Communications Script 2013 en

Lectures script
Introduction to Communications
Summer semester 2013
Prof. Dr.-Ing. Dr. h.c. Gerhard P. Fettweis

Technical University of Dresden
Faculty of Electrical Technology
Vodafone Chair
Mobile Communications Systems
D-01062 Dresden
Status:
Version:
February 12, 2013

2.1.8
CONTENTS
Contents
1 Preface
2 Goal of the Lecture
2.1
Systems for Information Transmission . . . . . . . . . . . . . . . . . . . . . . . . .
2.2
Concept . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3
Transmission Medium . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.4
Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
10
2.5
Signal Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
10
3 Signal Theory
12
3.1
Sine Signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
12
3.2
Generalized Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
12
3.2.1
Dirac Delta Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
13
3.2.2
Integration of Generalized Functions . . . . . . . . . . . . . . . . . . . . .
14
3.2.3
Dierentiation of Generalized Functions . . . . . . . . . . . . . . . . . . .
15
3.2.4
Sifting Property . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
15
3.2.5
Calculation with Dimensional Parameters . . . . . . . . . . . . . . . . . .
15
Fourier Transformation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
16
3.3.1
Denition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
16
3.3.2
Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
16
3.3
4 Linear Time Invariant System
22
4.1
Denition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
22
4.2
Transfer Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
22
4.3
Impulse Response . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
24
4.4
Convolution as the Basic Operation in Communications . . . . . . . . . . . . . . .
24
4.4.1
Denition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
24
4.4.2
Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
25
4.4.3
Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
25
Consideration in Time and Frequency . . . . . . . . . . . . . . . . . . . . . . . . .
27
4.5.1
Fourier Transformation of Rect Function . . . . . . . . . . . . . . . . . . .
27
4.5.2
Zeit- und Frequenzbegrenzung . . . . . . . . . . . . . . . . . . . . . . . . .
28
4.5.3
Raised-Cosine-Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
29
Central Limit Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
30
4.5
4.6
CONTENTS
5 Bandpass Signal
32
5.1
Up and Down Conversion of Real Baseband Signals . . . . . . . . . . . . . . . . .
32
5.2
Up and Down Conversion of Complex Baseband Signal . . . . . . . . . . . . . . .
37
5.3
General Bandpass Signal Equivalent Low-Pass Signal . . . . . . . . . . . . . . .
41
6 Analog Modulation
42
6.1
Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
42
6.2
Modulation Schemes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
43
6.3
Amplitude Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
43
6.3.1
Description in Time Domain . . . . . . . . . . . . . . . . . . . . . . . . . .
43
6.3.2
Description in Frequency Domain . . . . . . . . . . . . . . . . . . . . . . .
44
6.3.3
Power Balance
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
45
6.3.4
AM-Modulators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
46
6.3.5
AM-Demodulators . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
46
6.3.6
Dierent Types of Amplitude Modulation . . . . . . . . . . . . . . . . . .
48
Phase and Frequency Modulation (Angle Modulation) . . . . . . . . . . . . . . . .
49
6.4.1
49
6.4.2
50
6.4.3
Power Balance
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
52
6.4.4
FM-Modulator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
54
6.4.5
FM-Demodulator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
55
Comparison between Amplitude and Angle Modulation . . . . . . . . . . . . . . .
56
6.4
6.5
7 Analog-Digital-Conversion
57
7.1
Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
57
7.2
Time Discretization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
57
7.2.1
Dirac-Comb-Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
58
7.2.2
Sampling
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
58
7.2.3
Sampling Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
60
7.2.4
Signal Reconstruction Digital-Analog-Conversion . . . . . . . . . . . . .
62
7.3
Value Discretization
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
64
7.3.1
Quantization Characteristic Curve
. . . . . . . . . . . . . . . . . . . . . .
65
7.3.2
Quantization Error Signal . . . . . . . . . . . . . . . . . . . . . . . . . . .
66
7.3.3
Signal-to-Noise Ratio . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
67
7.3.4
Oversampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
68
7.3.5
Error Modelling of AD Converter . . . . . . . . . . . . . . . . . . . . . . .
69
Script Introduction to Communications
CONTENTS
8 Digital Modulation
70
8.1
Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
70
8.2
Types of Modulations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
70
8.3
BPSK Binary Phase Shift Keying . . . . . . . . . . . . . . . . . . . . . . . . . .
71
8.3.1
72
8.3.2
Sender . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
74
8.3.3
Phase Diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
74
8.3.4
75
8.3.5
Receiver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
78
8.3.6
Data Transmission over Noisy Channel . . . . . . . . . . . . . . . . . . . .
79
8.3.7
Calculation of Bit Error Ratio . . . . . . . . . . . . . . . . . . . . . . . . .
81
8.4
QPSK Quaternary Phase Shift Keying . . . . . . . . . . . . . . . . . . . . . . .
85
8.5
Further ASK/PSK Methods . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
85
8.6
Frequency Shift Keying . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
87
References
88
A Formulas
90
A.1 Denitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
90
A.2 Fourier Transformation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
90
A.3 Notes on Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
93
B Appendix
94
B.1 Analytical Signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

C Excercises
94
96
C.1 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
96
C.2 Solutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

D Exams
147
D.1 Exam SS 2006 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147

D.2 Solutions: Exam SS 2006 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
D.3 Exam WS 2006/2007 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
D.4 Solutions: Exam WS 2006/2007 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
D.5 Exam SS 2007 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
D.7 Exam WS 2007/2008 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
CONTENTS

D.9 Exam SS 2008 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
D.11 Exam WS 2008/2009 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
D.13 Exam SS 2009 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 228
D.15 Exam WS 2009/2010 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 244
D.17 Exam SS 2010 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
D.19 Exam WS 2010/2011 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 272
D.21 Exam SS 2011 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
D.23 Exam WS 2011/2012 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
D.25 Exam SS 2012 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
D.27 Exam WS 2012/2013 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 337
Preface
The script contains the lectures for Communications course given in the Summer semester. This
English version of the script is made in order to help the foreign, German language non-speaking
students, to follow the lectures. The content of the script doesnt however dier from the one in
German which is normally used. The same holds for the contents of the lectures.
The rst version of the German script was created in 1996 by a student Michael Hosemann and
from 1997 to 1999 was revised by Dipl. Ing. Achim Nahler. The second complete revision took
place from 1999 to 2001 by Dipl. Ing. Matthias Henker. Additional revisions were done by Dipl.
Ing. Denis Petrovic, Dipl. Ing. M.Sc. Peter Zillmann, Dipl. Ing. Andreas Frotzscher, Dipl. Ing.
Stefan Krone, Dipl. Ing. Fabian Diehm and Dipl. Ing Jan Dohl.
This English version of the script was translated from German by the M.Sc student Jafar Sadeque
Adnan (e-mail: adnanbuet@yahoo.com) under the supervision of Dr.-Ing. Denis Petrovic.
The script is now managed by Dipl.-Ing. Bjorn Almeroth (Telefon: (463) 41040, e-mail:
bjoern.almeroth@ifn.et.tu-dresden.de).
Please report any errors or mistakes to improve the script.
Books for self study
Books on signal theory and LTI systems: [Hof98], [Fli91], [WS93], [Fet96]
Books on communications: [Kam96], [L
uk95], [Pro95], [Cou93]
2 GOAL OF THE LECTURE
Goal of the Lecture
The task of communications consists of carrying information from a sender to a receiver. The
communications technology can be roughly divided into two large areas, the
transmission techniques and the
switching techniques.
In this lecture the emphasis will be on the problems of transmission techniques. Since this area
is also very heterogeneous, only the most important and elementary things will be dealt here.
Examples for applications of communications-engineering are :
Audio broadcast and television
analog: AM-Radio (Medium wave), FM-Radio (USW),
digital: DAB (digital audio broadcasting), DVB (digital video broadcasting),
Telephone
Fixed network,
Cellular network.
2.1
Systems for Information Transmission
One can describe the information transmission systems by a block diagram as in Fig. 2.1.
Source
Redundancy
reduction
(Data
compression)
Increasing
redundancy
(higher error
robustness)
Sourcecoder
Channelcoder
Channel-source-adaptation
Sink
Sourcedecoder
Channeldecoder
lecture contents
Modulator
Multiplexer
Multi-medium
utilization
Channel
Demodulator
Demultiplexer
Figure 2.1: General transmission system in communications
Interference
(noise,
fading)
2.2 Concept
2.2
Concept
The building blocks source, source coder, channel coder, modulator and multiplexer are summarized under the term sender. Accordingly the modules demultiplexer, demodulator, error correction
elements, source decoders and sink belong to the receiver. Senders and receivers can be both stationary (e.g. television station) and can be mobile (e.g. Handy). They are however always power
limited. The channel, as transmitting medium, is bandwidth limited. The transmitted information
is inuenced by noise, amplitude oscillations (fading, caused by drift and shadowing), interference
appearances, time dispersion (delay spread, caused by multi-path propagation) and frequency dispersion (Doppler spread, caused by movement of sender, receiver and/or scatterer/reectors etc.).
2.3
Transmission Medium
The selection of the transmitting medium depends greatly on the demands imposed on the transmission channel (e.g. concerning frequency range or signal bandwidth, in addition, regarding the
concurrent number of users). Possible transmitting mediums are
Twisted Pair (twisted copper cable),
e.g. Telephone cable (link terminal)
coax cable,
e.g. Antenna cable, TV cable
wave guide,
e.g. antenna feed with high frequencies (GHz range)
Optical bre,
e.g. transmission with very high data rates
Radio channel.
e.g. cellular, radio and television broadcast
From the point of radio communications, one can dierentiate between indoor and outdoor applications. A typical example of indoor applications could be a WLAN (wireless local area network)
in one Building. Outdoor applications are e.g. cellular radio networks widely spread throughout
the country, at present GSM-900, DCS-1800 (D1, D2, Eplus and E2 network) and in the future
also UMTS. Also the frequency or wavelength ranges can be dierentiated, begin from the wellknown MW and USW ranges up to the millimeter wave bands and far infrared ranges of optical
communications technology.
10
2 GOAL OF THE LECTURE
2.4
Properties
In the following, we name few characteristics of the communication systems.

Simplex/Duplex A distinction criterion is whether a system can operate in simplex or duplex
mode. Simplex mode means that messages are transferred only into one direction (e.g. broadcast), while in the duplex operation the information can be transferred in both directions
(e.g. Telephone).
Single-Cast/Multi-Cast There are Single-Cast-System (Telephone: 1 source, 1 receiver) and
Multi-Cast-System (Broadcast: 1 source with many receivers)
Packet/circuit-switching Another way to distinguish between systems is based on switching,
circuit-switched (e.g. the good old telephone) or package-switched (e.g. data communication in the InternetIP-protocol).
2.5
Signal Level
Often signals with large power dierences are given. Typical values for signal power P lie between
1W and 1kW. That corresponds to a dierence of 109 . For this reason a logarithmic scale is
advantageous. One example of such a scale is the dBm scale, where the power level LP is normalized
on Pref = 1 mW, as follows
P = power(s(t))
P
LP = 10 log10
dBm
Pref
P
LP = 10 lg
dBm mit Pref = 1 mW
Pref
(2.1)
(2.2)
abs. Power in [mW] 0.1 0.5 1 2 4 8 10 100 1000

rel. Power in [dBm] -10 -3 0 3 6 9 10 20
30
Table 2.1: Absolute power vs. dBm-level
In Tab. 2.1 some absolute powers and their corresponding dBm values are given.
2 W-Handy (D-Net): Pmax = 2 W, equivalent representation as level LPmax =
2W
10 lg 1mW
dBm = 10 lg(2 103 )dBm = (10 lg(2) + 10 lg(103 ))dBm = 33dBm. In GSMStandard it is specied that the level at the base station has to be minimum -102 dBm, i.e.
it can result in a level dierence up to 135 dBm respectively.
0.8 W-Handy (E-Net): 29dBm. In E-Net signals are transmitted with less power than in
D-Net. In addition at frequency range where this system operates (1.8 GHz) attenuation
is much stronger. Therefore in E-network a larger number of base stations is necessary for
transmission in contrast with the D-network, which is a drawback regarding to infrastructure
costs.
11
2.5 Signal Level
Also signal voltages can be given in the logarithmic scale. As reference voltage 0.775 V
is usually used and it corresponds 0 dBu (Choice of reference voltage: Which voltage
is necessary, in order to develop a power of 1 mW at a standard resistance of 600?
= 0.7752 V2 /600 = 1mW). The selection of another standard resistance or reference voltage
will shift the dB-scale accordingly.
Likewise system amplications can be specied as equivalent to level. Indicating e.g. x and y
respectively as input and output of a system, yields the levels as follows,
Power(y)
Power(x)
Power amplication (gain)
= 10 lg(gP ) dB
gP =
Lg P
Amplitude(y)
Amplitude(x)
Amplitude amplication
= 20 lg(gA ) dB
resp. gA =
Lg A
(2.3)
(2.4)
In passive systems attenuation is often given instead of amplication.

Power(x)
Power(y)
1
aP =
gP
Power attenuation
LaP = 10 lg(aP ) dB
LaP = LgP
aP =
Amplitude(x)
Amplitude(y)
1
aA =
gA
Amplitude attenuation
LaA = 20 lg(aA ) dB
LaA = LgA
resp. aA =
Often the relation between user and interfering signals is of special interest. The ratio

Signal power
SN R = 10 lg
dB
Noise power
(2.5)
(2.6)
(2.7)
(2.8)
(2.9)
gives the signal-to-noise-ratio.

Please note,
that level specication in dB always denes ratios between two powers or amplitudes (e.g.
amplication factor, signal-to-noise ratio)
that level specication denes always absolute performances or voltages in dBm (Reference:
power Pref = 1 mW), dbW (Reference: power Pref = 1 W), dBu (Reference: power Uref =
0.775 V)
that while calculating with levels the followings hold:
dB dB = dB
dBm dB = dBm
dBm dBm = dB
dBm + dBm = not dened
(2.10)
(2.11)
(2.12)
(2.13)
(2.14)
12
3 SIGNAL THEORY
Signal Theory
3.1
Sine Signal
Many electrotechnical and also communications engineering problems can be modelled by the
following dierential equations.
+ 1 Q(t) = 0
U = LQ(t)
C
1
+ RQ(t)
U = LQ(t)
+ Q(t) = 0
C
Q(t)
= i(t)
frictionless vibration equation
(3.1)
damped oscillation equation
(3.2)
current ow in the resonant circuit
(3.3)
Assuming that Q(t) has the form Q(t) = Q0 et+ the nontrivial solution of a frictionless vibration
equation is
Q(t) = Q0 e j(0 t+)
(3.4)

with 0 = 1/(LC). Only real part of the solution has practical interest here, which is as well a
solution of the dierential equation.

Q(t) = Re e j(0 t+) = cos(0 t + )
(3.5)
Since the solution of the damped oscillation equation also consists of sine functions (general
harmonic functions), these functions are used as basic signals in the electrical and communications
technology 1 .
In physics, one usually works with the rotative frequency . In communications, one uses usually
results in the expression

the frequency f for practical reasons, so that with the substitution f = 2
Q(t) = Q0 cos(2f0 t + ).
3.2
Generalized Functions
A very important criterion for the classication of signals is the characteristic of absolute integrability. A Signal s(t) is absolutely integrable, when

|s(t)| d t < ms <
(3.6)
Many interesting signals are however not absolutely integrable, as for example the sine signals.
In order to examine and also to describe these signals, the term generalized functions is introduced [BS96]. In mathematics this theory is known as distribution theory [GZZZ95]. Only a short
introduction relevant to communications technology is given here.
A generalized function s(t) can be understood as (see exercise 4) a limit of the series
{sn (t)} = {s0 (t), s1 (t), s2 (t), . . .}
(3.7)
with
s(t) = lim sn (t)
n
(3.8)
Sometimes in the script functions are called signals, in order to clarify situations (even if this is not completely
correct).
13
3.2 Generalized Functions
3.2.1
Dirac Delta Function
In physics, usually the mass m = m0 of a body is assigned to a point x = x0 in a space R3 space
(center of gravity of the body), i.e. the entire mass is concentrated at one point. The resulting
mass density (x) possesses the following characteristics

0 for x = x0
(x) =
(3.9)
for x = x0

(x) d x = m0
(3.10)
R3
i.e. in many cases a function is looked for which fullls the following demands

0 for x = 0
(x) =
for x = 0

(x) d x = 1
(3.11)
(3.12)
A quotation from a mathematics paperback [BS96]

There exists no classical function y = (x) with characteristic like (3.11) and (3.12).
. . . Nevertheless the physicists had worked for an approximation of the dirac delta
function, and nally it was successfully introduced by physicist Paul Dirac in 1930.
Experience from the history of mathematics shows that successful formal
calculations can always be strictly justied in a suitable formulation . . . .
Also in electrical engineering, the Dirac pulse (the Dirac delta function is often dened shortly)
possesses an important meaning.
The Dirac impulse (x) is an arbitrarily narrow signal with unit area, which cause a dened
eect at the output of a system. Not the concrete process of the signal (x), but its energy content
(area) is crucial.
The Dirac impulse (x) can be approximated as in Fig.1 3.1 by a series of functions.
The following conditions must hold:
1. Asymptotic (limit) value

(x) = lim n (x) =
n
2. Unit area
for x = 0
for x = 0
(3.13)
n (x) d x = 1
(3.14)
3. Symmetry
n (x) = n (x)
(3.15)
The Dirac pulse is a Model. In practice, it is not possible to generate a voltage waveform e.g.
u(t) = (t). The Dirac impulse can be symbolically plotted as in Fig. 3.2.
14
3 SIGNAL THEORY
n (x)
n (x)
1
2n
n (x)
n/
1
2n
n1
a) n (x) = n rect(nx)
1
n
1
2n
b) n (x) = n triang(nx)
c) n (x) =
1
2n
n
1
1+(nx)2
Figure 3.1: Dirac impulse approximation by a series of functions (n > 0)
f (x)
f (x)
A0
x0
b) f (x) = A0 (x x0 )
a) f (x) = (x)
Figure 3.2: Graphical representation of Dirac impulse
3.2.2
Integration of Generalized Functions
Based on a generalized function denition, it is sensible to dene the integral over the limit value
of the function sequence

(3.16)
s(x) d x =
sn (x) d x = lim
sn (x) d x
n
Example: Integration of Dirac pulse

x
() d = lim
n () d
(3.17)
with e.g. (n > 0)

n (x) :=
becomes
n
1
1 + (nx)2
x
1

() d = lim arctan n
n
0 for x < 0
(3.18)
for x = 0 (here n (x) is symm.)

for x > 0
(3.19)
(3.20)
15
3.2 Generalized Functions
= (x)
(3.21)
The function (x) is often called step function or unit step (see Fig. 3.3). This function is also a
n (x)
1
x
Figure 3.3: Step function (x)
generalized function and can be, for example,presented as (see exercise 4).
1
1
arctan nx +
n
2
(x) = lim n (x) = lim

n
3.2.3
(3.22)
Dierentiation of Generalized Functions
The dierentiation of generalized functions, also analogous to the integration, can be dened as
d
d
s(x) = lim
sn (x)
n d x
dx
The derivative of the step function provides again the Dirac impulse:
cise 4).
3.2.4
(3.23)
d
(x)
dx
= (x) (see exer-
Sifting Property
A very interesting characteristic of the Dirac pulse is the so-called shifting property (see exercise 4).

s(x) (x x0 ) d x = lim
s(x) n (x x0 ) d x
(3.24)
= s(x0 )
3.2.5
(3.25)
Calculation with Dimensional Parameters
In the preceding paragraph, x was always used as argument of the Dirac pulse or step function.
There x was assumed as dimensionless. In communications, however,one generally operates with
dimensional parameters like time t or frequency f .
In such cases one needs to make sure whether such dimensional measures are valid as arguments
of functions, then e.g. sin(5) or rect(1) should be dened, instead of sin(5 s) or rect(1 Hz).
Here such measures must be standardized.
Possible substitutions are e.g. x t/T or x f /F . In many books t and f are worked out as
dimensionless measures, i.e. t and f are standardized measures with T = 1 s or F = 1 Hz, even if
are referred explicitly.
16
3 SIGNAL THEORY
In this script, the following x should be treated as dimensionless; t and f are on the other hand
dimensional parameters representing respectively time and frequency.
For the Dirac pulse in time therefore the following denition yields

1
t
(t) = lim
(3.26)
rect
T +0 T
T

1
t
= lim
(3.27)
triang
T +0 T
T
1
1

= lim
(3.28)
2
T +0 T
1+ t

By denition here applies (t) d t = 1 or (f ) d f = 1 and accordingly the dimension from

(t) 1 s1 and from (f ) follows 1 s = 1 Hz1 .
3.3
3.3.1
Fourier Transformation
Denition
The idea of the Fourier transformation is based on the fact that a signal s(t) can be represented as
a superposition of the harmonic functions A ej 2f t where A is amplitude and f is frequency. The
Fourier transformation is treated in detail in the lectures Systemtheorie and Mathematik .
Therefore here only once the denition is recalled and insisted on some important characteristics
for communications [BS96].
The Fourier transformation is dened in engineering as 2 (see also Appendix A.2)

S(f ) =
s(t) e j 2f t d t = F {s(t)}
(3.29)
A time domain function s(t) can be transformed into a function S(f ) in complex or frequency
domain. The back or inverse Fourier transformation is dened as

S(f ) ej 2f t d f = F 1 {S(f )}
(3.30)
s(t) =
3.3.2
Properties
The relationship between the time domain function and corresponding frequency domain function
obtained using the Fourier transformation (shortly: FT) is one-to-one. FT pair is symbolically
S(f ). Usually functions in time domain are denoted with lowercase letters
denoted as s(t)
while uppercase letters usually correspond functions in frequency domain.
The integrals (3.29, 3.30) are not always convergent, e.g. the transformation of the important sine
signals does not succeed in the context of classical functions, so that the terms of the generalized
functions and distributions must also be used here.
On the other hand, if the signal s(t) is absolutely integrable (see Section. 3.2), then the Fourier
integral is also convergent and the Fourier transform S(f ) exists and converges for f
towards 0. Due to the symmetry of the FT this applies also in the reverse case.
2
In Mathematics, as well as in Physics sometimes slightly deviated denitions are used.
17
3.3 Fourier Transformation
Reversibility It applies
s(t) = F 1 {F {s(t)}}
Proof:
s(t ) e j 2f t d t ej 2f t d f
s(t) =
t =
f =
=
t =
(3.32)
ej 2f (tt ) d f d t
s(t )
=
t
(3.31)
(3.33)
f =
s(t )(t t ) d t
(3.34)
= s(t)
(3.35)
Due to symmetry of FT S(f ) = F {F 1 {S(f )}} is also valid.

