2 60/240
As the bilinear transform warps the frequency scale, we have to determine the passband
edge and stopband edge in the analog domain by inverse transformation of the values
for the digital domain:
p = 2/T tan(p /2)
s = 2/T tan(s /2)
c /s = tan(p /2)/tan(s /2) = 1.73205080758

This leads to the following filter length:
log10 (/)
Butterworth filter: Nmin
= 9.613 10
log10 (s /p )
Chebyshev filter: Nmin
log10 (( 1 22 +
log10 (s /c +
5.212 N = 6
Elliptic filter: Nmin
1 22 (1 + 2 ))/(2 ))
q
(s /p )2 1)
K(p /s )K( 1 (/)2 )

q
K(/)K( 1 (p /s )2 )
= 3.2 N = 4
36
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
20 log10(H(ej ))
|H(ej )|
0
1
20
0.8
40
0.6
60
0.4
80
0.2
100
0.2
0.4
0.6
0.8
0.2
Passband detail
40
60
80
100
120
0.2
0.6
0.8
Stopband detail
0.1
0.4
0.3
0.5
0.6
0.7
0.8
0.9
Figure 4: Butterworth MATLAB: [b_b,a_b]=butter(10,4/12);
20 log10(H(ej ))
|H(ej )|
0
1
20
0.8
40
0.6
60
0.4
80
0.2
100
0.2
0.4
0.6
0.8
0.2
Passband detail
0.4
0.6
0.8
Stopband detail
0
20
2
30
4
40
6
50
60
0
0.2
0.4
0.4
0.6
0.8
Figure 5: Chebyshev 1 MATLAB: [b_c1,a_c1]=cheby1(6,1,4/12);
37
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
20 log10(H(ej ))
|H(ej )|
0
1
20
0.8
40
0.6
60
0.4
80
0.2
100
0.2
0.4
0.6
0.8
0.2
Passband detail
0.4
0.6
0.8
Stopband detail
0.5
40
0
50
0.5
60
70
1
0.1
0.2
0.3
0.4
0.4
0.6
0.8
Figure 6: Chebyshev 2 MATLAB: [b_c2,a_c2]=cheby2(6,40,6/12);
20 log10(H(ej ))
|H(ej )|
0
1
20
0.8
40
0.6
60
0.4
80
0.2
100
0.2
0.4
0.6
0.8
0.2
Passband detail
0.4
0.6
0.8
Stopband detail
20
25
2
30
35
40
45
50
10
55
0.1
0.2
0.3
0.4
0.4
0.6
0.8
Figure 7: Elliptic MATLAB: [b_e,a_e]=ellip(4,1,40,4/12);
38
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 22
(multirate digital signal processing)
The output y(n) = x(n) if no aliasing occurs as result of downsampling. That is,
X(ej ) = 0 for /3 || .
(a) x(n) = cos(n/4). X(ej ) has impulses at = /4, so there is no aliasing.
y(k) = x(k).
(b) x(n) = cos(n/2).
y(n) 6= x(n).
X(ej ) has impulses at = /2, so there is aliasing.
2
2
(c) x(n) = ( sin(n/8)
)2 = 1/64 ( sin(n/8)
n
n/8 ) = 1/64 (sinc(n/8)) . The spectrum
of sinc(n/8) is a rectangular in the range of /8 /8. The squared
function leads to a triangular spectrum with double bandwidth (multiplication in
time domain convolution in frequency domain). So the highest signal frequency
is |max | = /4 < /3 and no aliasing will occur.
39
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Problem 23
We can analyze the system in the frequency domain:
X(ej )
X(e2j )
2
X(e2j )H1 (ej )

H1 (ej )
Y (ej )
2
Y (ej ) is X(e2j ) H1 (ej ) downsampled by 2:

h
Y (ej ) = 1/2 X(e2j/2 )H1 (ej/2 ) + X(e2j(2)/2 )H1 (ej(2)/2 )

h
= 1/2 X(ej )H1 (ej/2 ) + X(ej(2) )H1 (ej( 2 ) )

h
= 1/2 H1 (ej/2 ) + H1 (ej( 2 ) ) X(ej )

= H2 (ej )X(ej )
h
H2 (ej ) = 1/2 H1 (ej/2 ) + H1 (ej( 2 ) )
40
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Solution to Problem 24
(a) h(n) = 0 for |n| > (RL 1). For a causal system we have a delay by RL 1
samples.
(b) General interpolation condition:
h(0) = 1
h(nL) = 0,
n = 1, 2, . . .
(c)
y(k) =
RL1
X
k=(RL1)
h(k)v(n k) = h(0)v(n) +
RL1
X
k=1
h(k)(v(n k) + v(n + k))
This requires only RL 1 multiplications (assuming h(0) = 1).

