Adsp Ws1415 Exercises

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DIGITAL
SIGNAL PROCESSING AND
SYSTEM THEORY
Initial Remarks
Sebastian Rohde
Digital Signal Processing and System Theory (DSS)
Office: Audio Lab
Phone: 0431 880-6141
e-mail: ser@tf.uni-kiel.de,
Office hours: Vary. Please make an appointment by email!
Course of the Exercise

youll get hand-outs with problems and we recommend you to
solve them at home as preparation for the exam
the solutions to the problems will be presented in the exercises
at any time you can ask your questions on the material
Exams
see information in the lecture
Literature
see information in the lecture
For updates and downloads please visit the course webpage
http://www.dss.tf.uni-kiel.de/teaching/lecture/teaching_lectures.html
and click on Further details in the ADSP section.
Digital Signal Processing and System Theory, Prof. Dr.-Ing. Gerhard Schmidt, www.dss.tf.uni-kiel.de
Advanced Digital Signal Processing, Exercises WS 2014/2015
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Notation
Symbol
Meaning/Usage
Continuous time variable
Discrete time variable
0 (t)
Continuous time impulse signal
0 (n)
Unit impulse signal (discrete)
1 (n)
Unit step signal (discrete)

Analog frequency in radians per second
= 2/T
Sampling period in seconds
Analog frequency in Hz
fs
Sampling frequency in Hz
Digital frequency in radians

= 2f /fs
v(t)
s V (j)
Continuous Time Fourier transform
v(n)
s V (ej )
Discrete Time Fourier Transform
v(n)
s V ()
Discrete Fourier Transform
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Problem 1
(relationship between continuous and discrete signals)
A complex-valued continuous-time signal va (t) has the Fourier transform shown in figure
1. This signal is sampled to produce the sequence v(n) = va (nT ).
Va (j)
Figure 1: Fourier transform of va (t)

(a) Sketch the Fourier transform V (ej ) of the sequence v(n) for T =
2 .
(b) What is the lowest sampling frequency that can be used without incurring any
aliasing distortion, i.e. so that va (t) can be recovered from v(n)?
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Problem 2
domain)
(overall system for filtering a continuous-time signal in digital
Figure 2 shows an overall system for filtering a continuous-time signal using a discretetime filter. The frequency response of the ideal reconstruction filter Hr (j) and the
discrete-time filter are shown below.
p(t) =
va (t)
n= 0 (t
vi (t)
nT )
Convert from
impulse train
v(n)
y(n)
H(ej )
to discrete-time
sequence
Convert to
impulse train
Hr (ej )
5 105
yi (t)
Hr (ej )
yr (t)
H(ej )
2104
/4
/4
Figure 2: Overall system.

(a) For Va (j) as shown in figure 3 and 1/T = 20kHz sketch Vi (j) and V (ej ).
V a(j)
1
2104
Heff (j)
2104
Figure 3: Spectrum of Va (j) and Heff (j)

For a certain range of values of T, the overall system, with input va (t) and output yr (t), is
equivalent to a continuous-time lowpass filter with frequency response Heff (j) sketched
in figure 3.
(b) Determine the range of values of T for which the information presented above is
true, when Va (j) is bandlimited to || 2 104 as shown in figure 3.
(c) For the range of values determined in (b), sketch c as a function of 1/T .
Note: This is one way of implementing a variable-cutoff continuous-time filter using fixed
continuous-time and discrete-time filters and a variable sampling rate.
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Problem 3
(quantization)
A sinusoid signal v(n) = 5 sin( 0s n) with f = 5 Hz and fs = 10 kHz has to be quantized