Linearity It applies in general

ai si (t)
ai Si (f )
(3.36)
i.e. signals can be decomposed into component-signals to get the result more easily.
Proof:
ai si (t) e
j 2f t
dt =

ai
si (t) e j 2f t d t
ai Si (f )
(3.37)
(3.38)
FT od Conjugate Complex Signals It states,

S (f )
s (t)
Proof:
F {s (t)} =
j 2f t
s (t) e
dt =

=
(3.39)
s (t) ej 2(f )t d t
(3.40)
s(t) e j 2(f )t d t

= S (f )
(3.41)
S(f )
By substituting t = t one gets a very similar relationship

s (t)
S (f )
(3.42)
Time Reversal With substituting t = t in the Fourier integral one gets the interrelation
s(t)
S(f )
(3.43)
18
3 SIGNAL THEORY
Symmetry The integrals of Fourier and inverse Fourier transform dier only by the sign in the
exponent of the exponential function. It states therefore the duality principle, i.e. the Fourier and
inverse Fourier transform are exchangeable to their signs. The relation, which applies e.g. in time
domain, can also be transferred to frequency domain.
s(f )
(3.44)
S(f ) ej 2f t d f
(3.45)
S(t)

Proof:
s(t) =
Substituting f = t and by Fourier transform yields,

j 2f t
S(t) e
dt =
S(t) ej 2(f )t d t = s(f )
(3.46)
FT of Real Signals For a real signal s(t), s(t) = s (t) and therefore (see FT conjugate complex
signals)
(3.47)
S(f ) = S (f )
From this we get,
Re{S(f )} + j Im{S(f )} = Re{S(f )} j Im{S(f )}
(3.48)
|S(f )| ej arg(S(f )) = |S(f )| e j arg(S(f ))
(3.49)
X(f )
Re{X(f )}
f
Im{X(f )}
Figure 3.4: Real- and imaginary part of the Fourier transform S(f ) of a real signal s(t)
The following consideration is somewhat more illustrative

S(f ) =
s(t) e j 2f t d t

=
s(t) (cos 2f t j sin 2f t) d t
(3.50)
(3.51)
As s(t) is pure real-valued, we can conclude the following

Re{S(f )} =
s(t) cos 2f t d t
even function in f , as cos-term is even in f

Im{S(f )} =
s(t) sin 2f t d t
odd function in f , as sin-term is also odd in f
Both considerations show that both real part of the spectrum Re{(S(f )} and the amplitude
spectrum |S(f )| are even functions; and both imaginary part of the spectrum Im{S(f )} and
phase arg(S(f )) are odd functions. These characteristics are demonstrated in Fig. 3.4.
19
FT of Pure Imaginary Signals For pure imaginary signals s(t), s(t) = s (t) and so (see FT
of pure real signals)
S(f ) = S (f )
Re{S(f )} + j Im{S(f )} = Re{S(f )} + j Im{S(f )}
|S(f )| ej arg(S(f )) = |S(f )| e j(arg(S(f ))+)
(3.52)
(3.53)
(3.54)
The following consideration is somewhat more descriptive

S(f ) =
s(t) e j 2f t d t

=
s(t) (cos 2f t j sin 2f t) d t
(3.55)
(3.56)
Since s(t) is pure imaginary, we have,

Re{S(f )} =
s(t) sin 2f t d t

Im{S(f )} =
s(t) cos 2f t d t
odd function in f , as sin-term is also odd in f

even function in f , as cos-term is also even in f
Here imaginary part of the spectrum Im{S(f )} and the amplitude spectrum |S(f )| are even
functions, as well as real part of the spectrum Re{(S(f )} and phase dierence arg(S(f )) are
odd functions.
Shifting Theorem FT of Time Delayed Signals A shift of the signal s(t) in time domain
corresponds to a linear, frequency dependent, phase shift of the spectrum S(f )
e j 2f t0 S(f )
s(t t0 )
Proof (Substitution t = t t0 ):

F {s(t t0 )} =
j 2f t
(3.57)
s(t t0 ) e
dt =
s(t ) e j 2f (t +t0 ) d t

= e j 2f t0
s(t ) e j 2f t d t = e j 2f t0 S(f )
(3.58)
(3.59)
Shifting Theorem FT of Frequency Shifted Signals Due to the symmetry of FT, reverse
is also valid (see also modulation)
S(f f0 )
ej 2f0 t s(t)
Proof:
S(f f0 ) =
j 2(f f0 )t
s(t) e
dt =
(3.60)

s(t) ej 2f0 t e j 2f t d t = F s(t) ej 2f0 t
(3.61)
20
3 SIGNAL THEORY
Scaling Property If a signal is stretched or squeezed in time, the spectrum in frequency axis
is squeezed (stretched) and also the reverse holds.

1
f
, a = 0
(3.62)
s(at)
S
|a|
a
Proof (Substitution t = at and distinction of cases a > 0, a < 0):

j 2f t
s(at) e
1
dt =
|a|
s(t ) e
j 2f ta
1
S
dt =
|a|

f
a
(3.63)
The fact which lies behind this prediction is that fast changing of time signals possess a broad
spectrum and slowly changing signals have a narrow spectrum.
FT of Dirac Impulse The Fourier transformation of Dirac pulse results directly from its sifting
characteristic:

F {(t)} =
(t) e j 2f t d t = e j 2f 0 = 1
(3.64)
The Dirac pulse possesses thus an innitely widened constant spectrum. This characteristic follows
directly as regarding to the limiting value of previously treated time elongation or scaling property.
Due to the symmetry of the FT and the shifting theorem, the following holds:
1
e
j 2f0 t
(f )
(f f0 )
(t)
(t t0 )
1
e
(3.65)
j 2f t0
(3.66)
FT of Sine Signals The extremely important basic signals cos 2f0 t and sin 2f0 t are not
absolutely integrable, as

| cos 2f0 t| d t = or
| sin 2f0 t| d t =
(3.67)
It is however well known that one can write

1 j 2f0 t
+ e j 2f0 t
e
2

1 j 2f0 t
sin 2f0 t =
e j 2f0 t
e
2j
cos 2f0 t =
(3.68)
(3.69)
Thus one obtains the following transformation rules (see Fig. 3.5)
cos 2f0 t
sin 2f0 t

1
(f + f0 ) + (f f0 )
2
1

1
(f f0 ) (f + f0 ) = j (f + f0 ) (f f0 )
2j
2
With the help of inverse transform these rules can be easily checked.
(3.70)
(3.71)
21
Re{S(f )}
A
2
Im{S(f )}
(Im{S(f )} = 0)
f0
A
2
f0
(Re{S(f )} = 0)
f0
s(t) = A cos 2f0 t
f0
s(t) = A sin 2f0 t
Figure 3.5: Spectrum of cosine and sine function
Convolution Theorem Multiplication of two signals in time domain corresponds to a convolution of the corresponding spectrum in frequency domain. Due to the symmetry of the FT, the
reverse case is also true.
s1 (t) s2 (t)
s1 (t) s2 (t)
S1 (f ) S2 (f )
S1 (f ) S2 (f )
(3.72)
(3.73)
Proof: see eqn. (4.8-4.13)

Modulation The multiplication of a signal s(t) with a sine signal corresponds to a shift of the
corresponding spectrum S(f ) in frequency domain.
S(f f0 )
1
(S(f f0 ) + S(f + f0 ))
2
1
(S(f f0 ) S(f + f0 ))
2j
s(t) ej 2f0 t
s(t) cos 2f0 t
s(t) sin 2f0 t
(3.74)
(3.75)
(3.76)
The proof follows with the help of the relations between sine signal and convolution theory
s(t) ej 2f0 t
S(f ) (f f0 )
(3.77)
as well as the sifting property of Dirac pulse (see Fig. 3.2.4, convolution with respect to f , see
also notes for convolution Fig. A.3)

S(f ) (f f0 ) =
S(f ) ( f0 ) d
(3.78)
= S(f f0 )
(3.79)
S(f ) (f f0 ) = S(f f0 )
(3.80)
The correspondence
will appear again and again in the script and would be very helpful to understand it.
22
4
4.1
4 LINEAR TIME INVARIANT SYSTEM
Linear Time Invariant System

Denition
A system H with input x and output y is given as in Fig. 4.1.
Figure 4.1: System H
At rst some terms are dened here to describe the system:

Causality A system is causal, if for any point in time t0 where t0 < t the system reaction y(t0 )
is independent of the value of x(t) after t0 . The system reaction is determined exclusively
by past values of the input signal. (cause-eect principle, see also exercise 11).
Time invariance A system is time invariant if the transfer function is time-independent, i.e. if
x(t) y(t), it follows that any time shifting of of the input signal causes the same time
shifting of output signal: x(t ) y(t ).
Linearity A system is linear if it holds x1 (t) y1 (t) and x2 (t) y2 (t) holds also x1 (t) + x2 (t)
y1 (t) + y2 (t) and a x1 (t) a y1 (t) (a R).
Source free system A system is source free if from x(t) 0 follows y(t) 0. Linear systems
are always source free.
In the sequel, the systems fullling the characteristics linearity and time invariance will be
studied. Such systems are called (linear time invariant systems), shortly LTI-systems. LTI-systems
do not have to be causal although practically realizable systems are always causal.
4.2
Transfer Function
Let the LTI system H be excited with a harmonic (complex) input signal x(t) = ej 2f0 t with a
frequency of f0 . In the stationary state the output signal is yf0 (t). So,
ej 2f0 t yf0 (t)
(4.1)
ej 2f0 (t ) yf0 (t )
(4.2)
j 2f0
j 2f0 t
2f0
e j
e e
yf0 (t)
(4.3)
Because of time invariance property
and from linearity

a
23
4.2 Transfer Function
At time t = 0
e j 2f0 yf0 (0) e j 2f0
(4.4)
ej 2f0 t yf0 (0) ej 2f0 t
(4.5)
(f f0 ) yf0 (0) (f f0 )
(4.6)
and with substituting t =

Using Fourier transform one gets,
Therefore, a LTI system with a sinusoidal excitation with frequency f0 always generates a sinusoidal output signal with the same frequency f0 (Properties of LTI-systems [Fli91]). This output
oscillation is however weighted with a complex factor yf0 (0), i.e. the input signal is altered only in
amplitude and phase. Assigning the weighted factor yf0 (0) to all frequencies f = f0 over f gives
the transfer function of the LTI system H in frequency domain. The transfer function is usually
denoted as H(f ). So H(f0 ) = yf0 (0) with < f0 < (see Fig. 4.2).
|H(f )|
|X(f )|
0.8
|Y (f )| = |H(f ) X(f )|
fg
fg
fg
fg
Figure 4.2: Amplitude spectrum of the transfer function |H(f )|, as well as that of input and
output signal |X(f )| and |Y (f )|
Let X(f ) be the Fourier transformation of x(t), i.e. the input signal x(t) can be represented as
superposition of innite number of sine waves. Exploiting linearity the relation between input and
output signal of an LTI system is as follows
Y (f ) = H(f ) X(f )
(4.7)
The individual spectral lines of X(f ) produce weighted spectrum Y (f ). A LTI system cannot
produce sine signals with frequencies which are not contained in the input signal.
The function H(f )
is a complex weighting function of the input signal spectrum,
indicates the transfer function of the LTI system H,
can be separated into amplitude response (amplitude |H(f )|) and phase response
(phase arg H(f )).
24
4.3
Impulse Response
Consideration in time domain gives

y(t) = F 1 {H(f )X(f )}

j 2f
h() e
d
x( ) e j 2f d ej 2f t d f
=

=
h()x( ) ej 2f (t(+ )) d d d f
(4.8)
(4.9)
(4.10)
by substitution = + follows

and by using the relation e j 2f

h( )x( ) d
ej 2f (t) d f d
(4.11)
(t ) (see Tab. A.2) follows

h( ) x( ) (t ) d d
(4.12)
And with sifting property nally yields

y(t) =
h(t ) x( ) d = (h x)(t)
(4.13)
The signal h(t) is the inverse Fourier transform of the transfer function H(f ) and is named as
impulse-response, because the system H generates exactly the impulse response y(t) = h(t)
at the output when the excitation is Dirac pulse x(t) = (t).

y(t) =
h(t )x( ) d
(4.14)

h(t )( ) d
(4.15)
=
= h(t)
(4.16)
As a summery, one can say that a linear and an time-invariant (LTI) system can be completely
described by its impulse response h(t) or its transfer function H(f ) = F {h(t)}.
4.4
4.4.1
Convolution as the Basic Operation in Communications

Denition
In eqn. (4.13) the relation between input and output signal of a LTI system H is obtained as

h(t ) x( ) d
(4.17)
y(t) =
25
4.4 Convolution as the Basic Operation in Communications
= (h x)(t)
(4.18)
This integral is known as convolution integral or briey convolution and possesses an very important meaning in system theory and communications technology (see also the annotation in
the Appendix A.3). The output signal y(t) results from the convolution of input signal x(t) with
impulse response h(t) of the system H.
The transfer characteristic of a LTI system may occur thereby alternatively in frequency or time
domain. Both representations are equivalent.
Y (f ) = H(f ) X(f )
y(t) = (h x)(t) = h(t) x(t) =
4.4.2
(4.19)
h(t ) x( ) d
(4.20)
Properties
The convolution is
commutative: g h = h g,
associative: f (g h) = (f g) h,
distributive: f (g + h) = f g + f h and
from g h = 0 gives g = 0 or h = 0
These characteristics can be proved by simple substitution of the integration variables, e.g. commutativity

x1 (t )x2 ( ) d =
x1 ( )x2 (t ) d
with = t
(4.21)

=
x1 ( )x2 (t ) d
(4.22)

=
x1 ( )x2 (t ) d
with :=
(4.23)
4.4.3
Example
Another characteristic of convolution will be given with an example of convolution of two rectangular functions. (see Fig. 4.3 and Exercise 12).
y(t) = rect(t/T ) rect(t/T )

x1 (t)
(4.24)
x2 (t)
rect( /T ) rect((t )/T ) d
(4.25)
26

x1 ( ), x2 (t )
x2
x1
y(t) = x1 (t) x2 (t)
T
2
t+
T2
T
2
T
2
a) 1. case: t < T y(t) = 0

x1 ( ), x2 (t )
x2
x1
y(t) = x1 (t) x2 (t)
T
2
0
T2
T
t+ 2
T
2
b) 2. case: T t < 0 y(t) = T + t

x1 ( ), x2 (t )
y(t) = x1 (t) x2 (t)
x1 x2 1
T2
t
T
2
T
2
t+
T
2
c) 3. case: 0 t < T y(t) = T t

x1
x1 ( ), x2 (t )
x2
1
y(t) = x1 (t) x2 (t)

T
T2
T
2
T
2
t+
T
2
d) 4. case: T t y(t) = 0
Figure 4.3: Convolution of two rectangular functions rect(t/T ) rect(t/T )

=
T
2
T2

rect((t )/T ) d =
T
2
T2
rect(( t)/T ) d
(4.26)
27
4.5 Consideration in Time and Frequency
t+T /2 d t = T + t
/2
= T
T /2
dt = T t
tT /2
for t < T
for
T t<0
(4.27)
for 0 t < T
for T t
The convolution of two signals y(t) = (x1 x2 )(t) causes a signal spreading in time. If x1 (t) is
dierent from zero only in the interval T1 and the signal x2 (t) is dierent from zero in interval T2 ,
then the output signal y(t) is in the interval T1 + T2 dierent from zero.
Further examples can be found in Exercise 11, 13, 14 and 15.
4.5
Consideration in Time and Frequency
In this paragraph relations between the time and frequency response of signals will be explored.
4.5.1
Fourier Transformation of Rect Function
The Fourier transformation of the rect function s(t) = rect(t/T ) yields

rect(t/T ) e j 2f t d t
S(f ) =

=
and using the relation sin x =
(4.28)
T
2
T2
e j 2f t d t =
1
(ej x
2j

j f T
1
ej f T
e
j 2f
(4.29)
e j x )
sin(f T )
sin(f T )
=T
f
f T
= T si(f T )
=
(4.30)
(4.31)
The function si(x) = sin(x)/x is called sinc function (see Fig. 4.4 and also denition in Appendix A.1).
si(t/T )
1
1
t
t/T
Figure 4.4: Sinc function si(t/T )
28
4.5.2
Limitation in Time and Frequency
Most signals are time limited, just because of the fact that a signal generator is rst switched on
and later switched o; therefore the output signal during switch-o is zero, so is time limited. The
question is whether signals can be produced to beboth time and frequency limited. Let us proceed
with a time limited signal.
Let s(t) = 0 for |t| > T /2. It follows then,
s(t) = rect(t/T ) x(t)

S(f ) = T si(f T ) X(f )
(4.32)
(4.33)
Convolution of X(f ) with sinc function si(f T ) causes a spectral dispersion. Since si(f ) possesses
a level decay proportional to 1/f , S(f ) represents an innitely expanded spectrum.
The impulse response or transfer function of a LTI system can be either time or frequency limited,
but never can be both at the same time4 . This means that a time limited signal possesses an
innitely widened spectrum and the reverse also holds.
|S(f )|
fg
fg
Figure 4.5: Quasi band limited signal
The spectrum S(f ) of a time limited signal may however tend to zero for |f | , but it
never becomes zero exactly. Therefore the following picture results in practice (see Fig. 4.5). The
spectrum S(f ) of the signal is lost starting from a certain critical frequency fg in noise, is thus
smaller than the noise level. This noise oor results for example from thermal noise of elements.
Such signals are called quasi bandlimited. In practice these quasi bandlimited signals are treated
like bandlimited signals.
Time domain
nite signal duration
innitely
signal
3
4
expanded
Frequency domain
innitely
expanded
spectrum
bandlimited spectrum
see duration-bandwidth product [Kam96] and applications of band limited non-periodic functions [Hof98]
The remarks made above are not as proof, rather serve as practical illustration of facts.
29
4.5 Consideration in Time and Frequency
4.5.3
Raised-Cosine-Filter
It is desirable to have signals which are both time and frequency limited. In the previous paragraph
it was however stated that this is not possible.
An ideal low-pass lter has the transfer function H(f ) = rect(f /2fg ), where fg is the critical
frequency. The impulse response h(t) is obtained as a result of the inverse Fourier transformation
of the H(f ) which gives sinc function h(t) = 2fg si(2fg t). This impulse response is neither
causal nor time limited. By shifting and chopping causality and time limitation can be
achieved 5 . The sinc function however has still the crucial drawback that its envelope decay is only
proportional with 1/t which is not sucient for many practical applications.
For this reason is Raised Cosine Filter Hrc used in many applications (see Fig. 4.6). The transfer
function and impulse response are given in the following equations.
for |ffg| < 1
1
Hrc (f ) =
f
( fg (1 ))) for 1
cos2 ( 4
0
for 1 +
hrc (t) = 2fg si(2fg t)
cos(2fg t)
1 (4fg t)2
|f |
fg
|f |
fg
<1+
(4.34)
(4.35)
This lter is likewise frequency-limited, has however in contrast to the ideal low-pass a smooth,
cosine-shaped transition between pass and stop band. In a limit, when 0 this lter is an ideal
low pass lter.
Stronger decaying of impulse response hrc (t) can be achieved by these less abrupt transition
between pass and stop band. The steepness at which the envelope declines is a function of (for
= 0 with 1/t and for = 1 with 1/t3 ). For > 0 this decay is strong enough to achieve a
sucient delimitation in time and frequency for most applications.
The following example demonstrates how causality and time limit can be achieved by shifting
and chopping

t 5Tg
with Tg = 1/fg
h(t) = hrc (t 5Tg ) rect
(4.36)
10Tg

hrc (t 5Tg ) for 0 < t < 10Tg thereby causal
(4.37)
=
0
otherwise
thereby time limited
In signal processing such chopping is called windowing and allows the description of signals
which can be observed in a limited time interval. However, for example, the calculation of the
Fourier transform is practically impossible as it requires an integration over the time period
< t < . A measurement of a signal can take place only in a nite period. Expected eects
regarding windowing are treated in Exercise 6.
5
In order to not to distort signals unnecessarily with ltering (detection made more dicult), the linear phasing is
particularly required in communications technology for lters. This leads also to equal requirement of symmetrical
impulse response of the lter. Since only causal systems are realizable, this impulse response must be shifted
accordingly in time.
30
H(f )
h(t)
2fg
fg
fg
2f1g
1
2fg
a) Transfer function H(f ) and impulse response h(t) of ideal low pass lters
Hrc (f )
hrc (t)
2fg
fg
fg
2f1g
1
2fg
b) Transfer function Hrc (f ) and impulse response hrc (t) of Raised Cosine Filters ( = 0.3)
Figure 4.6: Comparison between ideal low pass and Raised Cosine Filter
4.6
Central Limit Theorem
The central limit theorem states that the distribution function Fn of the sum of n independent
random variables Xi converges for n towards a normal distribution. There exist a set of
theories, which under all dierent assumptions state about the convergence [BHPT95], [BS96]. A
quite general form of the central limit theorem is Ljapunovs theorem, which is described here
briey.
Theory of Ljapunov
Conditions:
Xi independent,
not necessarily identically distributed
i = E{Xi } = nite
i2 = E{(Xi i )2 } nite > 0

n

Bn
3
lim
bi
= 0 with Bn =
n
i=1

and bi = E |Xi i |3

n

and =
i2
i=1
If
n
1
Z=
(Xi i )
i=1
(4.38)
31
4.6 Central Limit Theorem
belongs to normalized random variable X =
!n
Xi , then:
z
t2
1
e 2 d t
lim Fn (z) = (z, 0, 1) =
n
2
i=1
(4.39)
I.e. the distribution function Fn (z) converges asymptotically towards the standardized normal
distribution (z, , )|=0, =1 = (z, 0, 1). With
"
#
n

1
Z = !n
i
X
(4.40)
2
i=1 i
i=1

n
n

2

i Z +
i
X=
follows
The random variable

X =
!n
2
2
and D [X] = i=1 i .
!n
i=1
i=1
(4.41)
i=1
Xi is likewise asymptotically normalized with E[X] =
!n
i=1
The interpretation of the central limit theorem exists as follows: If a random variable can be understood as a sum of large number of independent random variables, of which
each supplies only an insignicant contribution to the total sum, then this random variable is
approximately normal distributed[BS79].
32
5
5.1
5 BANDPASS SIGNAL
Bandpass Signal
Up and Down Conversion of Real Baseband Signals
When signals are dened as low pass or baseband signal s(t), the following relation applies for the
spectra

any for |f | fg
S(f ) =
(5.1)
0
for |f | > fg
The direct transmission of baseband signals, as described in the Article 6.1, is not meaningful.
Baseband signals are therefore shifted to a higher frequency band by multiplication with a carrier
signal sc (t) = cos 2fc t. Such a signal operation is generally called mixing and particularly here
as upward mixing. The baseband signal should be regarded for the time being as real valued. As
a result a bandpass signal is obtained (see Fig. 5.1)

sBP (t) = Re s(t) ej 2fc t
(5.2)
(5.3)
= s(t) cos 2fc t

1
1
(5.4)
(f + fc ) + (f fc )
SBP (f ) = S(f )
2
2

1
(5.5)
= S(f + fc ) + S(f fc )
2
Since the baseband signal s(t) was assumed as real valued, the amplitude spectrum |S(f )| is an
even function (symmetric over f = 0) and thus the amplitude spectrum of the bandpass signal is
local symmetric around f = fc .
|S(f )|
|SBP (f )|
fg
fg
fg,o
fc
fg,u
fg,u
fc
fg,o f
Figure 5.1: Spectrum of a real baseband signal (left) and corresponding spectrum of the upmixed bandpass signal with fg,u = fc fg and fg,o = fc + fg (right)
So for bandpass signals