(d) Show, that only 2R Multiplications per output sample are required.
y(n) =
n+RL1
X
k=n(RL1)
v(k)h(n k)
+
Due to part b) it has been shown, that only h(0) = 1, and h(Ln) = 0 for n =
+
1, 2, ... This is a general condition related to the interpolation (include (L 1)

zeros).
If n = mL (m an integer), then we dont have any multiplications since h(0) = 1

and the other non-zero samples of v(n) hit at the zeros h(n). Otherwise the
impulse response spans 2RL 1 samples of v(n), but only 2R of these are nonzero. Therefore, we have 2R multiplications in total.
41
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
Amplitude
v(n)
0.5
0
0.5
1
10
0
n
10
Amplitude
h(0n)
0.5
0
0.5
1
10
0
n
10
Amplitude
h(1n)
0.5
0
0.5
1
10
Solution to Problem 25
0
n
10
Steps to do:
a)b)[c) Polyphase decomposition of decimation filters -> moving of all decimation factors
through alls branches before the filter decomposition (efficient structure)
d)e) M = 2, L = 3 -> decimator and interpolator can be changed/turned
f) Polyphase decomposition of interpolator filter -> moving of interpolator factor
through all branches (efficient structure) (z 3 > z)
42
DIGITAL
SYSTEM THEORY
TECHNICAL FACULTY,
OF KIEL
a)
Y (z)
b)
G(z)
G0 (z)
G1 (z)
G0 (z)
G1 (z)
G0 (z)
G1 (z)
G0 (z)
X(z)
z 1
c)
3
z 3
z2
d)
z 1
e)
z 1
z 1
2
f)
2
z 1
z 1
z 1
2
z 1
z 1
G1 (z)
G00 (z)
G01 (z)
G02 (z)
G10 (z)
G1 (z)
1
G12 (z)
z 1
43

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DIGITAL

SIGNAL PROCESSING AND

(relationship between continuous and discrete signals)

(b) A straight-forward application of the Nyquist criterion would lead to an incorrect

(overall system for filtering a continuous-time signal in digital

(a) vi (t) = va (t) p(t)

(c) Sketch of wc as a function of 1/T

(a) Number of quantization levels L

Bipolar code (sign and magnitude representation) of the quantized value

(e) Power of the quantization noise PN

(f) SNR/dB = 6.02b + 10.8 20 log10 (R/v )

Problem 4 (DFT and convolution)

(c) v(n) has length 3 = M1

where TA is the sampling period

(c) V () = DF T16 {v(n)} =

V () equals zero for all except = 2, 6, 10, 14 (III) needs to be computed

Now, using the results of (I), (II) and (III) we get V () as

V (2) = (2)(2)(2j) = (8j)

V (10) = V (16 10)

(d) The discrete sequence v(n) is also time-limited (finite-length sequence)

v(n) = sin( n) fM (n)

(1) and (2) should be equal.

(DFT, zero padding, leakage)

Let va (t) be a time-continuous periodic signal

a) one period L=7, DFTlength M=7, sampling frequency =2 280 s

DFT V () and Fourier transform V(ej)

DFT V () and Fourier transform V(e )

(a) Even indexed sequence: ve (n) = [v(0), v(2), v(4), v(6)],

c) four periods L=28, DFTlength M=28, sampling frequency =2 280 s1

DFT V () and Fourier transform V(e )

DFT expressions of the sequences:

(b) DFT of v(n) is given by

DFT V () and Fourier transform V(e )

DFT V () and Fourier transform V(e )

v1,1 (n)W2n +W82

v2,1 (n)W2n +W82

DFT V () and Fourier transform V(e )

Given x(n) = ej(/M )n

When M is even, X() =

g) L=30, DFTlength M=30, sampling frequency =2 280 s

DFT V () and Fourier transform V(e )

ej(/M )(l+M ) ej(2/2M )(l+M ) |n = l + M

ej(/M )(l+M ) ej(2/2M )(l+M )

ej(/M )(l+M ) ej(2/2M )(l+M )

ej(/M )n ej(2/2M ) + (1)

ej(/M )n ej(2/M )/2

h) L=15, DFTlength M=30 (zero padding), sampling frequency =2 280 s

DFT V () and Fourier transform V(e )

(a) Since x(n) is real valued x(n) = x (n)

(b) Given real valued sequences x1 (n)

sX1 () and x2 (n)

Complex sequence g(n) = x1 (n) + jx2 (n)

Real part GR()

Imaginary part GI()

Even and odd parts of G()

Problem 10 (signal flow graph)

X(z) = V (z) + 0.5 X(z)z 1

Problem 11 (signal flow graph)

Top figure in the question:

W (z) = X(z) + rcos(0 )W (z)z 1

Problem 12 (round-off effects in digital filters)

y(n) = 1/2 o (n) 1/24 (1/4)n1 1 (n 1) + 2/3 (1)n1 1 (n 1)

y(1) = 1/2 + 1/4 1/2 = 5/8,

y(2) = 1/2 + 1/4 5/8 = 21/32,