(vq = Q[v(n)]) with a midtreat quantizer. The range of the signal is 5 V and the word
length of the quantizer 4 bits. The quantizer at digital full scale.
(a) How many quantization levels L does the quantizer have? What is the value of ?
(b) Sketch the input-output characteristic of the quantizer. How different is a midtreat
quantizer to a midrise quantizer.
(c) For time index n = 1250 calculate the quantized value vq (n), the quantization error
eq (n) and represent vq (n) using bipolar code (sign and magnitude representation).
The quantization error over time can be modeled as a noise that is added to the input
signal.
(d) Sketch the real system and the mathematical model of the system with the added
quantization noise.
(e) Calculate the power Pn of the quantization noise.
(f) Determine the SNR in dB and in linear scale.
The signals amplitude is changed to 1 V, while the range R of the quantizer remains
the same as before.
(g) How is SNR affected with this change?
(h) What world length has to be chosen to achieve an SNR > 45 dB?
Problem 4
(DFT and convolution)
Let h(n) be the sequence {1, 1, 0, 0, 0, 0, 0, 0} and y(n) = {1, 1, 1, 1, 0, 0, 0, 0}.

(a) Calculate the DFT of length 8 for both sequences.
8 v(n).
(b) Determine with help of the DFT a sequence v(n) such that y(n) = h(n)
(c) Let z(n) be the result of the linear convolution of h(n) and v(n): z(n) = h(n)v(n).
Is z(n) = y(n)?
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Problem 5
(DFT)
The time-limited signal

v0 (t) =
sin(0 t) f or 0 t < 4/0

0
otherwise
is sampled with tn = nTA = n 4 0 to produce the time-limited sequence v(n).

(a) Sketch v0 (t).
(b) Determine v(n).
(c) Determine the DFT of v(n).
(d) Determine the Fourier Transform V (ej ) of v(n).
(e) Explain the connection between the DFT{v(n)} and V (ej ).
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. Use MATLAB to solve the following subproblems.
(a) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 7
and M = 7.
(b) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 7
and M = 14 (zero padding).
(c) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 28
and M = 28.
(d) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 28
(e) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 14
(f) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 14
(g) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 30
and M = 30.
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(h) Sketch va (t), v(n), the Fourier transform V (ej ) and the DFT VM () for L = 15
Problem 7
(FFT)
Let v(n) be a time-discrete signal

v(n) = [v(0), v(1), v(2), v(3), v(4), v(5), v(6), v(7)].
(a) Separate the signal v(n) into even and odd time-indices v1 (n) and v2 (n) respectively
and find the DFT expression for each separated sequence.
(b) Now compute the DFT of v(n) using the above expressions.
(c) Sketch the signal flow diagrams when DFT is directly applied to v(n) and as shown
in part (b). Show the reduction in complexity by computing the number of complex
multiplications for each method.
(d) Can the complexity be reduced further? If yes then find the final expression.
(e) Sketch the complete signal flow for part (d).
Problem 8
(FFT)
2
The M -point DFT of the M -point sequence x(n) = ej(/M )n , for M even, is
2
X() = M ej/4 ej(/M ) .
2
Determine the 2M -point sequence y(n) = ej(/M )n , assuming that M is even.

Problem 9
(FFT of real and complex sequences)
Suppose that an FFT program is available that computes the DFT of a complex sequence.
If we wish to compute the DFT of a real sequence, we may simply specify the imaginary
part to be zero and use the program directly. However, the symmetry of the DFT of a
real sequence can be used to reduce the amount of computation.
(a) Let x(n) be a real-valued sequence of length M , and let X() be its DFT with
real and imaginary parts denoted XR () and XI (), respectively; i.e.,
X() = XR () + j XI ().
Show that if x(n) is real, then XR () = XR (M ) and XI () = XI (M )
for = 1, ..., M 1.
(b) Now consider two real-valued sequences x1 (n) and x2 (n) with DFTs X1 () and
X2 (), respectively. Let g(n) be the complex sequence g(n) = x1 (n) + j x2 (n),
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with corresponding DFT G() = GR () + j GI (). Also, let GOR (), GER (),
GOI () and GEI () denote, respectively, the odd part of the real part, the even
part of the real part, the odd part of the imaginary part, and the even part of the
imaginary part of G(). Specifically, for 1 M 1,
GOR () = 1/2{GR () GR (M )},
GER () = 1/2{GR () + GR (M )},
GOI () = 1/2{GI () GI (M )},
GEI () = 1/2{GI () + GI (M )},
and GOR (0) = GOI (0) = 0, GER (0) = GR (0), GEI (0) = GI (0). Determine expressions for X1 () and X2 () in terms of GOR (), GER (), GOI () and GEI ().
Problem 10
(signal flow graph)
The signal flow graph in figure 4 describes the input-output relationship of v(k) and
y(k).
3
v (k )
1/2
y (k )
Figure 4: Signal flow graph of a filter
Determine the differential equation, the transfer function H(z) =