SBP (f ) =
any for fg,u |f | fg,o

0
for fg,u > |f | > fg,o
(5.6)
Such a rectied bandpass signal can be radiated over an antenna. In the receiver, of course, once
more the conversion of the signal to baseband is necessary. Consequently the bandpass signal is
supplied again to a mixer, where it is multiplied by the same carrier signal, likewise in the sender.
r(t) = sBP (t) cos 2fc t
(5.7)
33
5.1 Up and Down Conversion of Real Baseband Signals
= s(t) cos 2fc t cos 2fc t

1
1
= s(t) + s(t) cos 22fc t
2
2

1

= s(t)
u(t) = r(t)
2
|f |<fc
(5.8)
(5.9)
(5.10)
The received signal r(t) contains both the original baseband signal s(t) and two images at 2fc .
Therefore the baseband signal s(t) can be recovered by an ideal low pass ltering with a critical
frequency of fc (see Section. 5.2, 5.3). These ltering is symbolized as |f |<fc .
|SBP (f )|
A
fg,o
fc
fg,u
fc
fc
fg,o
|R(f )|, |U (f )|
Low-pass
2fc
fg,u
fg
fg
fc
2fc
Figure 5.2: Spectrum of the received real bandpass signal (above) and pertinent spectrum of
the signal r(t) at the output of the mixer or u(t) at the output of the low-pass
(below).
Sender
s(t)
Receiver
sBP (t)
cos 2fc t
r(t)
LP
u(t) = 12 s(t)
cos 2fc t
Figure 5.3: Up and down mixture of a real baseband signal s(t) with ideal carrier synchronization
During a radio transmission senders and receivers are spatially separated from each other and
one can not guarantee in advance that the oscillators run synchronously in phase for the up
and downward mixing. Both oscillators operate independently and therefore there exists a small
frequency oset f and an unknown initial phase dierence 0 between both carrier signals.
sc,S (t) = cos 2fc t
sc,E (t) = cos(2(fc f )t 0 )
= cos(2fc t (2f t + 0 ))

carrier signal in the sender

carrier signal in the receiver
(5.11)
(5.12)
(5.13)
(t)
34
5 BANDPASS SIGNAL
The down-converted received signal u(t) results thereby in the case of non-ideal carrier synchronization to (see Fig. 5.4)
r(t) = sBP (t) sc,E (t)
= s(t) cos(2fc t) cos(2fc t (t))
1
1
= s(t) cos((t)) + s(t) cos(22fc t (t))
2
2

1

= s(t) cos (t)
u(t) = r(t)
2
|f |<fc
(5.14)
(5.15)
(5.16)
(5.17)
A time-varying s(t) factor cos (t) is impressed to the baseband signal. Particularly a phase oset
Sender
s(t)
Receiver
sBP (t)
cos 2fc t
r(t)
LP
u(t) = 12 s(t) cos (t)
cos(2fc t (t))
Figure 5.4: Up and down mixture of a real low-pass signal s(t) with non-ideal carrier synchronization
of n/2 aects adversely, because then cos 0 and u(t) goes to zero (buried in noise ). The
carrier signal in the receiver must be synchronized thus to the phase of the carrier in the sender.
The simplest version is to add the carrier sc,S (t) of a certain level to the signal sBP (t) and then
transmit. In the receiver, the carrier can be regenerated with a very narrow bandpass lter and
then used for the down mixing. The disadvantage of this procedure is however the worsened power
balance, because the transmitting power is not available only in the user signal sBP (t), but is rather
divided in user and carrier signal. It is implemented with analog modulation methods
(Amplitude modulation AM).
A further possibility is to insert certain known data (training sequences) in the baseband signal
s(t). Then try to detect these training sequences in the receiver and to estimate the frequency
oset. It must be guaranteed before that the received signal u(t) cannot degrade to zero. It is
implemented with digital modulation methods.
In addition the following consideration: The problem with a phase oset of n/2. If however,
with a phase shift of /2 is applied into the carrier cos(2fc t (t) + /2) before mixing, then
the signal u(t) can be received error free despite the phase oset of n/2. This transposed
carrier corresponds to a sine carrier sin(2fc t (t)). So the bandpass signal sBP (t) can be
down converted with two carriers in parallel (see Fig. 5.5). So the signal down converted with
a cosine carrier is known as (in phase component) uI (t) and the one down converted with sinus
carrier is the (quadrature phase component) uQ (t). Therefore the name I-Q down mixing comes
for this process 6 .
After mixing the signals rI (t) and rQ (t) arise as
rI (t) = sBP (t) cos(2fc t (t))
6
rQ (t) = sBP (t) sin(2fc t (t))
Often also the term I-Q Demodulation or quadrature demodulation is used.
(5.18)
35
5.1 Up and Down Conversion of Real Baseband Signals
Receiver
rI (t)
LP
Sender
s(t)
sBP (t)
uI (t) = 12 s(t) cos (t)
cos(2fc t (t))
sin(2fc t (t))
cos 2fc t
rQ (t)
LP
uQ (t) = 12 s(t) sin (t)
Figure 5.5: I-Q down mixing of the real bandpass signal sBP (t)
= s(t) cos(2fc t) cos(2fc t (t))
= s(t) cos(2fc t) sin(2fc t (t))

1
= s(t) cos((t)) + cos(22fc t (t))
2
(5.19)

1
= s(t) sin((t)) + sin(22fc t (t))
2
(5.20)
The images at 2fc are ltered here by a low-pass, and the received signals result uI (t) and uQ (t)
in the baseband as

uI (t) = rI (t)
uQ (t) = rQ (t)
(5.21)
|f |<fc
|f |<fc
1
= s(t) sin (t)
2
1
= s(t) cos (t)
2
(5.22)
One may interpret the inphase and quadrature component (both real) of the received signal as
real and imaginary part of a complex signal
r(t) = rI (t) + j rQ (t)
j(2fc t(t))
= sBP (t) e

1
= s(t) ej (t) 1 + e j 22fc t
2
(5.23)
(5.24)
(5.25)
and after low-pass ltering as complex baseband signal

u(t) = uI (t) + j uQ (t)

1
= s(t) cos (t) + j sin (t)
2
1
= s(t) ej (t)
2
(5.26)
(5.27)
(5.28)
the transmitted baseband signal s(t) can be represented as pointers in the complex number plane.
This pointer has an amplitude of s(t) and an argument of (t) (see Fig. 5.6). As s(t) is real
valued, it represents the real part of the axis in the coordinate system of the sender. The I
and Q-components uI (t) and uQ (t) stretch against the coordinate system of the receiver. The
36
5 BANDPASS SIGNAL
Im {s(t)}
uQ (t) = Im {u(t)}
Re {s(t)}
Coordinate system
of the receiver
(t)
uI (t) = Re {u(t)}
Coordinate system
of the sender
Figure 5.6: Phase diagram of the baseband signal s(t)
phase dierence between the carrier signals in the sender and receiver leads to a twist of the two
coordinate systems exactly around this phase dierence (t). The synchronization of the receiver
with the sender means thus the harmonization two coordinate systems. During transmission of
digital signals (see Chapter 8) a wrong synchronization may lead to a false perspective over
the received data and nally to an erroneous conclusion. Thus e.g. with = a transmitted +1
would be interpreted as 1. During transmission of analog signals wrong synchronization leads to
amplitude uctuations in the demodulated received signal.
Here it becomes also clear why during asynchronous down mixture the multiplication with only
one carrier cos(2fc t + (t)) is not sucient. The received signal u(t) = 12 s(t) ej (t) would only
comprise the projection on the real axis uI (t) = 12 s(t) cos (t) instead of the entire send signal. In
contrast by I-Q down conversion, always a constant signal power
1
Pu = |u|2 (t) = s2 (t)
4
(5.29)
exists in the receiver.

can be estimated by using a known
From the complex received signal u(t) the phase dierence (t)
training sequence (in digital modulation). By a complex multiplication afterwards with e j (t)
,
the coordinate system in the receiver is rotated back and is then identical to the coordinate
system of the sender with ideal carrier synchronization (t)
= (t) (see Fig. 5.7).

j sin (t))}
(5.30)
= Re {(uI (t) + j uQ (t)) (cos (t)
Re u(t) e j (t)
= uI (t) cos (t)
+ uQ (t) sin (t)
(5.31)
1
= s(t) cos((t) (t)
(5.32)
)

2
(t)
37
5.2 Up and Down Conversion of Complex Baseband Signal
For ideal carrier synchronization, i.e. an approximate error (t) = 0, gives

1

Re u(t) e j (t) = s(t)
2
(5.33)
uI (t) = 21 s(t) cos (t)
cos (t)
sin (t)
uQ (t) = 12 s(t) sin (t)
1
s(t)
2
Figure 5.7: Principle of the phase correction for real baseband signals in the receiver ((t) =
0)
5.2
Up and Down Conversion of Complex Baseband Signal
In general, the baseband signal s(t) should be regarded as complex valued. Whereas Re {s(t)} and
Im {s(t)} indicate the quadrature components of the baseband signal s(t).
sI (t) = Re {s(t)}
sQ (t) = Im {s(t)}
s(t) = sI (t) + j sQ (t)
in phase component
quadrature phase component
complex base band signal
(5.34)
(5.35)
(5.36)
For the up mixing in this case gives (compare with (5.3))

= Re {(sI (t) + j sQ (t)) (cos 2fc t + j sin 2fc t)}
= sI (t) cos 2fc t sQ (t) sin 2fc t
(5.37)
(5.38)
(5.39)
This means that to each real bandpass signal one complex baseband signal can be assigned or
equivalently two real valued baseband signals which can be independent and represent real and
imaginary part of the complex baseband signal.
This process is called I-Q up conversion (see Fig. 5.9) 7 .
As the signal sBP (t) is real valued, the spectrum SBP (f ) is symmetrical about f = 0. Since for
complex signals s(t) the symmetry of the corresponding amplitude spectrum |S(f )| generally does
7
One can regard this up mixing also as a modulation procedure (quadrature amplitude modulation QAM),
i.e. both the quadrature components sI (t), sQ (t) are independent signal source. The modulation product sBP (t) is
then called QAM signal. (see Digital modulation methods).
38
5 BANDPASS SIGNAL
not hold, the local symmetry of the amplitude spectrum of the BP signal around f = fc also
does not hold. (see Fig. 5.8).

1
1
1
1
SBP (f ) = SI (f )
(f + fc ) + (f fc ) SQ (f )
(f fc ) (f + fc )
(5.40)
2
2
2j
2j

1
= SI (f + fc ) j SQ (f + fc ) + SI (f fc ) + j SQ (f fc )
(5.41)
2
With SI (f ) + j SQ (f ) = S(f ) and SI (f ) j SQ (f ) = SI (f ) j SQ (f ) = S (f ) follows (see
also eqn. 3.47)

1
(5.42)
SBP (f ) = S ((f + fc )) + S(f fc )
2
|S(f )|
A
fg
fg
|SBP (f )|
A
fg,o
fc
fg,u
fc
fc
fg,o
|R(f )|, |U (f )|
Low-pass
2fc
fg,u
fg
fg
fc
2fc
Figure 5.8: Spectrum of the complex baseband signal s(t) (above), the real bandpass signal
sBP (f ) (middle) and the signal r(t) at output of the mixture or u(t) at output of
the low-pass (below)
The real bandpass signal sBP (t) can be transmitted e.g. over a radio link. Then the I-Q down mixing back into the baseband (quadrature demodulation) takes place in the receiver. (see eqn. (5.24)).
r(t) = sBP (t) e j(2fc t(t))

= Re s(t) ej 2fc t e j(2fc t(t))
(5.43)
(5.44)
39
5.2 Up and Down Conversion of Complex Baseband Signal
Sender
sI (t)
cos 2fc t
sin 2fc t
sQ (t)
Receiver
sBP (t)
rI (t)
LP
uI (t)
cos(2fc t (t))
sin(2fc t (t))
rQ (t)
LP
uQ (t)
Figure 5.9: I-Q up and down mixing of a complex baseband signal s(t) = sI (t)+j sQ (t) (Quadrature modulator and quadrature demodulator)
With Re {z} = 21 (z + z ) (see Exercise 1) gives

1
s(t) ej 2fc t +s (t) e j 2fc t e j(2fc t(t))
2

1
=
s(t) ej (t) +s (t) e j 22fc t ej (t)
2
r(t) =
(5.45)
(5.46)
Consideration of the I-Q down mixing as multiplication with a complex carrier signal e j 2fc t(t)
shows, with the help of corresponding fourier transformation, that the spectrum R(f ) of the
signal results r(t) at the output of the mixer originates from a left shift of the spectrum of
the bandpass signal sBP (t).(see Fig. 5.8). With the approximation (t) = const yields the
spectrum

1
R(f ) =
(5.47)
S(f ) ej +S ((f + 2fc )) ej
2
The transmitted baseband signal s(t)can be recovered by a low-pass lter with the critical frequency fc (see Section 5.1).

(5.48)
u(t) = r(t) |f |<fc
1
= s(t) ej (t)
2
(5.49)
So the inphase and quadrature components are

1
sI (t) cos (t) sQ (t) sin (t)
2

1
uQ (t) = sQ (t) cos (t) + sI (t) sin (t)
2
uI (t) =
(5.50)
(5.51)
In Section 5.1 the problem of the carrier synchronization was discussed in details. Also the correction of the phase mismatch (t) is necessary during the transmission of complex baseband signals,
in order to interpret the transmitted information correctly. The phasor diagram in Fig. 5.10 should
clarify this once more.
The carrier synchronization can also be implemented here in two ways. 1.) The carrier signal of
certain level is added to the bandpass signal of the sender and can be regenerated in the receiver
40
5 BANDPASS SIGNAL
uQ (t) = Im {u(t)}
Im {s(t)}
Re {s(t)}
Coordinate system
of the receiver
(t)
uI (t) = Re {u(t)}
Coordinate system
of the sender
Figure 5.10: Phasor diagram of the baseband signal s(t)
by a narrow bandpass lter, i.e. the down convertion follows synchronous in phase analog
modulation. 8 2.) The down convertion is executed independent of the carrier signal of the sender
(phase asynchronously) and the correction takes place additionally with estimation of the phase
(rotating back of
misalignment (t)
and multiplication of the received signal u(t) with e j (t)
the coordinate system /phasors) (see Fig. 5.11) Digital Modulation.
1
= s(t) ej((t)(t))
u(t) e j (t)
2
1
= s(t) ej (t)
2
1
= s(t)
for (t) = 0
2
(5.52)
(5.53)
(5.54)
So the baseband signals sI (t) and sQ (t) can be transferred in the same frequency band and
separated in the receiver again.
u(t) = 12 s(t) ej (t)
1
s(t)
2
e j (t)
Figure 5.11: Principle of phase correction for complex baseband signals in the receiver ((t) =
0)
8
This corresponds to an analog QAM, which is however not applicable in practice; since even a small phase
misalignment (t) leads to a crosstalk between both source signals. Instead one uses single-sideband modulation
with two independent sidebands ISB, see also Section 6.3.6
41
5.3 General Bandpass Signal Equivalent Low-Pass Signal
5.3
General Bandpass Signal Equivalent Low-Pass Signal
In general sinusoidal carriers are used for transmitting information. These can be generally modulated in amplitude A, frequency f and phase (see Section 6.2). A general bandpass signal can
be represented as
sBP (t) = A(t) cos(2f (t)t + (t))
(5.55)
the carrier cos(2fc t) can be rearranged into

= A(t) cos(2fc t + 2f (t)t + (t))

(5.56)
(t)

= Re A(t) ej (t) ej 2fc t

s(t)
(5.57)
Carrier
Since the carrier signal does not contain any information, it can be omitted while considering
signal transmission. The (complex) low-pass signal s(t) which equivalently represents bandpass
signal sBP (t) is given by
s(t) = A(t) ej (t)
= A(t) cos (t) + j A(t) sin (t)
= sI (t) + j sQ (t)
(5.58)
(5.59)
(5.60)
This low-pass signal (baseband signal) represents the complex Envelope (complex envelope
amplitude and phase) of the real bandpass signal.
A bandpass system can be thus be represented exactly with an equivalent low-pass system. Computer simulation of the time-continuous systems requires sampling at high enough sampling frequencies (sampling theorem). For example, a time step of T 1 ns is necessary for the simulation
of a radio transmission according to GSM standard in the 900 MHz band. The resulting simulation
times would be very large. Since the GSM bit rate (270,83 kbit/s) is low and the (radio) channel
characteristics change slowly, the system in the baseband can be simulated with a much smaller
sampling rate than the original bandpass signal.
42
6 ANALOG MODULATION
Analog Modulation
6.1
Introduction
Modulation is understood as imposing a source signal on a carrier signal. The source signal can
be a low-pass signal or baseband signal. The goal is to transmit the source signal over a certain
channel. The channel is an abstract model for describing the transmitting medium, e.g. coax cable,
LWL-cable, radio link, for wire or free space transfer of electromagnetic waves.
Source
Mod.
Channel
Demod.
Sink
Figure 6.1: Transmission of source signal over a channel
At the beginning it requires to clarify why should modulation be used. Principal reasons for this
are
the adjustment of the source signal into the transmitting medium (a source signal with a dc
component cannot be directly transferred e.g. over a radio link.),
the possibility to transfer several independent source signals over the same medium
(e.g.frequency multiplex - Shifting the source signal into the desired frequency range
(e.g. USW-broadcasting, television channel); Time multiplex - temporal progressive rate of
several source signals (each source has, in each case, access to the channel for a certain period
of time )).
At the same time
to increase the quality of the transfer (e.g. signal-to-noise ratio SN R)
to minimize the transmitting power,
to minimize the structural complexity of sender and recipient.
If the transmission has to be implemented over a radio link, it is to be noted that an eective
radiation is possible if the wavelength of the signal lies in the order of magnitude of the antenna.
=
c
f
or,
/m =
300
f /MHz
(6.1)
For example, an antenna of length 1 m can radiate signals in the frequency range 100 MHz-1 GHz.
In principle, low-pass signals cannot be transmitted as radio signals.
43
6.2 Modulation Schemes
6.2
Modulation Schemes
As carriers, particularly application of sinusoidal signals is found in communications (conceivable

however also dierent signal forms, e.g. pulse-type carriers are with pulse modulation methods).
The sinusoidal carrier signal generally has the form (see Fig. 5.3)
sc (t) = A0 cos(2f0 t + 0 )
(6.2)
and contains thereby three parameters A0 , f0 , 0 ; which can be used for information transmission
(see Fig. 6.2). One parameter (or perhaps more) can be changed by the modulation to be proportional to the source signal (modulation signal). The source signal is an analog signal corresponding
to analog modulation and depends on selected parameters particularly of
Amplitude Modulation
AM:
A0 A(s(t)),
Frequency Modulation
FM:
f0 f (s(t)) and
Phase Modulation
PM:
0 (s(t)).
s(t)
Source
Modulator
sBP (t) = A(t) cos (t)
sc (t) = A0 cos(2f0 t + 0 )
Figure 6.2: Principle of modulation with sinusoidal carrier
6.3
Amplitude Modulation
The source signal s(t) is imposed with the amplitude of carrier signal sc (t). At the beginning
classical double sideband amplitude modulation(DSB) with carrier will be regarded. This type
of modulation is found e.g. in the medium wave broadcast application. The source signal s(t) is
assumed to be zero-mean with symmetrical amplitude distribution.
6.3.1
Description in Time Domain
The source signal s(t) together with an overlaid dc component A0 forms the modulated amplitude
A(t) of the carrier signal sc (t), is also called envelope of the carrier sc (t).

s(t)
sBP (t) = (A0 + s(t)) cos 2fc t = A0 1 +
cos 2fc t
(6.3)

A0
A(t)
The ratio
max(|s(t)|)
smax
=
=m=
(6.4)
A0
A0
denes the modulation factor or modulation index and it may vary the measure of carrier amplitude. If the modulation index is selected as 1, ( A0 smax 0), it is guaranteed that the
44
6 ANALOG MODULATION
information is contained in the amplitude of the carrier. The demodulation becomes thereby very
simply (see envelope detector, e.g. medium wave broadcast). As opposed to that, if the modulation
index is selected as > 1 ( A0 smax < 0), then the information is contained in the amplitude
and phase of the carrier. Where the baseband signal A(t) = A0 + s(t) has a zero crossover (sign
change), a rapid phase change in the modulation product sBP (t) appears (see Fig. 6.3). During
the demodulation, therefore also the carrier phase must be included (see synchronous demodulator,
e.g. short wave amateur radio).
A0 + smax
sBP (t)
sBP (t)
A0 + smax
A0
A0
A0 smax
A0 smax
Figure 6.3: Amplitude modulated transmitting signal sBP (t) with = 0.84 (left) and = 1.65
(right)
6.3.2
Description in Frequency Domain
The Fourier transformation of eqn. (6.3) yields

SBP (f ) = F {sBP (t)} = F {A0 cos 2fc t + s(t) cos 2fc t}
1

A0
=
(f + fc ) + (f fc ) + S(f + fc ) + S(f fc )

2
2
Carrier
(6.5)
(6.6)
Mixer product
In the case of double sideband AM, the spectrum of the bandpass signal arises as a result of
shifting the low-pass spectrum of the source signal by the carrier frequency fc , in addition to
the carrier spectral line (see Fig. 6.4).
Example: 1-Ton-Modulation. The Spectrum SBP (f ) is shown in Fig. 6.5 (see also Exercise 21).
s(t) = Am cos 2fm t
sBP (t) = A0 (1 + cos 2fm t) cos 2fc t
SBP (f ) =
(6.7)
with =
Am
A0
A0
A0
(f + fc ) + (f fc ) +
(f + (fc + fm ))+
2
4

(f + (fc fm )) + (f (fc fm )) + (f (fc + fm ))
(6.8)
(6.9)
45
6.3 Amplitude Modulation
|S(f )|
S0
fg
fg
|SBP (f )|
lower
Sideband
carrier
upper
Sideband
A0
2
B = 2fg
S0
2
fc
fc
Figure 6.4: Spectrum S(f ) of the source signal s(t) (above) and Spectrum SBP (f ) of the amplitude modulated Bandpass signal sBP (t) (below)
|S(f )|
Am
2
fm
fm
|SBP (f )|
A0
2
mA0
4
fc fm fc fc + fm
fc fm
fc
fc + fm
Figure 6.5: Spectra of the source and transmitting signals for a 1-Ton-Modulation ( = 0.67)
6.3.3
Power Balance
The signal power of the transmitted signal sBP (t) is composed of the signal power of the two
sidebands PSB and that of the carrier signal Pc together. The two sidebands (shifted LP-spectrum
of the source signal) represent the information signal. The carrier signal however does not contain
information about the source signal. The total transmitted power is given as
PAM = Pc + PSB
(6.10)
The eciency AM is given as

AM =
PSB
PAM
(6.11)
46
6 ANALOG MODULATION
With help of the above example of 1-Ton modulation, the eciency AM can be calculated as a
function of the modulation index .
2 A20
A20
+22
=
4
16
PAM = 2
AM =
A20
1+
A20
2

Carrier power Pc
2 A20
4

Sideband power PSB
(6.13)
2
2
1+
(6.12)
(6.14)
2
2
Full rejection of the modulator can be achieved by = 1 only with an eciency of AM = 33 %.

(The modulation with a square wave signal provides the highest eciency of AM = 50 %. This
case is only theoretical, where the square wave signal needs an innitely high bandwidth.)
6.3.4
AM-Modulators
Fig. 6.6 shows the principle of an AM-Modulator. It is assumed that the source signal is normalized
to smax ( s(t) [1; +1]).
s(t)
A0
sBP (t)
cos 2fc t
Figure 6.6: Principle of an AM-Modulators
6.3.5
AM-Demodulators
The recovery of the source signal s(t) from the modulated band-pass lter signal sBP (t) is called
demodulation. With an inspection of AM in the frequency range, it is clear that the source signal
s(t) contained in the modulation product sBP (t) is not located in its original frequency but shifted
with the carrier frequency. Thus a frequency conversion is necessary in the receiver in order to
bring the source signal back into the baseband from the carrier frequency. This is always done
with the help of amplitude modulation with same carrier signal (see also Section 5.1). Thus in
the receiver for the demodulation of the received signal sBP (t) (distortion-free transmission it is
assumed here) the carrier signal is necessary once more. With regard to double sideband amplitude modulation with carrier ( < 1), the carrier signal with a sucient amplitude is contained
in the modulation product sBP (t) and can be used directly for demodulation (see Envelope demodulator). In Section 6.3.6 further versions of AM are presented, in which the carrier signal is
only is partly ( > 1) or not at all, contained in the modulation product. The carrier signal must
be supplied externally to the demodulator (see Synchronous demodulator). The book Analoge
Modulationsverfahren[Mau92] is recommended for interested readers.
47
6.3 Amplitude Modulation
Envelope Detector The envelope detector

is the simplest circuit for the demodulation of an amplitude modulated carrier signal (see
Fig. 6.7 and Exercise 21) and assumes that the source signal is contained in the envelope of
the carrier wave exclusively ( < 1),
consists of a half-wave rectier, a RC low-pass and a RC high-pass;
is practically found in each LW-, MW-, SW-radio;
forms together with an antenna and a Tank circuit to the transmitter selection, to the
detector receiver as the rst and simplest radio;
needs no external supply of the carrier signal;
is an incoherent AM demodulator, where phase position of the received signal is insignicant.
u(t)
sBP (t)
s(t)
rectier
low pass
high pass
Figure 6.7: Circuit diagram of an envelope demodulator (detector recipient)
The adjoining amplitude modulated signal is rectied by means of the diode (see Fig. 6.8 a)). The
following RC low-pass reconstructs the approximate envelope of the carrier wave (see Fig. 6.8 b)).
The RC high-pass removes the dc component. (see Fig. 6.8 c)).
uDi (t)
u(t) s(t)
uT P (t)
t
a) after half-wave rectication
t
b) after LP ltering
t
c) after HP ltering
Figure 6.8: Processing in dierent places at the envelope demodulator
48
6 ANALOG MODULATION
Synchronous Demodulator The Synchronous demodulator

in the comparison with envelope demodulator somewhat more complex,
shows however smaller distortions of the demodulated Signal,
needs smaller input voltages, is thus more sensitive,
can also be used with over modulation, i.e.. > 1 (reduced carrier),
needs the phase position of the carrier,
multiplies the received signal by the phase of the carrier signal and mixes so that the source
signal gets back into the baseband (see Section 5.1),
is also well-known as product demodulator.
Fig. 6.9 shows the principle of a synchronous demodulator. The carrier signal must be regenerated
Synchronous down mixer
sBP (t)
LP
HP
u(t) s(t)
Carrier retrieval
narrow BP with fc
PLL
Figure 6.9: Principle of the Synchronous demodulator (with AM with reduced carrier)
and strengthened the received signal sBP (t) from phase position. This is done with the help of
a very narrow bandpass lter and PLL circuits (phase locked loop). The Low-pass lter follows
the mixer removes those images by 2fc (see Section 5.1). The high-pass lter removes the dc
component from the output signal.
Still there exist some further versions, e.g. quadrature demodulator for single-sideband AM.
6.3.6
Dierent Types of Amplitude Modulation
So far mainly double sideband is considered with carrier ( < 1), like it is found e.g. in the medium
wave broadcast applications. However still some other versions of AM exist. Reasons to use other
AM methods are [Mau92]
to improve power balance by reducing or suppressing the carrier,
need to reduce the bandwidth (in modulation with a real source signal, the upper and lower
sideband contains the same information, see Section 5) single sideband modulation.
Through these measures, however, reference values for frequency and phase of the carrier are lost
which must be known during the demodulation.
49
6.4 Phase and Frequency Modulation (Angle Modulation)
The AM methods are dierentiated with regard to modulated sidebands and carrier power.
Sidebands
Double sideband DSB
Vestigial sideband VSB
Single sideband SSB with the distinction whether upper or lower sideband is transferred
upper sideband USB, lower sidebandLSB
Independent sideband ISB
Carrier
with carrier
with reduced carrier -RC
with suppressed carrier -SC
6.4
Phase and Frequency Modulation (Angle Modulation)
If the source signal s(t) is impressed on the frequency of the carrier sc (t), one speaks of frequency
modulation FM. On the other hand, impressing the source signal on the phase of the carrier
is phase modulation PM. In both cases, the frequency as well as the instantaneous phase of the
carrier is modied modulation product. One can generally speaks therefore of angle modulation
FM [Mau92].
6.4.1
The Modulation product has the general form

sF M (t) = A0 cos (t) = A0 cos s(t)
(6.15)
With the fundamental relationship (angular speed (t) results from the derivative of the angle
(t) with respect to time t)
d
(t) = 2f (t) =
(t)
(6.16)
dt
can be also written as
t

2f ( ) d + (t0 )
with t0 < t
(6.17)
sF M (t) = A0 cos
t0
Frequency modulation FM:

f (t) = fc + kF M s(t)
with the frequency shift F = kF M smax
(t) = 2fc t + 2 kF M
s( ) d + (t0 ) 2fc t0
t0

carrier
(6.18)
mod. Phase
(6.19)
50
6 ANALOG MODULATION
Thus the modulation product results to

t

s( ) d + 0
sF M (t) = A0 cos 2fc t + 2 kF M
(6.20)
t0
Phase modulation PM:

(t) = 2fc t + kP M s(t) + 0
kP M
f (t) = fc +
s(t)
2
Thus the modulation product yields
with the phase shift = kP M smax

d
with s(t)
=
s(t)
dt

sP M (t) = A0 cos 2fc t + kP M s(t) + 0
(6.21)
(6.22)
(6.23)
The modulation product sF M (t) or sP M (t) has a constant Amplitude. Both the frequency shift F
and phase shift are usually given by international agreements (e.g. USW-broadcast: FM with
F = 75 kHz). Those Amplication factors kF M , kP M must be selected accordingly.
Example: 1-Ton Modulation
s(t) = Am cos 2fm t
t

for 0 < t
(6.24)
Am cos 2fm d + 0

t0 =0

2 kF M Am
= A0 cos 2fc t +
sin 2fm t + 0
2fm

F
sin 2fm t + 0
= A0 cos 2fc t +
fm

sP M (t) = A0 cos 2fc t + kP M Am cos 2fm t + 0

= A0 cos 2fc t + cos 2fm t + 0
(6.25)
(6.26)
(6.27)
(6.28)
Here the common view is underlined again by FM and PM. Both procedures supply the same
modulation product with the 1-Ton modulation (see Fig. 6.10). Here frequency and phase shifts
are connected by the following context:
F
=
(6.29)
fm
A distinction of FM and PM becomes only possible with modication of the frequency fm of the
source signal s(t). Despite this, both procedures diers in modulation gain, bandwidth, modulator
and demodulator circuits, noise behavior etc.. Whereupon is not to be further explained in the
context of this lecture. Further readings may be continued in the lecture Rundfunksysteme).
6.4.2
Frequency and phase modulation are non linear modulation method in contrast to AM.
While the modulation product (bandpass signal) of the AM is obtained simply by shifting
the spectrum (low-pass) of the source signal around the frequency fc , the spectrum of the
modulation product by angle modulation is not so easily denable. Fundamentally it can be said,
however, that the necessary bandwidth of the modulation product sF M (t) is usually larger than
one of the modulating source signal (see Fig. 6.11) 9 .
9
Theoretically the bandwidth of the modulation product is yet innite, but practically one can assume the
spectrum outside of a certain bandwidth B in the disappeared noise.
51
sF M (t)
A0
t
A0
Figure 6.10: 1-Ton angle modulated Signal
|S(f )|
B = 2fg
fg
fg
|SF M (f )|
B 2fg
fc
fc
Figure 6.11: Spectrum S(f ) of the source signal s(t) (above) and Spectrum SBP (f ) of the angle
modulated bandpass lter signal sF M (t) (below)
The bandwidth depends thereby on the rejection of the modulator (frequency shift F or phase
shift ). Sucient approximation for practice represents the Carson rule, which is the worst
case consideration (1-Ton modulation), and is usually bigger than the bandwidth required.
FM: BF M = 2(F + fg )
PM: BP M = 2( + 1)fg
(6.30)
(6.31)
Here fg the critical frequency of the source signal. The USW broadcast results in the case of a
frequency shift by F = 75 kHz and a critical frequency of fg >= 15 kHz (mono transmitter), a
necessary bandwidth of BF M = 180 kHz.
For the above example of the 1-Ton modulation the spectrum can be given still quite clearly. Since
frequency and phase shifts over eqn. (6.29) are interconnected, the modulation product can be
generally indicated as (see eqn. (6.26) and (6.28))

(6.32)
sF M (t) = A0 cos 2fc t + sin 2fm t + 0
The diculties in the calculation of the spectrum lie in the fact that here trigonometric functions
emerge again as argument of trigonometric functions. With expansions in powers series derive
from the following relation [SS88]
cos( + x sin ) =
Jn (x) cos( + n)
(6.33)
n=
52
6 ANALOG MODULATION
The modulation product can be now written as (with 0 = 0)
sF M (t) = A0

Jn () cos 2(fc + n fm )t
(6.34)
n=
In addition the Bessel function Jn () of rst type and n-th order is indicated (see Fig. 6.12).
1
J0()
J ()
1
J2()
0.5
J ()
J ()
4
Jn()
0.5
5
6
Phasenhub
10
Figure 6.12: Bessel function Jn () rst type, 0. to 4. Order, It applies additionally Jn (x) =
(1)n Jn (x)
The Fourier transformation from (6.34) yields the spectrum of the modulation product, out of
which exist many innite spectral lines.

A0
Jn () (f + (fc + n fm )) + (f (fc + n fm ))
SF M (f ) =
2 n=
(6.35)
Fig. 6.13 shows the dependency of output spectrum with respect to phase shift (Frequency
shift F = fm ).
6.4.3
Power Balance
Discussion is quite dicult with angle modulation from the point of view of power balance because
the carrier contained in the modulation product is also to be contained in the information which
53
|SF M (f )|
A0
2
fc
BCarson
fc
|SF M (f )|
BCarson
A0
2
fc
= 1
fc
|SF M (f )|
BCarson
A0
2
fc
= 2.4
fc
|SF M (f )|
BCarson
A0
2
fc
= 5
fc
|SF M (f )|
A0
2
fc
= 0.3
BCarson
fc
= 10
Figure 6.13: Spectrum SF M (f ) of the Modulation product sF M (t) in 1-Ton Modulation with
respect to Phase shift
54
6 ANALOG MODULATION
is transmitted. The total operating performance of the Modulation product /transmitting signal
is constant due to the non-changing envelope.
PF M =
A20
2
(6.36)
The total output divides into the power of the sidebands and the carrier power. The allocation
depends on the selected rejection(, F ).
For the 1-Ton modulation again very good numerical values can be indicated.
PF M
A2
A2
= 0 = 0
2
2

J20 ()
+ 2

Carrier power

J2n ()
n=1
(6.37)
Side band power
With the help of the diagram in Fig. 6.12 two values can be read o immediately: Pc = PF M for
= 0 (unmodulated carrier) and Pc = 0 for 2.4.
6.4.4
FM-Modulator
There exists a large variety of modulators and will not be dealt here further ( Lecture Rundfunksysteme). For FM modulators often voltage-controlled oscillators come to the application, e.g.
control of a varactor diode in the tank circuit A) directly over the source signal FM, b) over
the high-pass-ltered (dierentiated) source signal PM (see Fig. 6.14).
a)
b)
s(t)
sF M (t)
VCO
s(t)
s(t)
HP
VCO
sP M (t)
Figure 6.14: FM modulator with the help of a voltage-controlled oscillator (VCO)
Voltage-controlled oscillators are usually named VCO briey. The Oscillator frequency fV CO is
dependent on the adjoining input voltage u(t) (here u(t) s(t) FM or u(t) s(t)
PM, see
Fig. 6.15).
fV CO
fc
umax
umax
u(t)
Figure 6.15: Dependence of the oscillator frequency of a VCO on input voltage u(t)
55
6.4.5
FM-Demodulator
There exist tremendous variety of demodulators here too. Only one version is briey pointed out
here. A possible FM demodulation is the FM to AM conversion with the following AM demodulation. A ank frequency discriminator (ank demodulator) executes such a conversion with the help
of an easily detuned tank circuit (see Fig. 6.16). The impedance Z(f ) of a tank circuit is periodic
|Z(f )|
output signal
sAM (t)
fc
sF M (t)
fr
input signal
Figure 6.16: Frequency discriminator characteristic of a tank circuit
whose maximum value occurs with resonant frequency fr . Beside of the resonant frequency fr
impedance Z(f ) decreases, which can be adjusted by damping of the resonant circuit. In a certain
area of the characteristic this is almost linear and can be used for the FM to AM conversion. In
addition the FM signal can be fed into the resonant circuit. The varying frequency leads to an
impedance modication, which again leads to a change in voltage. Therefore an amplitude modulated signal is present at the output, which can be demodulated like an AM signal (see envelope
demodulator in Section 6.3.4).
The FM to AM conversion can be usually produced by dierentiation of the frequency modulated
signal.

s( ) d
(6.38)
t0
d
sF M (t)
dt
t

s( ) d
= A0 2fc + 2 kF M s(t) sin 2fc t + 2 kF M
t0

sAM (t) =
A(t)
(6.39)
(6.40)
Carrier
56
6 ANALOG MODULATION
6.5
Comparison between Amplitude and Angle Modulation
Properties of Amplitude Modulation AM

very easy method
for < 1 the information of the source signal is exclusively contained in the envelope of the
modulation product demodulation becomes particularly simple (see envelope detector)
for > 1 the information is also contained in the phase of the carrier and must be considered
with the coherent demodulation phase synchronization is necessary (see synchronous
demodulator)
by double sideband modulation with (full) carrier (DSB) puts practically more than 70 %
of the transmitted power in the carrier signal
the level of demodulator output signal is linearly dependent on the input level. Thus it can
result strong amplitude oscillations when uctuation occurs in the transmission path.
the transmission amplier must be linear in a large dynamic range
Properties of Frequency and Phase Modulation FM, PM
the modulation product possesses constant envelope
therefore the transmission amplier can be operated without power reserve
quite noise insensitive(Modulation gain > 1)
higher bandwidth required
Narrow band FM (small frequency, phase shift) behaves like AM
Criteria
AM
FM / PM
Bandwidth
SSB: B = fg
DSB: B = 2fg
Distortions
no distortion from suppression
Requirement of a
transmission system
Application
linear amplitude frequency response required, amplier with

scope-reserve
analog
signal
transmission
with inferior quality, inferior
transmission-bandwidth, simpler
receiver, carrier-frequency technic
PM: B = 2( + 1)fg
FM: = 2(F + fg )
(theoretically B )
increasing distortion with increasing phase shift above threshold
no linear amplitude frequency response required, amplier can be
operated without power-reserve
analog signal-transmission with
better quality
57
Analog-Digital-Conversion
7.1
Motivation
Signals around us are usually continuous both in value and time. Such signals are generally called
analog signals. The direct processing of these analog signals takes place by means of analog systems
(see lecture System Theory). Since however digital systems are less vulnerable against distortions,
universally applicable, parameterized and recongurable, and often oer more possibilities for
signal manipulation, it is meaningful in many cases to implement the processing of analog signals
by means of digital systems. Therefore an Analog-Digital- or Digital-Analog-Conversion of the
signals is necessary. Such conversions are also needed when digital signals are transferred with the
help of analog signals, e.g. over a radio interface. (see Fig. 7.1).
Analog
Signal
ADC
Digital System
e.g. -Processor
DAC
Source/Sink
Digital
Signal
Analog
Signal
Source/Sink
DAC
Analog System
e.g. Radio channel
ADC
Digital
Signal
Figure 7.1: Main applications of AD or DA converter
This paragraph is particularly concerned with possible problems of the analog-to-digital conversion. Hereunder it would clarify, how and which signal property is changed by such conversions,
and whether reconstruction of analog signal is possible again. In regard with practical implementations, error sources should be analyzed and considered for the development of converters.
A digital signal is characterized by discrete in time and discrete in value. In many applications
digital signals are attained from analog signals, by means of an analog-digital converter, where
continuous in time and value analog signal is converted into discrete in time and value (digital)
signal. The discretization in time is called sampling, whereas the discretization in amplitude is
called quantization.
The advantages of digital signal processing are in particular:
The processing is independent of technology and temperature.
The processing of signals in digital systems is usually simpler i.e., determination of signal
manipulations is realizable only with the help of digital signal processing.
The signals can be easily stored in a memory.
7.2
Time Discretization
Time continuous signals s(t) are function of time, i.e., exactly one function value is assigned to
them at each point in time t. Time discrete signals {s(n)} can be thought as sequences, in which
58
ANALOG-DIGITAL-CONVERSION
exactly one value is assigned to each n. At the same time each n is assigned to a certain time point
t. The transformation s(t) {s(n)}, which ideally takes place without any information loss, is
called time discretization in time or sampling.
7.2.1
Dirac-Comb-Function
For a better description of sampling, a new function, the dirac-comb function XT (t) is introduced.
This function consists of equally spaced time-shifted dirac pulses and is dened as follows.
XT (t) =
(t nT )
(7.1)
n=
Since the dirac-comb function is a periodic function with period T , one can extend it into a Fourier
series (see Exercise 16)

1 j 2 n t
XT (t) =
e T
(7.2)
T
n=
Its Fourier transformation can be obtained easily as
F {XT (t)} =
=
1
n
f
T n=
T
(7.3)
1
X 1 (f )
T
T
(7.4)
So it is clear that the spectrum of a dirac-comb function in time domain corresponds again to the
dirac-comb function in frequency domain.
XT (t)
7.2.2
1
X 1 (f )
T
T
(7.5)
Sampling
Sampling means taking of function values (samples) of a time-continuous signal. The sampling
values are combined into a sequence.
s(tn ) {s(n)} with nZ tn < tn+1
(7.6)
The sampling can take place at any successive points in time, however only equidistant sampling
is to be regarded here, since it possesses the most practical meaning.
s(nT ) {s(n)} with T sample period
(7.7)
The sampling process can be modelled by masking (setting to zero) all function values s(t = nT ).
Since after masking of samples the area s(t) d t tends to zero, then the (remaining) sampling
values must be weighted accordingly.
Starting point should be a real sampling or gating circuit with a gate open time of T0 (see Fig. 7.2).
So the sampled signal can be described by
59
7.2 Time Discretization
s(t)
T0
s (t)
T s
= s(t)
rect
n=
tnT
T0
T0
0
2T
3T
4T
Figure 7.2: Input signal s(t) and output signal ss (t) of the real sampler

t nT
T
ss (t) = s(t)
rect
T
T0
0
n=
with T0 < T
(7.8)
The factor T /T0 is essential, so that the window function remains dimensionless (unit less) as well
as the area remains constant independently of sampling rate 1/T and gating time T0 constant. If
one starts making the duration of the sampling pulse shorter and nally aims at T0 0, then the
model of ideal sampler becomes 10 (see Fig. 7.3 and 7.4)
ss (t) = s(t)
T (t nT )
(7.9)
n=
With the help of Dirac comb function introduced above the sampling can now be described as
s(t)
ss (t) = s(t)
T (t nT )
n=
2T
3T
4T
Figure 7.3: Input signal s(t) and output signal ss (t) of the ideal samplers (Dirac impulses can
also be drawn only as symbol, see also Fig. 3.2)
ss (t) = s(t)
T (t nT ) = s(t) T XT (t)
(7.10)
n=
Here T denes the sample period and fs = 1/T is the sample rate.
10
In older versions of the scripts the conventional way in the literature was taken, the
!factor T allowed in time
domain (it appeared therefore in the frequency domain as 1/T ). It yields ss (t) = s(t) n= (t nT ) (see also
the exam in 1998-1999). In principle therefore nothing changes, but working with dimension aicted parameters,
like time t and frequency f , is facilitated by the way they have been selected here. (see Article. 3.2.5). Since these
dimension aicted parameters are worked out in the script consistently, it is only possible to compare s(t) and
ss (t) and later S(f ) and Ss (f ) together and to draw a diagram.
60
s(t)
T
ss (t)
(t nT )
n=
Figure 7.4: Model of an ideal sampler
The sampled signal ss (t) is a time-continuous signal. The multiplication of the signal s(t) with the
dirac comb function T ShT (t) in time domain corresponds to the convolution of the spectra S(f )
and F {T XT (t)}. Thus with eqn. (7.5)
ss (t) = s(t) T XT (t)

Ss (f ) = S(f ) X 1 (f )
(7.11)
(7.12)
Considering the additional important characteristic (3.80) of the dirac delta function yields

n
f
Ss (f ) = S(f ) X 1 (f ) = S(f )
T
T
n=

n
n
S f
=
S(f ) f
=
T
T
n=
n=
(7.13)
(7.14)
By convolution of the original spectrum S(f ) with the periodic spectrum of dirac comb function,
the resulting sampled signal spectra is continued periodically with period fs = 1/T (compare
to (7.14)).