response h(k) of the system.
Y (z)
V (z)
and the impulse
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Problem 11
(signal flow graph)
Show that the systems in figure 5 are equivalent.
()
v k
()
y k
z 1
(
0)
(
0)
r os
r os
r 2 sin2
z 1
(
0 )
()
()
v k
y k
z 1
2r os(
0)
z 1
z 1
r2
Figure 5: Signal flow graph of two systems
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Problem 12
(round-off effects in digital filters)
The flow graph of a first-order system is shown in figure 6

v (k)
y (k)
1/4
Figure 6: First order system

(a) Assuming infinite-precision arithmetic, find the response of the system to the input
v(n) =
.5 for n 0
0 for n < 0
What is the result for large n?

Now suppose that the system is implemented with fixed-point binary arithmetic. The
coefficients and all variables in the network are represented in sign-magnitude notation
with 5 bit (b0 b1 b2 b3 b4 ), b0 denoting the sign. The result of a multiplication of a sequence
value by a coefficient is truncated before additions occur.
(b) Compute the response of the quantized system to input of part a), and plot the
responses of both the quantized and unquantized systems for 0 n 5. How do
the responses compare for large n?
(c) Now consider the system depicted in figure 7, where
v(k) =
.5(1)n for n 0
0
for n < 0
Repeat part a), and b) for this system and input.

v (k)
y (k)
1/4
Figure 7: System for part c)
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Problem 13
(round-off effects in digital filters)
Determine the variance of the round-off noise at the output of the two cascade realizations
of the filter with system function
H(z) =H1 (z) H2 (z)

1
1
, H2 (z) =
H1 (z) =
1
1 0, 5z
1 0, 25z 1
(1)
(2)
y(n)
v(n)
z 1
z 1
1/2
1/4
e1 (n)
e2 (n)
v(n)
y(n)
z 1
z 1
1/2
1/4
e1 (n)
e2 (n)
Figure 8: Two cascaded realizations of filters H1 (z) and H2 (z).
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Problem 14
(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,
Problem 15
H0 (/2) = 1/2.
(filter design)
An ideal discrete-time Hilbert transformer is a system that introduces /2 radians of

phase shift for 0 < < and /2 radians of phase shift for < < 0. The amplitude
frequency response of the Hilbert transformer is shown in figure 9. Such systems are
called ideal /2-radians phase-shifters.
Hd (ej )
1
Figure 9: hilbert transformer

(a) Give a close-form equation (use the step function) for the ideal frequency response
Hd (ej ) of an ideal discrete-time Hilbert transformer that also includes the constant (nonzero) group delay for < < . Plot the phase response of this
system for < < .
(b) What type(s) of FIR linear-phase systems (I, II, III, IV) can be used to approximate
the ideal Hilbert transformer in part a)?
(c) Suppose we want to use the window method with a rectangular window to design
a linear-phase approximation to the ideal Hilbert-transformer. Use Hd (ej ) given
in part (a) to determine the ideal impulse response hd,i if the FIR system is to be
such that hi = 0 for i < 0 and i > L 1.
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(d) What is the delay of the system if L = 21? Sketch (use matlab) the magnitude
of the frequency response of the FIR approximation for this case, assuming a
rectangular window.
(e) What is the delay of the system if L = 20? Sketch (use matlab) the magnitude
of the frequency response of the FIR approximation for this case, assuming a
rectangular window.
Problem 16
(filter design)
Consider a type III linear-phase FIR filter with an amplitude response given by
H03 () = 2
S1
X
i=0
hi sin((S i)).
with S as in the lecture. This equation can be rewritten as