S(f nfs )
with fs = 1/T
(7.15)
Ss (f ) =
n=
Fig. 7.5 claries this duplication of the original spectrum. The signal s(t) was assumed as band
limited low-pass signal with critical frequency fg . The obtained duplicates of the original spectrum
are called images and they do not carry any further information.
!
Ss (f ) =
S(f nfs )
n=
images
S(f )
fs
fg
fg
fs
Figure 7.5: Spectral representation of the sampled signal (fs = 1/T )
7.2.3
Sampling Theorem
In Fig. 7.5 the sampled signal was assumed as band limited low-pass signal. Now naturally the
main questions that arise in digital signal processing and communications technology are whether
61
these hypothesis are necessary and which relation exists between the critical frequency fg and the
sampling rate fs of the signal.
Many researchers dealt with these questions and published the well known sampling theorem in
dierent occasions. The rst publications were in mathematics: e.g. La Vallee Poussin (1908), E.T.
Whittaker (1915). In the communications technology particularly two names should be quoted:
V.A. Kotelnikov (1933) and C.E. Shannon (1949), whereas formerly the sampling theory by Shannon found a broad publicity.
Sampling Theorem A band limited signal with fg can be reconstructed accurately from its
sampled values with the help of sampling series

t
s(t) =
s(nT ) si
n
T
n=
(7.16)
if during the sampling the condition

fs =
1
> 2fg
T
(7.17)
is kept.
The sampling rate fs = 2 > fg is also known as Nyquist rate. From the sampling theory the
following conclusions can be drawn (see Exercise 17, 18):
If the signal s(t) is band limited and if it is sampled under the condition specied in the
sampling theory (7.17), then the signal is uniquely described by the (samples) and can be
reconstructed accurately (7.16).
If the signal s(t) is sampled with a rate which is smaller than the Nyquist rate, then s(t) can
not be suciently described by its samples s(n > T ) and reconstruction of such signal will
produce distortions (errors). This eect is graphically representable in the frequency domain
(see Fig. 7.6). Due to the periodic continuation of the spectrum (f ) with fs , it overlaps
within the space of Ss (f ). Such overlaps is denoted as Aliasing.
Since aliasing error cannot be cancelled, the mentioned condition (7.17) must be guaranteed
for sampling. This is done by an anti-aliasing lter before sampling (see Fig. 7.7). The
anti-aliasing lter is an analog low-pass lter with H(f ) = 0 for f f2s .
A band limited signal s(t), which is sampled exactly with the Nyquist rate, can be accurately
reconstructed only if the phase relation between s(t) and sampling points nT is known.
If a signal is sampled with a rate fs which is larger than the Nyquist rate 2fg , oversampling
occurs. By oversampling one does not gain additional information (oversampling however
brings advantages for technical circuiting quantization noise, lter design).
62
Ss (f )
fs > 2fg
fs
fg
fg
fs
Ss (f )
fs < 2fg
Aliasing
fs
fg
fg
fs
Figure 7.6: Aliasing eect
Anti-Aliasing
s (t)
s(t)
Filter
ss (t)
ADC
Figure 7.7: To avoid aliasing errors an so-called anti-aliasing lter (analog low-pass lter) has
to be provided before the sampler.
7.2.4
Signal Reconstruction Digital-Analog-Conversion
In order to recover a time-continuous (Analog) signal from a time-discrete (Digital) signal, the
signal processing between the sampling points must be reconstructed. Particularly eqn. (7.17)
have to be fullled so that the reconstruction takes place without any error.
Images arise during sampling (see eqn. (7.15)), which are reections of the original spectrum. For
signal reconstruction these images must be removed by suitable ltering. For that purpose an
ideal low pass (analog) lter with a critical frequency fg,T P = fs /2 is used (see Fig. 7.8 and 7.9).
With the application of real lters (the ideal low pass is not realizable), the reconstructed signal
deviates srek (t) more or less strongly from the original signal s(t).
ss (t)
ideal
low pass
srek (t) = s(t)
Figure 7.8: The signal reconstruction takes place with an ideal low pass lter with a critical
frequency fg,T P = fs /2 (often called as reconstructing lters)
The DA converter with an ideal low pass lter with critical frequency fg,T P = fs /2 should be
considered.

f
1
S(f ) = Ss (f ) rect
with fs =
(7.18)
fs
T
63
ideal reconstructing
low pass lter
fs
Ss (f ) = F {ss (t)}
fg
f g fs
2
fs
Srek (f ) = S(f )
fg
fg
Figure 7.9: Spectral representation of the reconstruction of time continuous signal from a timediscrete signal (fg,T P = fs /2, fs = 1/T )
= (S(f ) X 1 (f )) rect(f T )
T
(7.19)
The Fourier transformation in time domain yields the sampling series specied in the sampling
theory (7.16)
"
#

1
t
(7.20)
s(t) = s(t)
T (t nT )
si
T
T
n=
"
#

t
=
(7.21)
s(t) (t nT ) si
T
n=

t
n
(7.22)
=
s(nT ) si
T
n=
64
7.3
Value Discretization
Since digital systems can process only time and value discrete signals, the samples s(nT ) must
be discretized additionally in amplitude. This process is also known as quantization and is a
nonlinear operation. In contrast to the (ideal) time discretization/sampling, the value discretization/quantization cannot be reversed. Therefore after the signal reconstruction in the DA converter
a (as small as possible) deviation from the original signal must be established. (see Fig. 7.10, 7.11).
This deviation is called quantization error.
sq (t)
s(t)
Figure 7.10: Quantization
1
0.8
0.6
0.4
Amplitude
0.2
0
0.2
0.4
0.6
0.8
1
0
0.2
0.4
0.6
Zeit t/T
0.8
1.2
Figure 7.11: Exemplary quantization of a sine signal with pertinent quantization error.
The order of operations, sampling and quantization, is arbitrary, in practice is however rst sampled and then quantized. This is done via a real gate switch with a gating time T0 > 0 (see
Fig. 7.2), followed by a short time integration (RC element) to the averaging of the signal in the
interval T0 and nally quantization of the average value. The quantized sampling value is picked
out and passed to the subsequent system.
65
7.3 Value Discretization
During quantization, the innite scope of the (value continuous) input signal is mapped in a nite
scope. The input signal must possess limited ranges of values, thus (s(t) is usually a voltage).
Ulower s(t) Uupper
(7.23)
In AD converter the range U = Uupper Ulower is divided into q intervals and to each value of
the input signal in that particular interval is assigned to a quantization step.
s(t) sq (t)
(7.24)
Since the number of values of the quantized signal is nite, the quantized signal can be represented
with the help of a signal of alphabets. Usually binary numbers are used for it. In this case the
number of necessary bits b are derived from the number q of quantization levels
b = ld q
(7.25)
Therefore ld designates the logarithm dualis and n = x is the smallest integer number n with
x n (e.g. 4 = 3.1).
7.3.1
Quantization Characteristic Curve
The quantization characteristic curve determines the power distortions and thus the quality of
quantization in connection with s(t) the amplitude density distribution (often uniform distribution is assumed) of the input signal. Even though the Quantization is nonlinear, often
it is distinguished between linear (uniform quantizer ) and nonlinear Quantization (nonuniform
quantizer ). It indicates thereby only the characteristic of quantization (companding characteristic
diagram).
Linear quantization characteristic diagram The Quantization step s is constant
s =
U
U
= b
q
2
with q = 2b
(7.26)
The output signal can take the following values (see Fig. 7.12):

(1, 3, 5, . . . , q 1)
mid-rise quantizer
s
2
sq (t)
{s (0, 1, 2, . . . , q/2 1); s (1, 2, 3, . . . , q/2)} mid-tread quantizer
(7.27)
Nonlinear quantization characteristic diagram With a linear quantization characteristic,

the maximum relative error emax (t)/s(t) = s/(2s(t)) for smaller input signals is clearly bigger
than for larger input signals. Therefore it is sensible to quantize smaller signal values more precisely
than larger signal values (see Fig. 7.13).
Dierent nonlinear characteristics are used out of which at most A-law and -law characteristics.
Since the handling of the nonlinear quantization characteristics goes beyond the framework of this
lecture, only the linear quantization characteristic is going to be considered here.
66
sq
sq
5s
2
2s
3s
2
s
1s
2
1s
2
s
s
3s
2
2s
5s
2
3s
Figure 7.12: Linear mid-rise (left) and mid-tread (right) quantizer
sq
Figure 7.13: Quantizer with nonlinear quantization characteristics (non uniform quantizer )
7.3.2
Quantization Error Signal
The (quantization) error signal e(t) is dened as the deviation of the quantized signal sq (t) from
the original signal s(t).
e(t) = s(t) sq (t)
(7.28)
If the quantizer is adequately well controlled and if the input signal s(t) is suciently variable in
time, the error signal e(t) can be assumed as white noise with constant amplitude density in the
interval [s/2; s/2] (see Fig. 7.14). Thus for linear quantization characteristic the quantization
error power Pe can be specied as

2
Pe = E[e ] =
s
2
s
2
e2
1
1
d e = (s)2
s
12
(7.29)
67
1
s
p(e)
s
2
s
2
Figure 7.14: Assumed amplitude density function p(e) of the error signal
7.3.3
Signal-to-Noise Ratio
By signal-to-noise ratio SN R one understands the ratio of the user signal power and noise power.
Considering the non-quantized input signal s(t) with average power Ps as user signal and the
originating quantization error e(t) with average power Pe as noise signal, gives the signal-noiseratio as
Ps
SN R = 10 lg
(7.30)
Pe
The average quantization error power is approximately known under certain conditions (see Article. 7.3.2). The average input signal power however strongly depends on the dynamic range of
the AD converter and the signal shape. Thus e.g. for a noise shaped signal as input (useful) signal
with uniform amplitude density over the entire (symmetrical around zero) dynamic range U

Ps,N =
=
U
2
U
2
s2
1
1
d s = (U )2
U
12
1 b
(2 s)2
12
(7.31)
(7.32)
and for a sine signal with the amplitude U/2 (likewise full rejection)
Ps,S
2
2
1
1
U
=
sin2 x d x = (U )2
2 0
2
8
1
= (2b s)2
8
(7.33)
(7.34)
Now the signal-to-noise ratio can be given for both signals.

12(2b s)2
SN RN = 10 lg
= 20b lg 2 dB
12(s)2
= 6.02b dB
b
(7.36)
12(2 s)
= (10 lg 1.5 + 20b lg 2) dB
8(s)2
= (1.76 + 6.02b) dB
SN RS = 10 lg
(7.35)
(7.37)
(7.38)
The dynamic range or the signal form changes the SN R only over a certain oset, however each
time the SN R is increased by 6 dB independent of dynamic range and signal shape, if during
quantization 1 bit is more used.
68
7.3.4
Oversampling
Since the output signal of the AD converter is discrete in time, the entire quantization error
power Pe must reside in the frequency range [fs /2; fs /2]. Thus the power density spectrum of
the error signal shown in Fig. 7.15 results on the assumption of a white quantization noise (see
Article. 7.3.2). It is now to be demonstrated that oversampling, i.e. the sampling rate greater than
|N (f )|
images
Pe
fs
fs
f2s
fs
2
fs
Figure 7.15: Power density spectrum N (f ) of the error signal
the least necessary Nyquist rate, implies an increased signal-to-noise ratio involved with the signal
reconstruction. An (over sampling ratio) osr is introduced.
osr =
fs
fN
fN = 2fg Nyquist rate
(7.39)
An oversampling factor osr = 1 implies sampling with the Nyquist rate. If an oversampling factor
osr > 1 is selected, then quantization noise power Pe distributes itself over a larger frequency range.
A part of the quantization noise power is thereby in a frequency range outside of original signal
N (f )
Reconstructing
low pass lter
Ps
SN R
osr = 1
Pe
0
fs /2 = fg
N (f )
Reconstructing
low pass lter
Ps
SN R
osr > 1
Pe
0
fg
fs /2
Figure 7.16: Dependency of the spectral power density N (f ) of error signal on the oversampling
factor osr
frequency range (0 f fg ). For signal reconstruction, as mentioned in Article. 7.2.4, a low-pass

lter with critical frequency fg is used. This low-pass lter lters a part of the quantization noise
power Pe (fg < f < fs /2) and thereby increases the signal-to-noise ratio (see Fig. 7.16).
69
The signal-to-noise ratio SN RRek after signal reconstruction increases thereby as follows
SN RRek = SN R + 10 lg osr
(7.40)
Each doubling of sampling rate increases the signal-to-noise ratio SN RRek by 3 dB (it validates
on the assumption that the quantization error remains white during oversampling).
As far an ideal low pass lter with Nyquist rate (osr = 1) is necessary for signal reconstruction,
a lter having osr > 1 with nite edge steepness can also be used, which is technically realizable
in contrast to an ideal low-pass lter. Therefore in practice, always osr > 1 is selected.
7.3.5
Error Modelling of AD Converter
By shaping the quantization noise it is possible to reduce the quantization noise power density in the frequency range of interest. A higher signal-to-noise ratio after reconstruction can be
achieved despite of small quantization resolution of b.
N (f )
Reconstructing
low pass lter
osr 1
Ps
Pe
fg
fs
Figure 7.17: Spectral power density distribution of quantization error with the application of
a noise forming AD converter (-converter)
70
8 DIGITAL MODULATION
Digital Modulation
8.1
Introduction
By modulation, generally the suppression of a source signal on a carrier signal is understood. (see
also Section 6.1). In contrast to analog modulation, however, the source signal d{k} is a digital
signal in case of digital modulation. Since digital signals represent only time-discrete sequences,
they are not suitable for direct modulation with a time-continuous carrier, since otherwise the
modulation product would be dened also only with appropriate cycle-time points kTs (Ts
symbol duration). Therefore the modulation must be viewed always in connection with the signal
formation so that the process of the modulating signal corresponding to the modulation product is
dened between the points of cycle-time kTs . Since the digital source signal is discrete in value
with M dierent values, the modulation product always presumes one of M dierent states of the
cycle-time points kTs . The exact transition of one state to another is actually irrelevant for the
data transmission, however has substantial inuences on
the bandwidth requirement of the modulation product (send-signal),
the resistance against interferences,
the technical implementation.
8.2
Types of Modulations
If sinusoidal carriers are used, the parameters amplitude A0 , frequency f0 and phase 0 are also
available for modulation (see Section 6.2). Since in rst applications of the digital modulation,
square pulses were used as modulating signals (hard shifting /switching of the selected parameters),
the methods possesses the following titles (see Fig. 8.1)

Amplitude Shift Keying
ASK:
A0 A d{k}

Frequency Shift Keying
FSK:
f0 f d{k}

Phase Shift Keying
PSK:
0 d{k}
Three exists numerous variants on the basis of these methods, most of which hold independent
names and can be dierentiated clearly. Distinguishing criterions are e.g.
Number of states of the digital source signal
2 = 21 states (binary) e.g. BPSK binary phase shift keying
4 = 22 states (quaternary) e.g. QPSK quaternary phase shift keying
8 = 23 states e.g. 8-PSK
16 = 24 , 64 = 26 , 256 = 28 states e.g. 16-QAM, 64-QAM, 256-QAM quadrature
amplitude modulation
whether a real or a complex baseband signal is formed from the digital source signal
71
8.3 BPSK Binary Phase Shift Keying
sBP (t)
sBP (t)
A0
sBP (t)
A0
A0
A0
A0
a) ASK (special OOK)
A0
b) FSK (special 2-FSK)
c) PSK (special BPSK)
Figure 8.1: Exemplary representation of the modulation of a sinusoidal carrier signal by means
of rectangular pulse (hard shifting /switching)
real baseband signal e.g. OOK on o keying (as alteration of the ASK, see Fig. 8.1 a)),
BPSK (as alteration of the PSK, see Fig. 8.1 c))
complex baseband signal e.g. QPSK, 8-PSK, 16-QAM
Dierent variants of signal shaping
rectangular baseband signal (hard switching) e.g. OOK
half-wave sinus baseband signal (soft switching) e.g. MSK minimum shift keying
time mismatch between Inphasen and quadrature component of the complex baseband
signal e.g. O-QPSK Oset QPSK, MSK
modulation procedures with or without memory
memory-free modulation e.g. BPSK, QPSK, 16-QAM
modulation with memory e.g. CPFSK continuous phase frequency shift keying,
GMSK gaussian minimum shift keying, DQPSK dierential QPSK
In the framework of this lecture a global picture to digital modulation cannot be presented. Instead
principles and important characteristics of the digital modulation are explained here using several
examples.
To Interested readers the lecture Mobile Nachrichtensysteme I is recommended for further
details. Additional literatures are e.g. [GG98], [Cou93], [Mau85], [Kam96], [Kro91], [Pro95].
8.3
BPSK Binary Phase Shift Keying
The binary phase shift keying BPSK is the simplest PSK-version and forms the basis for understanding of the higher order PSK-versions (QPSK,8-PSK). The binary source signal d{k} is
impressed as the phase (t) on the sinusoidal carrier signal

sc = Re A0 ej(2fc t+0 )
(8.1)
72
where
d{k}
for (k 1/2) Ts t < (k + 1/2) Ts
0 (t) = 2
M

0 for d{k} = 0
=
for d{k} = 1
(8.2)
(8.3)
whereby Ts is the symbol duration and M is the number of states of digital source signal d{k},
here M = 2 (binary). At time points (k +1/2)Ts hard switching takes place between the two phase
states (see Fig. 8.2)11 .
8.3.1
The modulation product can be described as

sBP (t) = Re
$
A0 ej (t) ej 2fc t

= Re s(t) ej 2fc t
(8.4)
By comparing eqn. (5.57) and (8.4) one recognizes that

s(t) = A0 ej (t)

t kTs
j d{k}
e
rect
= A0
Ts
k

t kTs
= A0
b{k} rect
Ts
k
(8.5)
(8.6)
with b {1, 1}
(8.7)
represents the equivalent low-pass signal (baseband signal) of the modulation product (see Section 5.3 and Fig. 8.2). All considerations can focus at the baseband. One obtains the band-pass
signal sBP (t) again by multiplying the equivalent baseband signal with the carrier and taking real
part of the product (Up mixing, see Section 5.1)12 . In the frequency domain this corresponds to a
shift in the spectrum of baseband signal about fc .
11
It will be emerged later that this hard switching of a phase state is very uncomfortable in others regarding
the necessary bandwidth.One will try to arrange therefore these state swapping softly. One possibility is the phase
transition from (d{k}) to (d{k + 1}) as nonvolatile but to arrange e.g. linear. With the knowledge from the
Section 6.4 it becomes however clear that such a phase change brings also a frequency change with itself. The result
would be a special FSK-method (CPFSK continuous phase frequency shift keying). With the PSK-procedures
therefore another way is chosen in order to economize bandwidth. The amplitude of the baseband signal (complex
envelope) is softly lowered at the phase saltuses, i.e. the phase saltuses are faded out. From this point of view one
can regard the PSK procedures as special ASK-methods (QAM) . Thence ASK- and PSK-methods are linear
Modulation methods. In contrary to analog modulation methods, only the FSK methods are nonlinear with the
digital modulation methods.
12
This corresponds to a two-sideband amplitude modulation with suppressed carrier DSB-SC (see Section 6.3.6)
73
A0
sBP (t)
Ts
Ts
2Ts
3Ts
4Ts
5Ts
A0
A0
{0}
s(t) = sI (t), sQ (t) = 0
{1}
Ts
{0}
{0}
{1}
{0}
{1}
Ts
2Ts
3Ts
4Ts
5Ts
A0
Figure 8.2: BPSK-modulated carrier signal (above) and pertinent Baseband signal (equivalent
to low-pass signal, below), (the carrier frequency fc ,with no special relation to the
symbol rate, must be fc Ts R)
A digital data source delivers a time-discrete sequence d{k}, which are rstly mapped to symbols
b{k} which in this case have dierent amplitudes 13 .
d{k} = 0
d{k} = 1
b{k} = +1
b{k} = 1
(8.8)
(8.9)
The problem is that these time-discrete values (data) need to be transferred over an analog channel.
Due to this, amplitude values b{k} are weighted with delayed impulses hT (t kTs ) [Kam96]. In
order to be able to consider the impulse hT (t) in usual way as impulse response of a time-continuous
LTI system with dimension of 1/time, eqn. (8.7) is transformed into
s(t) = b(t) hT (t)
with
b(t) =
b{k} Ts (t kTs )
(8.10)
(8.11)
as output signal of an ideal DA-converter (see Section 7.2) and

A0
t
rect
hT (t) =
Ts
Ts
(8.12)
13
Importantly the allocation of the symbols of the source signal to phase or amplitude of the carrier signal is
certain. This is arbitrary,however,must agree in sender and receiver. With high order modulations e.g. QPSK,
8-PSK special allocations (Gray code) can be selected,in order to keep the bit error rate small.
74
as pulse shaper /transmission lters (for the time being square pulse shaper). This model permits
quite good and uniform mathematical description of sender and receiver, even if in practical implementations the transmission pulse shaping takes place usually digitally and the DA-conversion
is only performed thereafter.
8.3.2
Sender
A block diagram of a BPSK-modulator can be derived from what is mentioned so far (see Fig. 8.3).

sBP (t) = Re

k
b{k} Ts hT (t kTs ) ej 2fc t

Symbol
Symbol b{k}
allocation
!
k
carrier
Send impulse
pulse shaper
send lter
DAC
d{k}
(8.13)
b(t)
hT (t)
Mixer
s(t) = sI (t)
Ts (t kTs )
sBP (t)
cos 2fc t
Figure 8.3: Circuit diagram of a BPSK-Modulator
8.3.3
Phase Diagram
Representation of baseband signal in complex plane has been found to be very suitable (parametric
representation with the time t as parameter). The time dependent progression of the baseband
signal depicts the complex envelope of the bandpass signal (modulated carrier). In the phase
diagram, possible states of the baseband signal at time instances kTs are represented (see Fig. 8.4).
Therefore, (see also Chapter 5)
s(t) = sI (t) + j sQ (t)
s(kTs ) = b{k} Ts hT (0) =
(8.14)
A0 ej 0 = +A0
for d{k} = 0
A0 ej = A0
for d{k} = 1
(8.15)
For both phase states 0 and , the exponential function becomes real-valued and thus the baseband
signal s(t) in BPSK is also purely real-valued.
sQ (t) = 0
s(t) = sI (t)
(8.16)
(8.17)
75
sQ (t)
d{k} = {1}
d{k} = {0}
A0
A0
sI (t)
Figure 8.4: Phase diagram of the baseband signal for BPSK
8.3.4
In analog modulation the considerations were made in the frequency domain by using Fourier
transformation of sent signals or of the modulating signal (amplitude density spectrum). This is
however possible only if the time dependent signal itself is known or at least a typical time dependent signal in which case typical spectrum can be assumed. In digital modulation the question
that arises is which the typical symbol sequences are. Since one may hardly indicate a typical
symbol sequence as this is also code dependent, there are two possible ways.
For calculating Fourier integrals over the time t it is necessary to integrate in the interval
(, ). In practice signals are considered however only in a nite period and therefore
such considerations do not give good insight. Rather we are interested in one period of e.g.
where occurs 100 transmission symbols. (see Exercise 6 Short time spectrum). During this
period certain data sequences are assumed, in order to examine the behavior of the receiver.
For example, longer transmission of only one symbol (e.g. constant {1}) becomes a constant
component of the baseband signal and leads to loss of clock synchronization. A periodic
symbol sequence (constant {0}, {1}) thereagainst leads to a disturbing line spectrum of the
send signal. From these knowledge the goal for coding can be found avoiding such unsuitable
symbol sequences.
As particularly favorable symbol sequences with frequently changing symbols has been found,
whereby in the symbol sequences no regular patterns may appear for channel coding. One
therefore assumes usually the symbol sequence, which can be transmitted, satises an uncorrelated stochastic random process. In this case, nothing more can be worked out with the
amplitude density spectrum (Fourier transform of send signal or baseband signal), since this
assumes the knowledge of the exact time dependent progression. Instead the power density
spectrum (Fourier transform of the autocorrelation function (AKF) of sent signal or baseband signal) is used . The power density spectrum of a signal describes how the signal power
is distributed as statistical average in the frequency domain. Its knowledge is important
in order to determine the necessary transmission bandwidth or to estimate the interferences
caused in data signal outside its assigned transmission bandwidth.
Here the average spectral power density (power density spectrum) will be considered on example
of BPSK. It should be assumed that the symbol sequence d{k} is a binary, uncorrelated, random
76
process. Furthermore both symbols d{k} = {0} and d{k} = {1} are assumed equally probable
with the probability of P = 0.5 (symbol sequence d{k} = {1} is zero mean and both transmission
impulses possess the same energy).
With this assumption the power density spectrum Ss,s (f ) of the baseband signal s(t) can be
calculated as

Ss,s (f ) = F {ss,s ( )} = F
s(t) s (t + ) d t

(8.18)
AKF of s(t)
Ss,s (f ) = |b| Ts |HT (f )|

2
(8.19)
i.e. the power density spectrum is proportional to the square of the transfer function of the pulse
shaper hT (t).
For the case of a square-wave pulse shaper considered here (see Fig. 8.5)
Ss,s (f ) = |b|2 Ts |F {hT (t)} |2

2

1
t
2

rect
= A0 Ts F

Ts
Ts
= A20 Ts | si(f Ts )|2
(8.20)
(8.21)
(8.22)
The power density spectrum Ss,s (f ) of the BPSK-modulated bandpass signal sBP (t) =
s(t) cos 2fc t (up-mixed baseband signal) arises simply as a result of weighted shifting of
the power density spectrum Ss,s (f ) of baseband signal s(t) around fc .
Since so far not dealing with power density spectra, a short proof would put down here; the shifting
theory of fourier transform can also be applied for power density spectra.

Ss,s (f ) = F {ss,s ( )} = F
sBP (t) sBP (t + ) d t
(8.23)
s(t) cos(2fc t) s (t + ) cos(2fc (t + )) d t

(8.24)
=F
Since sc (t) = cos(2fc t) is a determinate signal, it yields

T
1
cos(2fc t) cos(2fc (t + ) d t)
Ss,s (f ) = F ss,s ( ) lim
T 2T T

1
ss,s ( ) cos 2fc
=F
2
(8.25)
(8.26)
and from hereon (see Fig. 8.5)

Ss,s (f ) =

1
Ss,s (f + fc ) + Ss,s (f fc )
4
(8.27)
The power density spectrum Ss,s (f ) is thereby the square of the sinc function (see eqn. (8.22)).
77
Ss,s (f )
A20 Ts
T1s
A20 Ts
2
1
Ts
Ss,s (f )
A20 Ts
4
fc
fc
Figure 8.5: Power density spectrum Ss,s (f ) of the BPSK-modulated baseband signal (above)
and Ss,s (f ) the up-mixed bandpass signal sBP (t) = s(t) cos 2fc t (down) with
square-wave pulse shaper hT (t) = 1/Ts rect(t/Ts )
In Section 4.5 the slow damping behavior of the sinc function si() was addressed among other
things. This leads to the fact that the sampled (not band-limited) BPSK has a very high bandwidth
requirement (see Fig. 8.7). Therefore in many applications a band limitation of the transmitted
signals is assumed. This band limitation can also take place equivalently in the baseband, as
narrow-band pulse-shaper lters hT (t) are used. This is also called base band signalling.
In practice root raised cosine lters are often used as pulse-shaper hT (t) (see Fig. 8.6 and 8.7).
With smaller bandwidth requirement, the adjacent channel disturbances are smaller and a given
frequency range can be occupied with a higher number of channels than to use square pulses.
hT (t) =
1 4rt/Ts cos((1 + r)t/Ts ) + sin((1 r)t/Ts )

Ts
t/Ts (1 (4rt/Ts )2 )
(8.28)
hT (t)
1r+4r/
Ts
Ts
Ts
Figure 8.6: Root-Raised-Cosine-Impulse as baseband pulse shaper hT (t)
78
Leistungsdichtespektrum
0
10
level in dB
20
30
40
50
60
70
0.5
1.5
2.5
3.5
normalized frequency f*Ts
Figure 8.7: Comparison of the power density spectra Ss,s (f ) when using square-wave impulse
(dotted line) and time limited Root-Raised-Cosine-Impulse (solid line) as pulse
shaper hT (t). (logarithmic scale of the ordinate)
8.3.5
Receiver
The receiver has the task to demodulate the received signal and afterwards make decisions with
the help of a detector, which symbols have been sent (see Fig. 8.8). During the detection process
the goal is to make as fewer wrong decisions as possible. Depending on the expected transfer ratios
(channel model) and type of modulation, there exist many receiver concepts which with certain
complexity guarantee fewer error rates. A receiver that achieves the theoretically lowest error
rate is the so called optimum receiver. Such receivers can often exceed the limit of technical
realizability for certain channel models which requires the use of compromise solutions.
rBP (t)
Demodulator
Detector
d{k}
Figure 8.8: Simplied representation of a receiver for digital data communication
Matched-Filter Receiver In the context of this lecture only the match lter receiver will
be briey considered. For the AWGN-channel (see Section 8.3.6) this receiver 14 represents optimum receiver. The down-converted received signal is applied in parallel to several matched lters,
14
An equivalent realization represents the correlation receiver.
79
whereby each of these lters is matched on exactly one transmission symbol pulse (therefore the
name matched lter ). The lters correlate the received signal with the corresponding pulses. The
decision is made in favor of the symbol, whose matched lter shows the highest accordance (highest
output amplitude).
In the case of BPSK both the transmission pulses for the symbols dier d{k} {0, 1} only by
their sign, so that only one matched lter with
hR (t) = hT (t)
(8.29)
is needed15 . An analog-digital converter follows with the symbol rate fs = 1/Ts after ltering.
In the case of the BPSK detection symbol manipulation (bit by bit) is realized by a memoryless
threshold switch, i.e. upon the sign of the sampling value it is decided whether a {0} or {1} was
sent. As however, it comes to wrong decisions, the following channel decoding must try to seek
out and correct these erroneous bits again and again. Fig. 8.9 shows the circuit diagram of such
a matched lter receiver (with accepted ideal carrier and symbol synchronization).
Mixer
Filter
AD-converter
Decision maker
hR (t)
ADC
cos 2fc t
matched
lter
rBP (t)
fs =
1
Ts
d{k}
Detector
Figure 8.9: Principle diagram of a BPSK matched lter receiver
8.3.6
Data Transmission over Noisy Channel
The sent signal must bridge the spatial distance to the receiver. Dierent transmitting media can
be used thereby, e.g..
metal conductors
Two wire line
Coaxial conductor
Hollow conductor
Optical ber (LWL)
Mono mode ber
Multi-mode ber
free space
Radio link
15
Such a lter maximizes the signal signal-to-noise ratio SN R with given receiving conditions (Eb /N0 at the
receiver input, see Section 8.3.7) to the sampling time kTs at the decision maker/detector input. Under the condition
of an AWGN-channel this SN R maximization is equivalent to the minimization of bit error rate BER.
80
Mobile network channels

Satellite route
Depending upon medium and length of the transmission path the sent signal is more or less strongly
distorted, attenuated and overlayed with disturbing signals. In order to achieve the higher data
rate, the modest error rate and multiple use of a medium, the type of modulation, coding, carrier
frequency etc. must be adapted on the transmission path. Moreover it is inevitable to describe the
transmission path eectively. In communications the term channel is used for the transmission
path as an abstract model. The channel model provides the description of the most important
characteristics of the transmission path.
AWGN Channel Here only very simple channel model is presented, the AWGN-channel (additive white gaussian noise). It is characterized by its simplicity and easy mathematical treatment.
The received signal r(t) is understood as superposition of the undisturbed sent signals s(t) with
a White Gaussian noise n(t) (see Fig. 8.10).
r(t) = s(t) + n(t)
(8.30)
It is not important however whether the signals r(t) and s(t) are bandpass or baseband signals. In
case of assuming complex baseband signals the real and imaginary part of the signals are uncorrelated with those of the complex noise signal; and thus the in-phase and quadrature component
of the baseband signals can be considered separately.
If we are interested only in the performance in the presence of additive white noise we can focus
at the baseband (see Section 5.3), i.e. the modulation and demodulation with the carrier signal
(see Section 5.1) can be ignored. The data transmission system is indicated as a block diagram in
Fig. 8.10.
Canal
d{k} b{k} b(t)
hT (t)
s(t)
Send lter
pulse shaper
r(t)
n(t)
hR (t)
rE (t)
ADC
rE (kTs )
Detector
d{k}
Receiver lter
matched lter
Figure 8.10: Blocks diagram of a data transmission system in Baseband
In case of the BPSK, the baseband signal s(t) = si (t) = s(t) is real valued. Assuming ideal
synchronization we are interested in real part of the received signal r(t). From that the whole
system can be analyzed only in real domain (see eqn. (8.30)).
The superimposed noise signal n(t) is assumed to be as a zero mean, stationary, white Gaussian
stochastic process. All possible sources of noise ni (t) arising in the system are combined into one
noise source n(t) with noise power density of
N0
for < f <
(8.31)
2
The amplitude density distribution follows a normal distribution with = 0 (see Section 4.6)
Sn,n (f ) =
fn (n) =
(n)2
1
e 22
2
(8.32)
81
8.3.7
Calculation of Bit Error Ratio
By symbol or bit error rate one understands the probability with which the detector decides a
wrong symbol or bit. In the case of BPSK, each transmitted symbol consists of only 1 bit, so
that symbol and bit error probability are identical. These error probabilities are random variables
which specify the ratio of the number of wrong decisions to the number of all met decisions on
average. The computation of the bit error rate BER will be presented here as an example of
BPSK assuming an AWGN-channel.
Also assuming ideal synchronization, the sampled values
rE (kTs ) = sE (kTs ) + nE (kTs )
(8.33)
gained at the times t = kTs (see Fig. 8.10) are present at the entrance of the detector. The
sampling values sE (kTs ) result from the receiver ltering and sampling of the baseband signal s(t)
to the time-discrete convolution sum

sE (kTs ) = s(t) hR (t)
(8.34)
= Ts
t=kTs
b{n} h((k n)Ts )
(8.35)
n=
the sender and receiver lter in this case are combined into a single lter 16 h(t) = hT (t)
hR (t). Inter-symbol interference freeness ISI is assumed here which is not to be dealt further
(see Exercise 23). The Rectangular and Root-Raised-Cosine pulse shaper impulse response h(t)
introduced in the script fullls this ISI-freeness with the matched lter receiver. Sending of the
symbol d{k} = {0} is marked by d0 and the associated received value by s0 , accordingly d1 and
s1 for the sent symbol d{k} = {1}. It then holds
sE (kTs ) = Ts h(0) b{k}

s0
for the event d0
=
s1 = s0 for the event d1
(8.36)
(8.37)
The noise signal n(t) is band-limited by the received lter hR (t) and thereto possesses a nite
signal power, which corresponds to the variance of the noise process.

N0
2
=
|HR |2 d f
(8.38)
2
The amplitude probability density function remains unaected after ltering.
If the noise signal is n(kTs ) = 0 ( 2 0), the received signal rE (kTs ) adopts at the detector
entrance only the values s0 with a sent {0} or s1 with a sent {1}. If the noise signal power/variance
is larger than 0, the received value rE can deviate more or less from these two values. If one applies
a suciently large number of received values rE for the two events d0 and d1 (send a {0} or {1})
16
For a matched lter receiver with hR (t) = hT (t) the total impulse response h(t) is identical with the (impulse)
autocorrelation function of the sending lter hT (t).
82
in a histogram, thus results approximately the amplitude density distributions at the detector
entrance. (see Fig. 8.11)
(rE s0 )2
1
e 22
(8.39)
frE |d0 (rE |d0 ) =
2
(rE s1 )2
1
frE |d1 (rE |d1 ) =
e 22
(8.40)
2
small noise power
frE |d0 (rE |d0 )
frE |d1 (rE |d1 )
middle noise power
large noise power
s1
s0
rE
Figure 8.11: Amplitude density distribution frE (rE ) of the sampled values rE (kTs ) at the entrance of the detector for BPSK
The detector nds a decision rE (kTs ), which can take the values d0 or d1 with knowledge of the
received value. Four dierent decisions are possible (see Fig. 8.12).
P (d0 )
d0
P (d{k}
= d0 | d{k} = d0 )
d0
P (d{k}
= d0 | d{k} = d1 )
sent symbol
d{k}
d{k}
P (d{k}
= d1 | d{k} = d0 )
P (d1 )
d1
P (d{k}
= d1 | d{k} = d1 )
decided symbol
at the receiver
d1
Figure 8.12: Model of distorted channels
Two cases of decision are wrong thereby:
the detector decides a d{k}

= d0 , although d{k} = d1 was sent
the detector decides a d{k}

= d1 , although d{k} = d0 was sent
The probability of a wrong decision is
= d1 | d{k} = d0 ) P (d0 )
BER = P (d{k}
= d0 | d{k} = d1 ) P (d1 ) + P (d{k}
(8.41)
83
where P (d0 ) and P (d1 ) are the a-priori-probabilities for the occurrence of the events d0 and d1 ,
i.e. the probabilities for sending a {0} or {1}. The channel coding guarantees
P (d0 ) = P (d1 ) =
1
2
(8.42)
The conditioned probabilities for the wrong decisions result from the region under the appropriate
amplitude density distribution within the range, where the detector decides wrongly. In the case
of BPSK, the detector makes its decision on basis of the sign of rE (kTs ), i.e.,

P (d{k} = d0 | d{k} = d1 ) =
frE |d1 (rE |d1 ) d rE
(8.43)
0
0
P (d{k}
= d1 | d{k} = d0 ) =
frE |d0 (rE |d0 ) d rE
(8.44)
With eqn. (8.39, 8.40), the relation s0 = s1 > 0 and imposing the symmetry of the amplitude
density distributions the error probability BER results to

(r s )2
(rE s1 )2
1 0
1 1
1
E 20
2
e
e 22 d rE
BER =
d rE +
(8.45)
2 2
2 0
2
0

(r s )2
(rE +s0 )2
1
1
E 20
2
=
e
e 22 d rE
d rE =
(8.46)
2
2
0
0
These integrals can be alternatively reduced with help of the substitution x = rE s0 or x = rE+s
2
to the distribution function () of standardized normal distribution or to the complementary
error function erfc().
s0
x2
1
BER =
e 2 d x =
2
s
0
=
=
1 2
s
0
2
1
erfc
2
ex d x
s
0
2
(8.47)

(8.48)
The bit error ratio thus exclusively depends on the ratio s0 / at the detector entrance. This ratio
can also be indicated as signal-to-noise ratio. With s20 = s21 as power of the sampled data signal
and 2 as noise power, yields the SN R at detector entrance to
SN R =
s20
2
(8.49)
The bit error ratio can also be specied as

1
BER = erfc
2
"%
SN R
2
#
(8.50)
Since the signal-to-noise ratio at the detector input depends on the receipt lter hR (t), the bit
error ratio is rather interesting as a function of the signal-to-noise ratio at the receiver input.
As suitable signal-to-noise measure proves the ratio Eb /N0 , whereas Eb is the average energy of a
single character and N0 /2 is the noise performance density at receiver input (see eqn. 8.31). With
84
matched lter receiver for a given Eb /N0 at the receiver input the signal-to-noise ratio SN R is
maximized at the detector entrance and it applies

|b|2 Ts2 h2T (t) d t
Eb
SN R =
=
(8.51)
N0 /2
N0 /2
So nally, for BPSK-modulation
1
BER = erfc
2
"%
Eb
N0
#
(8.52)
Fig. 8.13 shows the dependence of bit error ratio with the ratio Eb /N0 for BPSK under the
condition of an AWGN-channel and a matched lter receiver
BER
100
101
102
103
104
15
10
10
Figure 8.13: Dependence of bit error ratio with Eb /N0 for BPSK
10 lg(Eb /N0 ) dB
85
8.4 QPSK Quaternary Phase Shift Keying
8.4
QPSK Quaternary Phase Shift Keying
Same like BPSK unless the baseband signal is complex here.
8.5
Further ASK/PSK Methods
8-PSK, 16-QAM
perhaps also multi-valued QAM with real baseband signal
dealt briey here only with phase diagram
BER
100
8-PSK
101
16-QAM
102
16-PSK
BPSK,
QPSK
103
104
105
8-QAM
10
12
14
16
18 Eb /N0 in dB
Figure 8.14: Bit error ratio for dierent type of PSK or QAM modulations with respect to
signal-to-noise ratio Eb /N0
86
BPSK
QPSK
sQ
8-PSK
sQ
dmin
dmin
sI
sI
dmin
sI
sI
16-QAM
sQ
dmin
16-PSK
sQ
8-QAM
sQ
dmin
sI
sQ d
min
sI
Figure 8.15: Phase diagram (due to closely related symbols, always arises in the case of same
noise power higher symbol or bit error rates )
8.6 Frequency Shift Keying
8.6
87
Frequency Shift Keying
auch nur kurz das Prinzip zeigen, vielleicht noch eine Sender- und Empfangerrealisierung vorstellen
88
REFERENCES
References
[BHPT95] O. Beyer, H. Hackel, V. Pieper, and J. Tiedge. Wahrscheinlichkeitsrechnung und mathematische Statistik. Mathematik f
ur Ingenieure und Naturwissenschaftler. B.G. Teubner
Verlagsgesellschaft, Leipzig, 7. edition, 1995. ISBN: 3-8154-2075-X.
[BS79]
I. N. Bronstein and K. A. Semendjajew. Taschenbuch der Mathematik, volume 1.

B.G. Teubner Verlagsgesellschaft, Leipzig, 24. edition, 1979. Lizenzausgabe f
ur den
Verlag Harri Deutsch, Thun 1989.
[BS96]
I. N. Bronstein and K. A. Semendjajew. Teubner-Taschenbuch der Mathematik, volume 1. B.G. Teubner Verlagsgesellschaft, Leipzig, 1996. ISBN: 3-8154-2001-6.
[Cou93]
Leon W. Couch II. Digital and Analog Communication Systems. Prentice-Hall International, Inc., 4th edition, 1993. ISBN: 0-13-571845-7.
[Fet96]
Alfred Fettweis. Elemente nachrichtentechnischer Systeme. B.G. Teubner, Stuttgart,

2. edition, 1996. ISBN: 3-519-16131-1.
[Fli91]
Norbert Fliege. Systemtheorie. Informationstechnik. B.G. Teubner, Stuttgart, 1991.

ISBN: 3-519-06140-6.
[GG98]
Ian Glover and Peter Grant. Digital Communications. Prentice Hall, Europe, 1998.
ISBN: 0-13-565391-6.
[GZZZ95] G. Grosche, V. Ziegler, D. Ziegler, and E. Zeidler, editors. Teubner-Taschenbuch der

Mathematik, volume 2. B.G. Teubner Verlagsgesellschaft, Leipzig, 7. edition, 1995.
ISBN: 3-8154-2100-4.
[Hof98]
R
udiger Homann. Signalanalyse und -erkennung. Springer-Verlag, Berlin, Heidelberg,
1998.
[Kam96]
Karl Dirk Kammeyer. Nachrichten

ubertragung. Informationstechnik. B.G. Teubner,
Stuttgart, 2. edition, 1996. ISBN: 3-519-16142-7.
[Kro91]
Kristian Kroschel. Daten

ubertragung.
ISBN: 3-540-53746-5.
[L
uk95]
Hans Dieter L
uke. Signal
ubertragung Grundlagen der digitalen und analogen
Nachrichten
ubertragungssyteme. Springer Verlag Berlin, 6. edition, 1995. ISBN: 3540-58753-5.
[Mau85]
Rudolf Mausl. Digitale Modulationsverfahren. Dr. Alfred H

uthig Verlag, Heidelberg,
1985.
[Mau92]
Rudolf Mausl. Analoge Modulationsverfahren. Dr. Alfred H

uthig Verlag, Heidelberg,
2. edition, 1992. ISBN: 3-7785-2130-6.
[Pro95]
John G. Proakis. Digital Communications. McGraw-Hill International Editions, 3rd

edition, 1995. ISBN: 0-07-113814-5.
Springer-Verlag, Berlin, Heidelberg, 1991.
REFERENCES
89
[SS88]
Norbert Sieber and Hans-J

urgen Sebastian. Spezielle Funktionen. Number 12 in Math
ematik f
ur Ingenieure, Naturwissenschaftler, Okonomen
und Landwirte. B.G. Teubner
Verlagsgesellschaft, Leipzig, 3. edition, 1988.
[WS93]
Gerhard Wunsch and Helmut Schreiber. Analoge Systeme. Springer-Verlag, Berlin,

Heidelberg, 1993.
90
A
A.1
FORMULAS
Formulas
Denitions
Sinc function one of the most frequently used function in communications and is dened as
sin x
x
si(x) = sinc(x)
sinc(x) =
si(x) =
sin x
x
(A.1)
(A.2)
where si() is user in German language and sinc() is the American counterpart.
The rect function is also frequently used in the script and dened as follows
1
rect(x) =
for
1
2
<x<
for |x| =
1
2
1
2
(A.3)
otherwise
The triangular function is used rather rarely and dened as follows here

triang(x) =
1 |x| for
0
A.2
1x1
(A.4)
otherwise
Fourier Transformation

S(f ) = F {s(t)} =
S() =
s(t) = F 1 {S(f )} =
1
=
2
s(t) e j 2f t d t
Fourier transformation
(A.5)
Inverse Fourier transformation
(A.6)
s(t) e j t d t
S(f ) ej 2f t d f
S() ej t d
The sucient (however not necessarily)

criteria for the convergence of Fourier integral is from
Dirichlet-Jordan (absolute Integrable |s(t)| d t < , s(t) is decomposable in many nite
constant, monotonous subintervals (Due to the duality of FT this is also valid for S(f ))) [BS79,
BS96].
Some properties of Fourier Transformation:
91
A.2 Fourier Transformation
For real signal s(t):

|S(f )| = |S(f )|
arg(S(f )) = arg(S(f ))
Re {S(f )} = Re {S(f )}
Im {S(f )} = Im {S(f )}
Amplitude spectrum is an even function

(A.7)
Phase response is an odd function
(A.8)
Real part of the spectrum is an even function (A.9)
Imaginary part of the spectrum is an odd function
(A.10)
Decomposition of real time signal in even and odd part:

s(t) = sg (t) + su (t)
(A.11)
mit
s(t)
sg (t)
su (t)
Parsevals Equation:
S(f )
Re{S(f )}
j Im{S(f )}

|s(t)| d t =
2
|S(f )|2 d f
(A.12)
(A.13)
(A.14)
(A.15)

If the function s(t) is absolutely integrable in the interval (, ) , i.e. |s(t)| d t < ,
then the function S(f ) = F {s(t)} in < f < is static and tends to zero for f =
[BS79]. Due to symmetry of Fourier transformation (see also Tab. A.1) the opposite is also
valid.
Band limited signals have an innite time expansion and time limited signals are band
unlimited (see also lecture Signalverarbeitung [Hof98] page 109f).
Important properties of dirac delta function:
Sifting property

s(t) (t t0 ) d t = s(t0 )
S(f ) (f f0 ) d f = S(f0 )
(A.16)
(A.17)
Adjournment property
s(t) (t t0 ) = s(t t0 )
S(f ) (f f0 ) = S(f f0 )
(A.18)
(A.19)
92
Linearity
ai si (t)
FORMULAS
ai Si (f )
Complex conjugate of time function
s (t)
S (f )
Complex conj. of frequency function
s (t)
S (f )
Mirroring
s(t)
S(f )
Symmetry
S(t)
s(f )
Time shifting
s(t t0 )
S(f ) e j 2f t0
Frequency shifting
s(t) ej 2f0 t
S(f f0 )
s(t) cos(2f0 t)
1
(S(f
2
+ f0 ) + S(f f0 ))
s(t) sin(2f0 t)
j
(S(f
2
+ f0 ) S(f f0 ))
Scaling
s(at)
1
S( fa )
|a|
Convolution in time domain
s1 (t) s2 (t)
S1 (f ) S2 (f )
Convolution in frequency domain
s1 (t) s2 (t)
S1 (f ) S2 (f )
Dierentiation
dn
s(t)
d tn
(j 2f )n S(f )
t
Integration
Limiting value
1
S(f )
j 2f
s( ) d
lim sn (t) = s(t)
+ 12 S(0) (f )
lim Sn (f ) = S(f )
Table A.1: Properties of Fourier Transformation
(t)
(t t0 )
e j 2f t0
(f )
ej 2f0 t
(f f0 )
cos(2f0 t)
1
((f
2
sin(2f0 t)
+ f0 ) (f f0 ))