H03 () =
S
X
c(i) sin(i).
i=1
Show that if the amplitude response is symmetric, i.e., H03 () = H03 ( ), then the
even-indexed impulse response samples hi are zero, if S is even.
Problem 17
(filter design)
Digital filter specifications are often given in terms of the loss function,
Hl () = 20log10 (|H(ej )|), in dB. In this problem the peak passband ripple p and
the minimum stopband attenuation s are given in dB, i.e., the loss specifications of the
digital filter are given by
p = 20log10 (1 1 )dB,
d = 20log10 (2 )dB.
(a) Estimate the order of an optimal equiripple linear-phase lowpass FIR filter with the
following specifications: passband edge Fp = 1.8kHz, stopband edge Fs = 2kHz,
p = 0.1dB, s = 35dB, and sampling frequency FT = 12kHz.
The estimation formula can also be used to estimate the length of highpass, bandpass,
and bandstop optimal equiripple FIR filters. Then the width of the smallest transition
band is used to estimate the filter order.
(b) Estimate the order of an optimal equiripple linear-phase bandpass FIR filter with
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the following specifications: passband edges Fp1 = 0.35kHz and Fp2 = 1kHz,
stopband edges Fs1 = 0.3kHz and Fs2 = 1.1kHz, passband ripple 1 = 0.002,
stopband ripple 2 = 0.001, and sampling frequency FT = 10kHz.
Problem 18
(Digital IIR Filter Design)
The system function of a discrete-time system is

H(z) =
2
1
e0.2 z 1
1
1
e0.4 z 1
(a) Assume that this discrete-time filter was designed by the impulse invariance method
with T = 2, i.e. hi = ha (iT ), where ha (t) is real. Find the system function Ha (s)
of a continuous-time filter that could have been the basis for the design. Is your
answer unique? If not, find another system function Ha (s).
(b) Assume that H(z) was obtained by the bilinear transform with T = 2. Find the
system function Ha (s) that could have been the basis for the design. Is your answer
unique? If not, find another Ha (s).
Problem 19
A discrete-time lowpass filter is to be designed by applying the impulse invariance method

to a continuous-time Butterworth filter having magnitude-squared function
|H(j)|2 =
1
2N
1 + ( cut
)
The specifications for the discrete-time signal are

0.89125 |H(ej )| 1,
j
|H(e )| 0.17783,
0 || 0.2,
0.3 || .
Assume that aliasing will not be a problem, i.e., design the continuous-time Butterworth
filter to meet passband and stopband specifications as determined by the discrete-time
filter.
(a) Sketch the tolerance bounds on the magnitude of the frequency response, |H(j)|,
of the continuous-time Butterworth filter such that after application of the impulse
invariance method, the resulting discrete-time filter will satisfy the given design
specifications. Do not assume that T = 1.
(b) Determine the integer order N and the quantity cut T such that the continuous-
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time Butterworth filter exactly meets the specifications determined in part (a) at
the passband edge.
Problem 20
Filter C is a stable continuous-time IIR-filter with system function H(s). Filter B is

a stable discrete-time filter with system function H(z). Filter B is related to Filter C
through the bilinear transformation. Is it possible that filter B is an FIR-filter? Explain
your answer.
Problem 21
A digital lowpass filter is required to meet the following specifications:

Passband ripple: 1dB
Passband edge: 40Hz
Stopband attenuation: 40dB
Stopband edge: 60Hz
Sample rate: 240Hz
The filter is to be designed by performing a bilinear transformation on an analog system
function. Determine what order Butterworth, Chebyshev, and Elliptic analog design
must be used to meet the specifications in the digital implementation. Use a table in a
mathematical handbook to solve the elliptic integrals. Show that for the Butterworth
design the estimation formula for the filter order N (slide (4.129) in the lecture) can be
written as
log(/)
N=
log(s /p )
q
with = 1/22 1. The figure shows the characteristical parameters for the given
specifications.
Problem 22
(multirate digital signal processing)
Consider the system shown in the figure. For each of the following input signals x(n),
indicate whether the output y(n) = x(n).
(a) x(n) = cos(n/4)
(b) x(n) = cos(n/2)
(c) x(n) = ( sin(n/8)
)2
n
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| H(ej ) |2
1
1
1+2
22
p
x(n)
s
y(n)
H(ej )
H(ej )
1
3
Problem 23
Consider the systems shown in the figure. Suppose that H1 (ej ) is fixed and known.
Find H2 (ej ), the frequency response of an LTI system, such that y2 (n) = y1 (n), if the
inputs to the systems are the same.
x(n)
y1 (n)
2
x(n)
H1
(ej )
H2
y2 (n)
(ej )
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Problem 24
The system shown in the figure approximately interpolates the sequence x(n) by a factor
L. Suppose that the linear filter has impulse response h(n) such that h(n) = h(n) and
h(n) = 0 for |k| > RL 1, where R and L are integers; i.e., the impulse response is
symmetric and of length 2RL 1 samples.
x(n)
L
v(n)
y(n)
H(ej )
(a) In answering the following, do not be concerned about causality of the system; it
can be made causal by including some delay. Specifically, how much delay must
be inserted to make the system causal?
(b) What conditions must be satisfied by h(n) in order that y(n) = x(n/L) for
n = 0, L, 2L, 3L, . . . ?
(c) By exploiting the symmetry of the impulse response, show that each sample of
y(n) can be computed with no more than RL multiplications.
(d) By taking advantage of the fact that multiplications by zero need not to be done,
show that only 2R multiplications per output sample are required.
Problem 25
Consider the noninteger sampling rate conversion in the figure. Develope step by step
an efficient structure for the sampling rate conversion, where most calculations are done
in the lowest possible sampling rate.
Y (z)
X(z)
G(z)
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Course of the Exercise

Continuous time variable

Discrete time variable

Continuous time impulse signal

Unit impulse signal (discrete)

Unit step signal (discrete)

Sampling period in seconds

Digital frequency in radians

Continuous Time Fourier transform

Discrete Time Fourier Transform

Discrete Fourier Transform

(relationship between continuous and discrete signals)

Figure 1: Fourier transform of va (t)

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

Figure 2: Overall system.

Figure 3: Spectrum of Va (j) and Heff (j)

A sinusoid signal v(n) = 5 sin( 0s n) with f = 5 Hz and fs = 10 kHz has to be quantized

(DFT and convolution)

Let h(n) be the sequence {1, 1, 0, 0, 0, 0, 0, 0} and y(n) = {1, 1, 1, 1, 0, 0, 0, 0}.

The time-limited signal

sin(0 t) f or 0 t < 4/0

is sampled with tn = nTA = n 4 0 to produce the time-limited sequence v(n).

(DFT, zero padding, leakage)

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

Let v(n) be a time-discrete signal

Determine the 2M -point sequence y(n) = ej(/M )n , assuming that M is even.

(FFT of real and complex sequences)

(signal flow graph)

Figure 4: Signal flow graph of a filter

Determine the differential equation, the transfer function H(z) =

and the impulse

(signal flow graph)

Show that the systems in figure 5 are equivalent.

Figure 5: Signal flow graph of two systems

(round-off effects in digital filters)

The flow graph of a first-order system is shown in figure 6

Figure 6: First order system

What is the result for large n?

Repeat part a), and b) for this system and input.

Figure 7: System for part c)

(round-off effects in digital filters)

H(z) =H1 (z) H2 (z)

Figure 8: Two cascaded realizations of filters H1 (z) and H2 (z).

An ideal discrete-time Hilbert transformer is a system that introduces /2 radians of

Figure 9: hilbert transformer

with S as in the lecture. This equation can be rewritten as

(Digital IIR Filter Design)

The system function of a discrete-time system is

(Digital IIR Filter Design)

A discrete-time lowpass filter is to be designed by applying the impulse invariance method

The specifications for the discrete-time signal are

(Digital IIR Filter Design)

Filter C is a stable continuous-time IIR-filter with system function H(s). Filter B is

(Digital IIR Filter Design)

A digital lowpass filter is required to meet the following specifications:

(multirate digital signal processing)

(multirate digital signal processing)

(multirate digital signal processing)

(multirate digital signal processing)

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