!
f Tn
X 1 (f ) =
T XT (t) = T
+ f0 ) + (f f0 ))
j
((f
2
(t nT )
n=
n=
Table A.2: Fourier transformation of generalized functions (with non-dimensional variables t

and f )
93
A.3 Notes on Convolution
rect(t)
sinc(f ) = si(f )
sinc(t) = si(t)
rect(f )
triang(t)
sinc2 (f ) = si2 (f )
j
t
sgn(f )
sgn(t)
1
j t
Table A.3: Some correspondence of Fourier transformation Fourier transformation of generalized functions (with non-dimensional variables t and f )
A.3
Notes on Convolution
The connection between the two general complex functions f, g

f ( )g(t ) d
(f g)(t) :=
(A.20)
is known as convolution. In (engineering) literature, as in this script, the notation is usually used as
f (t) g(t) instead of (f g)(t). This way of writing, however established, is not completely correct.
The problem outcrops when notations occur like f (t) g(t), f (at) g(t) or f (t) g(t t0 ).
Therefore it is always implicitly presupposed that the argument for the convolution is always t (in
time domain) or f (in frequency domain).
f g =gf
Commutativity
f (g h) = (f g) h
Associativity
f (g + h) = f g + f h
Distributivity
From f g = 0 follows f = 0 or g = 0.
Table A.4: Properties of convolution
94
APPENDIX
Appendix
under construction!
B.1
Analytical Signal
Since the transmitted signal sBP (t) has to be real, its Amplitude spectrum is an even function
while the phase spectrum is an odd function of frequencies (see Section. 3.3). From that (see
eqn. 3.47)
(f )
(B.1)
SBP (f ) = SBP
Therefore to describe the signal one needs knowledge of the spectrum for frequancies f 0. The
other half can be omitted.
The dependence between the spectrum S(f ) of the equivalent low pass signal s(t) (bandlimited
with fg < fc ) and spectrum from SBP (f ) of the upconverted signal sBP (t) gives

F {sBP (t)} = F Re s(t) ej 2fc t
(B.2)
Mit Re {z} = 12 (z + z ) (see Excercise 1) gives

1
1
j 2fc t
j 2fc t
F {sBP (t)} = F
(B.3)
+ s (t) e
s(t) e
2
2

1
(B.4)
= S(f fc ) + S ((f + fc ))
2
As one can see, the spectrum SBP (f ) is obtained through left- and right-shift of the spectrum S(f ) or in other words S (f ) of the equivalent low-pass signal s(t).
If one compares this with (see also Fig. B.1)

F s+ (t) = F s(t) ej 2fc t
(B.5)
(B.6)
= S(f fc )
it can be seen that exactly the left shifted part of the spectrum is missing. Signal s+ (t) is
called analytical signal. Also holds

S + (f ) = F s+ (t) = 0 f
ur f < 0
(B.7)
Signals sBP (t) and s+ (t) are clearly dependent as follows.

sBP (t) = Re s+ (t)
s+ (t) = sBP (t) + j H {sBP (t)}
(B.8)
(B.9)
Where H {sBP (t)} corresponds to the Hilbert transform of sBP (t).

To sum up the following holds
s(t)
+
j 2fc t
s (t) = s(t) e

sBP (t) = Re s+ (t) = Re s(t) ej 2fc t
complex baseband signal
(B.10)
analytical signal
(B.11)
real bandpass signal
(B.12)
Use of the complex (low pass equivalent) representation of the signals makes the system analysis
much easier and because of this reason one often uses this way to analyse systems . Real bandpass
signals sBP (t) can that be easily obtained.
95
B.1 Analytical Signal
|S(f )| = |F {s(t)} |
A
fg
fg
|S (f )| = |S(f fc )|
f
fc

|SBP (f )| = 12 S(f fc ) + S ((f + fc ))
A
fc
fc
Figure B.1: Spectra of complex baseband signal, analytical signal and real bandpass signal
f g < fc
96
C
C.1
C EXCERCISES
Excercises
Problems
The exercises are created in order to help the students check their knowledge and for preparing
examination. In your own interest you should use the solutions which are also provided as a
checklist for the solutions you have obtained and not to examine the solutions before even
reading the problems through. Most of the problems are solved as a part of the exercises on board
in parallel to the lectures.
Problems denoted with
go beyond this lecture or are more dicult to solve.
Repetition
1. Each complex number z C has the following equivalent notations
with x, y R
x = Re{z}
y = Im{z}
z = r cos + j r sin with r R 0, [0, 2)

r = |z| = x2 + y 2
arctan xy
2
= arg z = arctan xy +
arctan xy + 2
z = x + jy
z = re
(C.1)
(C.2)
(C.3)
(C.4)
(C.5)
for x > 0 y 0
for x = 0 y > 0
for x < 0
(C.6)
for x = 0 y < 0
for x > 0 y < 0
(C.7)
The following expression shows the relation between trigonometric and exponential functions
ej = cos + j sin
(C.8)
Eqn. (C.8) is referred to as Eulers Formula.

(a) Show with respect to eqn. (C.8) that the following addition theorems hold
cos(x y) = cos x cos y sin x sin y
x+y
xy
cos
sin x sin y = 2 sin
2
2
(C.9)
(C.10)
(b) Show that the following relations hold

z + z = 2Re{z}
z z = j 2Im{z}
z z = |z|2
where z stands for the conjugate complex of z.
(C.11)
(C.12)
(C.13)
C.1
97
Problems
2. A periodic function of time s(t), with a period T > 0 can be expanded into Furrier series
FR{s(t)} as
k
k
a0
ak cos 2 t + bk sin 2 t
+
s(t) = FR{s(t)} =
2
T
T
k=1
(C.14)
with the following Fourier coecients

2
ak =
T
2
bk =
T

T
2
T2
T
2
T2
k
s(t) cos 2 t d t
T
k = 0, 1, 2, . . .
(C.15)
k
s(t) sin 2 t d t
T
k = 1, 2, . . .
(C.16)
under the assumption that Fourier series (C.14) converges. The theory of Dirichlet [BS96]
purveys a convergence criterion sucient for many practical purposes. According to it, for
a monotone function s(t) with period T which which is piecewise continuous applies the
following:

lim
a0
k
k
+
ak cos(2 t) + bk sin(2 t)
2
T
T
k=1
n
s(t)
where s(t) is continuous
s(t0)+s(t+0)
2
otherwise
(C.17)
whereas s(ti 0), s(ti + 0) represent correspondingly the left and right limit of s(t) at a
point of discontinuity ti and both limits must exist. According to equation (C.14) the periodic
time function s(t) can be expanded into spectral allotments with the help of a Fourier series
expansion into an innite sum of weighted sine and cosine components of dierent frequencies
fk = Tk . This type of process is called Harmonic analysis or Spectral analysis. This has
great importance in the communications. However one often uses the (equivalent) complex
form of the Fourier series:
s(t) = FR {s(t)} =
cn ej 2 T t
(C.18)
n=
with the corresponding Fourier coecients

cn = |cn | e
j n
1
=
T
T
2
T2
s(t) e j 2 T t d t
(C.19)
Therefore the following relation applies between the coecients cn and ak , bk :

a0
2
ak j b k
ck =
2
ak + j b k
ck =
2
c0 =
k=0
(C.20)
k = 1, 2, . . .
(C.21)
k = 1, 2, . . .
(C.22)
98
C EXCERCISES
One takes |cn | and n = arg{cn } over the discrete frequency fn = Tn to get the corresponding amplitude and phase spectrum of the periodic time function s(t). This yields the so
called Line spectra, whereby the distance of two neighboring spectral lines comprises the
fundamental frequency f1 = T1 of s(t).
Further we consider the possibilities to obtain spectral representation for non-periodic functions. The Fourier series expansion, which can only be used for periodic functions, is not
suitable here as the signal is non periodic. However it provides a useful approach, if one
assumes that the non-periodic (absolutely integrable function in the interval (, +))17
is periodically extended to a function s(t) which is periodic with an innitesimal period
duration T [Hof98]. To carry out formally this limitation, putting eqn (C.19) in (C.18) gives:
#
" T

2
n
n
1
s(t) =
s(t) e j 2 T t d t ej 2 T t
(C.23)
T
T
n=
2
With the insertion of f = T1 as distance of two neighboring spectral lines and the frequency
variables f = Tn = n f results
#
" T

2
s(t) =
s(t) e j 2f t d t ej 2f t f
(C.24)
n=
T2
For T , f would be innitesimal small,

! i.e. neighboring spectral lines move innitely
towards each other. By summing
n= . . . f in eqn (C.24) with the integral
close
. . . d f yields
#
"
s(t) =
s(t) e j 2f t d t ej 2f t d f
(C.25)

S(f )
where the function S(f ) is now a continuous succession of frequency f as spectrum or Fourier
transformation of the time function s(t). Non-periodic time functions therefore possess a
continuous spectrum.
The transition of time to frequency domain is named as Fourier transformation and the
reverse process is inverse Fourier transformation. We may write as follows

S(f ) = F {s(t)} =
s(t) e j 2f t d t
(C.26)
and
s(t) = F
{S(f )} =
S(f ) ej 2f t d f
(C.27)
Also in the case of non-periodic time function s(t) the amplitude and phase spectra can be
given as |S(f )| and (f ) = arg{S(f )}.
(a) Show considering the relation (C.20)-(C.22), that the Fourier series in (C.14) can be
transformed toa form as given by eqn (C.18). Use the Eulers forms (C.8)
17
This property of s(t) is assumed additionally for all the following considerations.
C.1
99
Problems
ej + e j
2
ej e j
sin() =
2j
cos() =
(C.28)
(C.29)
(b) Calculate the Fourier coecients cn of the complex Fourier series for the real periodic
rectangular signal s1 (t) as presented in Fig. C.1. Sketch the corresponding amplitude
spectrum for = T4 in frequency domain from 12
to 12
.
T
T
s1 (t)
s0
2 0
Figure C.1: periodic rectangular signal
(c) Calculate the spectrum S2 (f ) of the (real) rect impulse s2 (t) in Fig. C.2. Sketch the
corresponding amplitude spectrum for = T4 in same frequency domain as in task (2b).
s2 (t)
s0
2 0
Figure C.2: Rect impulse
Basics
3. Figure C.3 shows an exemplary transmission system with given power-, gain-, and
attenuation-levels. The path loss is given as LS = 20 log(100d) dB where d stands for the
distance between transmitter and receiver antenna (in meter).
PN = 50 dBm
g1 = 23 dB
g2 = 60 dB
PT = 12 dBm
Modulator

Tx amplier
LS
SN R 7 dB
Demodulator
multistage Rx amplier
Figure C.3: transmission system
100
C EXCERCISES
(a) What is the maximum distance dmax that ensures a minimum signal to noise ratio of
SN R = 7 dB at the input of the demodulator?
(b) Which power level PT (in dBm) is necessary in order to reach a signal to noise ratio of
SN R = 20 dB at a distance of d = 50 m?
Signal Theory and LTI Systems
4. Let {n (t)} be a series of functions where n N, n > 0. An example graph is shown in
g. C.4.
1
1
0 for n < x < n

n (x) = n4 for |x| = n1
(C.30)
n
for n1 < x < n1
2
n
2
n (x)
n1
1
n
Figure C.4: The Dirac impulse (x) as the limit of certain known signals
(a) Show that the Dirac impulse (x) can be interpreted as the limit of the series {n (x)}
for n .
(C.31)
(x) = lim n (x)
n
(b) Show that the following relation holds (which is called the sifting property of the Dirac
impulse)

s(x)(x x0 ) d x = s(x0 )
(C.32)
Use the result of subtask (4b). The signal s(x) is assumed to be continuous.
5. Fig. C.5 represents the spectrum S(f ) (arg(S(f )) = 0). Calculate the corresponding time
signal s(t).
S(f )
a
2
A
2
fc fg fc fc + fg
fc fg
Figure C.5: Spectrum S(f )
fc
fc + fg
C.1
101
Problems
6. The following short-time spectrum is given

T
ST (f ) =
s(t) e j 2f t d t
(C.33)
(a) Deduce a connection between the short-time spectrum ST (f ) and the spectrum S(f )
as a function of time duration T .
(b) Calculate the short-time spectrum ST (f ) for the signal s(t) = cos(2f0 t) and sketch
|ST (f )| with T = 1/f0 , T = 10/f0 and T = 100/f0 . Compare the spectrums.
7. Prove the Parsevals Theorem

|x(t)| d t =
|X(f )|2 d f
(C.34)
8. (a) Calculate the spectrum of the Gauss impulse

g(t) =
(b) Calculate the area A =
1
2 2
t2
e 22
(C.35)
g(t) d t from g(t).
9. The Hilbert transformation transforms a time function again on a time function. A very
important class of signals, so called analytic signals is dened as
xA (t) = x(t) + j xH (t)
(C.36)
The Hilbert transformation is dened as

xH (t) = H {x(t)} = x(t)
1
t
(C.37)
The convolution in time domain corresponds multiplication in frequency domain (see

Fig. C.6).

1
1
(C.38)
F {x(t)} F
xH (t) = F
t
= F 1 {X(f ) ( j) sgn(f )}
(C.39)
(a) Given is a Signal x(t) = A0 cos(2f0 t). Calculate the Hilbert transformed xH (t) =
H {x(t)}.
(b) Sketch the spectrum XA (f ) = F {xA (t)} = F {x(t) + j xH (t)}. How do the spectrums
X(f ) and XA (f ) dier?
(c) Calculate the amplitude |xA (t)| = |x(t) + j xH (t)|. Which feature can you conclude into
from here?
102
C EXCERCISES
Figure C.6: Relation between Hilbert and Fourier transformation
10. Any real-valued signal x(t) can be divided into a sum of an even (symmetrical) and an odd
(skew-symmetrical) signal
x(t) = xg (t) + xu (t)
(C.40)
where
1
xg (t) = (x(t) + x(t))
2
1
xu (t) = (x(t) x(t))
2
(C.41)
(C.42)
(a) Show that, the following relation applies for real-valued signal x(t) = xg (t) + xu (t)
(compare eqn. (A.11) (A.14)).
X(f ) = Xg (f ) + Xu (f )
(C.43)
mit
Xg (f ) = Re{X(f )}
Xu (f ) = j Im{X(f )}
(C.44)
(C.45)
Besides, if the signal x(t) is causal, i.e.

x(t) = 0 for t 0
(C.46)
so applies

xg (t) =
1
x(t)
2
1
x(t)
2
for t 0
(C.47)
for t < 0
xu (t) = xg (t) sgn(t)
(C.48)
The real and imaginary part of the Fourier transformed causal signal are not independent
of each other, however they are related to each other by the Hilbert transformation.
(b) Show that for real-valued causal signal x(t) = xg (t) + xu (t) the following relation holds
Xg (f ) = ( j) H {Xu (f )}
(C.49)
C.1
103
Problems
11. A system is called strictly causal, if for each time instance t0 , the output value y(t0 ) depends
only on previous input values x(t), t < t0 , and not on current or future input values. Show
that for LTI-systems this denition can be expressed equivalently, in the following way:
An LTI-system with its impulse response h(t) is called strictly causal if for all t 0, we have
h(t) = 0; otherwise, the system is noncausal.
h(t) = 0 for t 0
12.
(C.50)
(a) Calculate the convolution product

triang(x) = rect(x) rect(x)
(C.51)
and
y(x) = rect(x) rect(x) rect(x) = triang(x) rect(x)
(C.52)
and sketch the behavior of the corresponding signal.

(b) Calculate and sketch the spectrum of the triangular function triang(t/T ).
13. Compute on the basis of Fig. C.10 on page 105 the impulse response of this low-pass lter
from the depicted spectra H(f ) = |H(f )|.
Hint: You can represent H(f ) as convolution product of two rectangular functions.
14.
(a) An input signal x(t) with x(t) = 0 for 0 > t > T1 and the causal impulse response h(t)
with h(t) = 0 for 0 > t > T2 are given, i.e. both signals are time limited (Signals with
nite duration). Show that the output signal y(t) = x(t) h(t) is also time limited, i.e.
y(t) = 0 for 0 > t > T3 , where T1 , T2 , T3 are positive real numbers. Calculate T3 in
reliance of T1 and T2 .
(b) Given is the non-causal LTI-System with the impulse response h(t) (see also Fig. C.7)

t
t
h(t) = rect
+ 1.5 +
6
(C.53)
T
T
In which interval the non-causal input signal x(t) must be known in order to compute
the initial value y(0)?
h(t)
1
-2
-1
Figure C.7: Impulse response h(t)
15.
t
T
(a) Figure C.8 shows the impulse response h(t) of a certain LTI-system. What is the
corresponding mathematical operation performed by this system? What is the name of
such a system?
104
C EXCERCISES
h(t)
A
Ts
TS
Figure C.8: Impulse response h(t)
(b) This system is excited by a rectangular pulse x1 (t) with the duration Ti .
A
h(t) =
rect
Ts
t
1
TS
2

x(t) = rect
t
1
Ti 2

(C.54)
Plot the output signal y1 (t) = (x1 h)(t) for the following cases:
Ti < TS
Ti = TS
Ti > TS
(C.55)
(C.56)
(C.57)
(c) The system is now excited with the signal x2 (t) shown in gure C.9. Plot the output
signal y2 (t).
x2 (t)
1
TS
2TS
3TS
4TS t
1
Figure C.9: Given pulse stream x2 (t)
Analog to Digital Conversion

16. Determine the spectrum (Fourier transform) of the following periodic pulse train
XT (t) =
(t nT )
(C.58)
n=
17. A telephony signal can be interpreted as a low-pass signal with a cut-o frequency of fg =
3.4 kHz.
(a) What is the minimum sampling frequency for perfect reconstruction of the signal
(Nyquist rate)?
C.1
105
Problems
|H(f )|
f2
f1
f1
f2
Figure C.10: Non-ideal low-pass lter with a smooth transition between pass- and stop-band
(b) The sampled signal should be reconstructed by a non-ideal low-pass lter with a lower
stop band frequency f2 and a smooth transition between pass and stop-band (see
Fig C.10). What is the minimum upper pass band frequency f1 and the minimum
sampling frequency for perfect reconstruction of the signal?
18. Imagine you stroll through the forests with a DAT-recorder in order to tape birds singing.
The sampling rate of the DAT-recorder is fs = 48 kHz. A model of the complete recording
and play-back setup is sketched in Figure C.11. The spectrum of the signals to be recorded
is shown in gure C.12 (it should be noted that lower case indices refer to continuous-time
signals while upper-case indices refer to discrete-time i.e., sampled signals).
A/D-
DAT-
D/A-
Converter
Recorder
Converter
Amplier
Microphone
Loudspeaker
Figure C.11: Recording and play-back system
|Sv (f )|
-10
-1
10
f /kHz
Figure C.12: Spectrum (absolute value) of singing birds
(a) Sketch the spectrum |SV (f )| of the recorded signal (after sampling) in the interval
96 kHz f 96 kHz.
(b) What are the necessary characteristics of the D/A-converter (lter characteristics, cuto frequency) that ensure undistorted play-back?
(c) Unfortunately there is your neighbor walking around with his dog. He (your neighbor)
plays an extremely loud dog whistle with a frequency of fh = 43 kHz. Sketch the
spectrum |Sh (f )| (before sampling) and |SH (f )| (after sampling) of the tone produced
by this very loud dog whistle in the interval 96 kHz f 96 kHz.
During play-back you realize that there is not much left of the birds singing when your
neighbor plays his dog whistle.
106
C EXCERCISES
(d) What is the reason for this? Sketch the spectrum |SA (f )| of the composite sampled
signal (birds singing and dog whistle) in the interval 48 kHz f 48 kHz. Describe
the character of the distortion!
(e) Where would an analog lter have to be placed that ensures a proper play-back (perfect
reconstruction) of the birds singing? What are the characteristics of such a lter (type
of lter, cut-o frequency)? Can a digital lter be used too (after sampling)?
(f) You regard such an analog lter as to costly. Moreover, you remember your partyaected ears that are completely deaf for all frequencies above 16 kHz. Therefore, you
decide to give a new dog whistle to your neighbor. Which frequency range (fh < 48 kHz)
is allowed for this new whistle if you want to enjoy the birds singing when playing back
the recording?
|Sf (f )|
-156
-146
146
156
f /kHz
Figure C.13: Spectrum (absolute value) of shrieking bats
(g) After all you want to record bat voices using the same equipment. The spectrum of
bat voices is sketched in Figure C.13. Is it principally possible to record and play back
these voices? If not, why is this not possible? If so, what would you have to change at
the setup with respect to (18b) and (18e) in order to enable perfect play-back? Sketch
the spectrum of the sampled signal |SF (f )| for a deeper understanding!
Bandpass Signals
19. Given is the sent signal

s(t) = Re u(t) ej 2fc t
(C.59)
where u(t) = uI (t) + j uQ (t) is an information bearing complex low-pass signal with a bandlimited spectrum [fT P,min , fT P,max ] with fT P,min < 0 < fT P,max .
(a) Which condition must be fullled for the carrier frequency fc > 0 so that s(t) is a
bandpass signal in interval [fBP,min , fBP,max ] with 0 < fBP,min < fBP,max ?
The signal s(t) is down-converted with a complex carrier signal of frequency fc on the receiver
side, i.e. thus producing a signal as
r(t) = rI (t) + j rQ (t) = s(t) e j 2fc t
(C.60)
(b) Calculate the spectra R(f ), RI (f ) and RQ (f ) as function of S(f ).

(c) Calculate and sketch the spectra RI (f ), RQ (f ) and R(f ) as function of U (f ). Can the
signal u(t) be recovered as interference free from the signal r(t)? If so, which operation /module is necessary for it?
C.1
107
Problems
20. Let s(t) = sI (t) + j sQ (t) be a complex baseband signal. The radio signal (real bandpass
signal) is generated by IQ-modulation

(C.61)
Down conversion as well as low-pass ltering with fcut o = fc is performed at the receiver.
Coherent (phase synchronous) reception is essential for many types of receivers which means

j 2fc t
u(t) = uI (t) + j uQ (t) = sBP (t) e
(C.62)

|f |<fc
(a) Determine the in-phase and quadrature components uI (t) and uQ (t) as a function of
sI (t) and sQ (t). The phase oset between transmitter and receiver is 0 , i.e.
u(t) = sBP (t) e
j(2fc t+0 )
(C.63)
|f |<fc
Which eect does a phase oset of 0 = (or 0 = /2) have?

Analog Modulation
21. A signal which is modelled by
s(t) = a1 cos 2f1 t + a2 cos 2f2 t
(C.64)
with f2 = 2f1 and a = a1 = a2 , should be broadcasted by double sideband amplitude

modulation (DSB). The modulation product m(t) can be described as
m(t) = (s(t) + A) cos 2fc t
(C.65)
where A is the amplitude of the carrier (sinusoid) and fc as the carrier frequency (here
fc = 10f1 ).
(a) Give an inequality for the ratio
than 1.
a
A
that keeps the modulation index =
max(|s(t)|)
A
smaller
(b) Sketch the spectrum of the modulated signal m(t) with A = 4a.
(c) Fig. C.14 shows a simplied envelope detector. The received signal is ltered to suppress
noise and adjacent channels, and rectied eventually giving the absolute value |m(t)|
of the modulated signal. Calculate and sketch the spectrum of |m(t)|.
Hints: The modulation index is always smaller than 1 ( < 1). The following equations
hold (Fourier series expansion):
(1)n
2
4
for full wave rectier: | cos 2fc t| =
cos 22nfc t (C.66)
n=1 (2n 1)(2n + 1)
108
C EXCERCISES
|m(t)|
m(t)
m(t)
+ adjacent channels
BP
s(t) + A
LP
s(t)
HP
Figure C.14: Simplied envelope demodulator (idealized, neglecting noise and channel distortions)
22. Fig. C.15 shows a non-coherent envelope demodulator for the medium wave range
(0.5 MHz < fc < 1.5 MHz). The received signal m(t) is down-converted to a constant intermediate frequency (here fIF < fc , see g. C.16), before it is fed to an envelope
demodulator (see exercise 21).
The base band signals s1 (t), s2 (t) (with a cut o frequency of fg = 4.5 kHz) represent the
programs of two dierent radio stations. In this case the receive signal m(t) is
m(t) = m1 (t) + m2 (t)
= (A1 + s1 (t)) cos 2fc t + (A2 + s2 (t)) cos 2fc,i t
(C.67)
(C.68)
The carrier frequencies of these two stations are fc and fc,i .

envelope demodulator
m(t)
(+ noise)
fIF
s(t)
BP
LP
HP
cos 2fM t
Figure C.15: Superhet
|M (f )|
fc,i fM fc
fc
fM
fc,i
Figure C.16: Spectrum |M (f )| of the received signal at the input of the mixer
(a) Give a relation between the carrier frequency fc , the local oscillator frequency fM , and
the intermediate frequency fIF in the receiver.
(b) Show that after down-conversion the signals s1 (t) at fc and s2 (t) at fc,i can be superimposing each other at the output of the mixer. This means that both stations will
be down-converted to the same frequency range f [fIF fg , fIF + fg ]. Therefore the
demodulation result in a mix of these two stations. Give the equations for the relations
between fc , fc,i , fIF and fM . The frequency fc,i is called image frequency. How can these
image signals be suppressed?
C.1
109
Problems
(c) What is the minimum intermediate frequency fIF that ensures image rejection for all
signals of medium wave range? Assuming the same conditions what would be the IF
for the FM-frequency range (88 MHz < fc < 108 MHz)?
(d) Assuming an IF of fIF = 0.5 MHz, what is the necessary tuning range of fM ?
(e) Which bandwidth is required for the band pass lters?
Digital Modulation
23. The condition for (zero ISI (Inter symbol interference)) is known as 1. Nyquist criteria
and the received impulse h(t) (at output of the receiver lter, see Fig. 8.10 on page 80,
h(t) = hT (t) hR (t)) must fulls the following condition (in time domain):

h(t + nT )

t=0
h(0) = 0 for n = 0
(C.69)
for n = 0
where T is the symbol duration (see Fig. C.17). Show that the 1. Nyquist criterion in the
frequency domain corresponds the following condition

n
H f
= const.
T
n=
(C.70)
h(t)
h(0)
Figure C.17: 1. Nyquist condition in time domain
24. A bit sequence d{k} = {0, 1, 1, 0, 1, 0, 0, 1} for k = 0, 1, 2, . . . , 7 with a bit rate fbit = 1/T is
to be transmitted in the baseband. For that purpose rect-shaped transmission impulses

t
A0
hT (t) =
rect
(C.71)
T
T
are used during bipolar transmission. Moreover symbol mapping of the form b{k} = 2d{k}1
applies, i.e.
d{k} = 1 b{k} = +1 +hT (t) to be sent
d{k} = 0 b{k} = 1 hT (t) to be sent
(C.72)
(C.73)
110
C EXCERCISES
The send signal
18
s(t) has then the form

s(t) =
7

b{k} T hT (t kT )
(C.74)
k=0
= A0
7

b{k} rect(t/T k)
(C.75)
k=0
7

= A0
(2d{k} 1) rect(t/T k)
(C.76)
k=0
(a) Sketch the send signal s(t) in the interval t [2T, 9T ]. Emphasize the signal values
s(kT ), k [0, 7].
The channel will be treated for the time being as ideal, i.e. the received signal r(t) is given
as
r(t) = s(t)
(C.77)
In the receiver the signal r(t) is ltered (receipt lters hR (t)), the ltered signal rE (t) =
r(t) hR (t) is then sampled at time points kT and nally the sampling values are supplied
to a detector.
(b) Outline the block diagram of the entire data transmission system (transmitter, channel
and receiver).
For the received lter hR (t) applies
1
hR (t) =
rect
T

t
T
(C.78)
Def
(c) Sketch the ltered received signal rE (t) = r(t) hR (t) = sE (t) in the interval t
[2T, 9T ]. Also denote the sample values sE {k} = sE (kT ), k [0, 7].
(d) Compute the average power PsE of the sampled signal rE {k} = sE {k} at the entrance
of the detector.
The channel is no more assumed as ideal. The received signal r(t) is to be interpreted as
superposition of the transmitted signals s(t) with a white noise signal n(t) the noise power
density N (f ) = N0 /2 for < f < .
r(t) = s(t) + n(t)
rE (t) = sE (t) + nE (t)
(C.79)
(C.80)
(e) Outline again the block diagram of the entire data transmission system (transmitter,
channel and receiver).
(f) Compute the power density spectrum NE (f ) of the ltered noise signal nE (t) at the
output of the receipt lter hR (t).
NE (f ) = |HR (f )|2 N (f )
18
(C.81)
Such a signal is also designated as pulse amplitude modulated signal (PAM-Signal) (PAM pulse amplitude
modulation).
C.1
111
Problems
(g) Compute the noise power PnE of nE (t) at the output of the receipt lter hR (t).

PnE =
NE (f ) d f
(C.82)
Hint: It applies
si2 (ax) d x =
|a|
(C.83)
Due to the assumed stationary noise signal n(t), the noise power PnE has not been changed
through sampling. The noise power at the entrance of the detector is thereby also PnE .
(h) Compute the signal-to-noise ratio SN R at the entrance of the detector.
Hint: For simplication, both the transmitting and the receipt lters given here are indicated
as non-causal lters. If these lters are to be realized in an instrument, the impulse responses
hT (t) and hR (t) must be delayed by T /2 in each case (see also Art. 4.5.3). From this a total
+ 1} at the receiver one bit duration
delay of T , i.e. a sent bit d{k} thus is detected as d{k
later.
25. BPSK (binary phase shift keying) is used in the following transmission system. The probability of the both transmit symbols d0 , d1 are equal (P (d0 ) = P (d1 ) = 0.5). Assuming a ideal
noise free transmission, the samples rE (kTs ) = Re {rE (kTs )} at the input of the detector
are rE = s0 for the symbol d0 or rE = s1 for the symbol d1 , respectively. Moreover it is
s0 = s1 > 0, Im {rE (kTs )} = 0.
(a) Plot the possible sample values of the received signal rE (kTs ) in the complex plane
(phase diagram at the detector input). Sketch the corresponding conditional probability
density functions (pdf) of the amplitude frE |d0 (rE |d0 ) and frE |d1 (rE |d1 ) for the real part
of the received signal rE (kTs ) = Re {rE (kTs )}.
(b) Now a noise is superimposed on the received signal. Describe the dierences of the new
phase diagram at the receiver. Sketch again the corresponding conditional probability
density function of the amplitude frE |d0 (rE |d0 ) and frE |d1 (rE |d1 ) for the case of additive
white Gaussian noise (AWGN). Mark the area, which corresponds to the probability of
the wrong decision P (d{k}

= d0 | d{k} = d1 ) (conditional probability, that you detect
d{k}
= d0 , even though d{k} = d1 was transmitted).
(c) The receiver is not properly synchronized to the transmitter now (non coherent down
conversion). Assume a phase oset of = 4 . Sketch again the phase diagram of the
samples rE (kTs ). Calculate the oset of the signal to noise ratio (SNR) for unchanged
noise power.
(d) Assuming ideal phase synchronization again. The standard deviation of the noise signal
is = 0.5 V. Calculate the signal amplitude |s0 | = |s1 | (in mV) corresponds to a
bit error rate of BER = 0.1. Take use of tables or math programs like MATLAB,
Mathematica or Maple.
112
C.2
C EXCERCISES
Solutions
1. (a) For further computations the following properties are useful

cos = cos
even function
sin = sin uneven function
(C.84)
(C.85)
e j = cos j sin
(C.86)
cos = Re{e }
(C.87)
sin = Im{e }
(C.88)
Thus the rst addition theorem can be checked

cos(x y) = Re{ej(xy) } = Re{ej x ej(y) }
= Re{(cos x + j sin x)(cos y j sin y)}
= Re{cos x cos y sin x sin y j cos x sin y + j sin x cos y}
= cos x cos y sin x sin y
(C.89)
(C.90)
(C.91)
(C.92)
and the second
& x y x y
'
y
y
x
x
sin x sin y = Im ej( 2 + 2 + 2 2 ) ej( 2 + 2 + 2 2 )
'
& x+y xy
x+y
xy
= Im ej 2 ej 2 ej 2 e j 2
& x+y xy
'
j 2
j 2
j xy
2
= Im e
e
e

x+y
xy
x+y
+ j sin
2 j sin
= Im
cos
2
2
2
xy
x+y
= 2 sin
cos
2
2
(C.93)
(C.94)
(C.95)
(C.96)
(C.97)
(b) These proofs are very simple

z + z = x + j y + x j y = 2x
= 2Re{z}
(C.98)
(C.99)
z z = x + j y x + j y = j 2y
= j 2Im{z}
(C.100)
(C.101)
z z = (x + j y)(x j y)
= x2 j xy + j xy + y 2
= x2 + y 2
= |z|2
(C.102)
(C.103)
(C.104)
(C.105)
and
also
C.2
113
Solutions
2. (a) By rearranging the eqn. (C.20)-(C.22) yields

a0 = 2c0
ak = ck + ck
bk = j(ck ck )
k=0
k = 1, 2, . . .
k = 1, 2, . . .
(C.106)
(C.107)
(C.108)
Using these relations in (C.14), then one gets

s(t) = FR{s(t)} = c0 +
k
k
(ck + ck ) cos 2 t + j(ck ck ) sin 2 t
T
T
k=1
(C.109)
and under consideration of the eqn. (C.28), (C.29) eventually,

k
k
k
k

ej 2 T t + e j 2 T t
ej 2 T t e j 2 T t
s(t) = FR {s(t)} = c0 +
+ j(ck ck )
(ck + ck )
2
2j
k=1
(C.110)
k
k
k
1 j 2 k t
ck e T +ck e j 2 T t +ck ej 2 T t +ck e j 2 T t
2 k=1
= c0 +
+ ck ej 2 T t ck e j 2 T t ck ej 2 T t +ck e j 2 T t
(C.111)
= c0 +
ck ej 2 T t +ck e j 2 T t
(C.112)
k=1
cn ej 2 T t
(C.113)
n=
(b)
1
cn =
T
=
s0
T
s0
T
s0
T
s0
T
T
2
T2
s1 (t) e j 2 T t d t
n
e j 2 T t d t
j 2 n t 2
1
T
e
2
j 2 Tn
j 2 n
n
1
j 2 T
T 2 e
2
=
n e
j 2 T
n
1 1 + j n
=
n
e T e j T
2j
T

sin( n
)
T
n
s0
1
=
n sin
T
T
T
=
(C.114)
(C.115)
(C.116)
(C.117)
(C.118)
(C.119)
114
C EXCERCISES
= s0
For =
n
si
T
T
(C.120)
T
4
s0 n
si
4
4
Fig. C.18 shows the corresponding amplitude spectrum.
(C.121)
cn =
|cn |
s0
4
12
T
T8
T4
4
T
8
T
12
T
f=
n
T
Figure C.18: Amplitude spectrum of a periodic rectangular signal
(c)

S2 (f ) =
= s0
s2 (t) e j 2f t d t
(C.122)
e j 2f t d t
(C.123)
The denite integral in (C.123) was calculated for the case f = Tn in the solution for
subtask (2b) (compare (C.115)-(C.120)). By exploiting the result indicated there one
nally gets
(C.124)
S2 (f ) = s0 si(f )
and with =
T
4
s0
T
T si(f )
4
4
The corresponding amplitude spectrum is represented in Fig. C.19.
S2 (f ) =
(C.125)
|S2 (f )|
s0
T
4
12
T
T8
T4
4
T
8
T
12
T
Figure C.19: Amplitude spectrum of the rect impulse
3. (a) Noise power at the input of the demodulator:

PN,dem = PN + g2
(C.126)
C.2
115
Solutions
= 50 dBm + 60 dB
= 10 dBm
(C.127)
(C.128)
Intelligence signal power at the input of the demodulator:

PT,dem = PT LS + g1 + g2
= 12 dBm 20 lg(100d) dB + 23 dBm + 60 dB
= 95 dBm 20 lg(100d) dB
(C.129)
(C.130)
(C.131)
Signal-noise-ratio SN R at the input of the demodulator:

SN R = PT,dem PN,dem
= 95 dBm 20 lg(100d) dB 10 dBm 7 dB
78 dB 20 lg(100d)
103.9
d
100
d 79.4 m
(C.132)
(C.133)
(C.134)
(C.135)
(C.136)
(b) Solving for PT gives

SN R PT LS + g1 + g2 PN g2
PT SN R + LS g1 + PN
20 dB + 20 lg(5000) dB 23 dB 50 dBm
21 dBm
126 mW
(C.137)
(C.138)
(C.139)
(C.140)
(C.141)
4. (a) From denition

1.)
(x) =

+ for x = 0
(C.142)
for x = 0
(x) d x = 1
2.)
(C.143)
3.)
(x) = (x)
(C.144)
(b) This integral converges to

lim

s(x)n (x x0 ) d x = lim
1
x0 + n
1
x0 n
= s(x0 ) lim
= s(x0 )
n
s(x) d x
2
1
x0 + n
1
x0 n
n
dx
2
(C.145)
(C.146)
(C.147)
116
C EXCERCISES
(c) By integration over x one gets

n (x) =
n () d
x
(C.148)
for x < n1
n
1
2
n
d for
n
2
d for
1
n
n 1
0
n
x+
1
n
x<
1
n
(C.149)
for x < n1
1
2
for
for
1
n
1
n
x<
1
n
(C.150)
The limit value gives

lim n (x) = (x)
(C.151)
with
0
(x) =
for x < 0
(C.152)
for x = 0
for 0 < x
Fig. C.20 shows the function n (x)

n (x)
1
n1
1
n
Figure C.20: Approximation of unit step function (x)
5. The computation of s(t) can take place explicitly using the inverse Fourier transformation
which requires an enormous computation. This can be reduced under utilization of some
properties and correspondences of the Fourier transformation.
(a) Linearity The task can be divided into subtasks (see Tab. A.1)
S(f ) = S1 (f ) + S2 (f )
a
S1 (f ) = ((f + fc ) + (f fc ))
2

f + fc
A
f fc
rect
S2 (f ) =
+ rect
2
2fg
2fg
(C.153)
(C.154)
(C.155)
C.2
117
Solutions
(b) Frequency shifting (Modulation, spectrum shifts towards left and/or right)
Bandpass signals can be analyzed as low-pass signal (see Tab. A.1)

f
1
1
s2 (t) = F {S2 (f )} = A cos(2fc t) F
rect
(C.156)
2fg
(c) Scaling (time suppression
frequency expansion, time expansion
suppression) the bandwidth can be normalized (see Tab. A.1)

f
S
2fg s(2fg t)
2fg
(d) By FT (see Tab. A.2 and A.3)
&
'
1 a
((f + fc ) + (f fc )) = a cos 2fc t
s1 (t) = F
2
frequency
(C.157)
(C.158)
f
ur s1 (t) und mit
f
2fg
rect

2fg si(2fg t)
(C.159)
wird s2 (t) zu
s2 (t) = A cos(2fc t)2fg si(2fg t)
(C.160)
The nal result is

s(t) = s1 (t) + s2 (t) = (a + 2Afg si 2fg t) cos 2fc t
(C.161)
6. Problem: The computation of the Fourier transformation S(f ) assumes the knowledge of
the progression of the function s(t) for < t < . This is practically not possible, so
that only the process can be analyzed within the time window 0 t T for computation.
Short-time spectrum ST (f )
Question: What is the eect of this so called windowing? How should this windowing take
place, so that the deviation is as small as possible between S(f ) and ST (f )?
(a) The computation short-time spectrum ST (f ) can be considered also as Fourier transformation, where all unknown regions of the time signal s(t) be masked by a window
function w(t).

ST (f ) =
j 2f t
s(t) e
dt =
s(t)w(t) e j 2f t d t
(C.162)
with (rect window function)

w(t) = rect
t T /2
T

(C.163)
118
C EXCERCISES
By using convolution theorem

ST (f ) = (S W )(f ) = S(f ) W (f )
s(t)w(t)
(see Art. A.3)
(C.164)
I.e. the short-time spectrum ST (f ) results from the convolution of the spectrum S(f )
and the spectrum W (f ) of the the window function w(t).

(C.165)
ST (f ) = S(f ) T si(f T ) e j f T
Convolution with the sinc function (rect windowing) leads to a kind of Kantenverschleifung to the spectrum. The rect windowing is very simple in its application (computation of short-time spectrum), leads however to a large deviation between ST (f ) and
S(f ). Therefore usually other window functions are used (Blackman, Kaiser, Hamming,
Hanning etc.).
(b) Example for the eect of window length T
1
((f + f0 ) + (f f0 ))
2
cos(2f0 t)
ST (f ) =
(C.166)

T j f T
e
si((f + f0 )T ) e j f0 T + si((f f0 )T ) ej f0 T
2
(C.167)
Fig. C.21 shows the short-time spectrum ST (f ) for dierent window lengths T . It
6
0.7
50
45
0.6
5
40
0.5
35
Amplitude
Amplitude
Amplitude
30
0.4
0.3
25
20
2
15
0.2
10
0.1
5
0
4
1
0
1
normierte Frequency f/f
0
4
1
0
1
0
4
a) |ST (f )| for T = 1/f0
1
0
1
b) |ST (f )| for T = 10/f0
c) |ST (f )| for T = 100/f0
Figure C.21: Amplitude the short-time spectrum |ST (f )|
becomes evident from it that with increasing window length T the deviation becomes
smaller for Fourier transformation.
7. Starting from |x(t)|2 = x(t) x (t), x (t)
X (f ) and x1 (t) x2 (t)
one gets

2
|x(t)| d t =
x(t) x (t) e j 20t d t

= X(f ) X (f )
= F {x(t) x (t)}
f =0
f =0

=
X(u) X ((f u)) d u
f =0
X1 (f ) X2 (f )
(C.168)
(C.169)
(C.170)
C.2
119
Solutions
=

X(u) X (u) d u
Subst. u = f, d u = d f

X(f ) X (f ) d f =
|X(f )|2 d f
(C.171)
(C.172)
8. (a) First the spectrum of the function g1 (t) = et is computed:

2
et e j 2f t d t
G1 (f ) =

2
t2
e
cos 2f t d t j
et sin 2f t d t
=
(C.173)
(C.174)
Since both g1 (t) is and cos t each are even function and sin t an odd function

2
et sin 2f t d t = 0
(C.175)
then [BS79]
et cos 2f t d t
G1 (f ) = 2
0
2 f 2
= e
(C.176)
(C.177)
Since
g(t) =
=
1
2 2
1
2 2
t2
e 22
g1 (
(C.178)
t
2 2
(C.179)
the spectrum G(f ) can be calculated with the help of the scaling property (see Tab. A.1)
from G1 (f ).
1
G(f ) =
2 2 G1 ( 2 2 f )
(C.180)
2 2
1
(C.181)
= G1 ( 2 2 f )
= e2
2 2 f 2
(b) The area A under g(t) (dc component) can be given as

g(t) d t =
g(t) e j 20t d t
A=

= G(f ) f =0 = G(0) = 1
(C.182)
(C.183)
(C.184)
9. (a) In order to compute the Hilbert transformation, the detor over the fourier transformation the usually is the simpler solution method. However both ways are to be demonstrated here.
xH (t) = H {x(t)}
(C.185)
120
C EXCERCISES
1
= A0 cos 2f0 t
t

1
A0
=
cos 2f0 (t ) d

sin 2f0
cos 2f0
A0
cos 2f0 t
=
+ sin 2f0 t
d

even function
uneven function 0
sin 2f0
2
= A0 sin 2f0 t
d
0

sine integral
(C.186)
(C.187)
(C.188)
(C.189)
= A0 sin 2f0 t
(C.190)
bzw.
xH (t) = F 1 {F {x(t)} ( j) sgn(f )}
= F 1 {F {A0 cos 2f0 t} ( j) sgn(f )}

1
1
A0 ((f + f0 ) + (f f0 )) ( j) sgn(f )
=F
2

j
1
=F
A0 ((f + f0 ) (f f0 ))
2
= A0 sin 2f0 t
(C.191)
(C.192)
(C.193)
(C.194)
(C.195)
(b) For computing the spectrum XA (f ) we may use the result from the last subtask (see
Fig. C.22)
XA (f ) = F {xA (t)} = F {x(t) + j xH (t)}
= X(f ) + j XH (f )
1
j
= A0 ((f + f0 ) + (f f0 )) + A0 j((f + f0 ) (f f0 ))
2
2
A0
A0
((f + f0 ) + (f f0 ))
((f + f0 ) (f f0 ))
=
2
2
= A0 (f f0 )
X(f )
XA (f )
A0
f0
A0
f0
f0
f0
Figure C.22: Spectrums of the signals x(t) and xA (t)
the spectrum XA (f ) is single-sided, but the signal xA (t) is complex,

the spectrum X(f ) is double-sided, but the signal x(t) is real.
(C.196)
(C.197)
(C.198)
(C.199)
(C.200)
C.2
121
Solutions
(c) The signal xA (t) possesses an constant amplitude (constant envelope).

|xA (t)| = |x(t) + j xH (t)|
= |A0 (cos 2f0 t + j sin 2f0 t)|
(

= A20 cos2 2f0 t + sin2 2f0 t

A0 0
A0 for
=
A0 < 0
A0 for
10. (a) For xg (t) applies
(C.201)
(C.202)
(C.203)
(C.204)
1
F {xg (t)} = F
(x(t) + x(t))
(C.205)
2

1
(C.206)
(x(t) + x(t)) e j 2f t d t
Xg (f ) =
2

x(t) + x(t)
x(t) + x(t)
cos 2f t d t j
sin 2f t d t (C.207)
=
2
2

=

even function
uneven function 0
(x(t) + x(t)) cos 2f t d t

(C.208)
uneven function
x(t) cos 2f t d t

j 2f t
x(t) e
dt
= Re
(C.209)
= Re {F {x(t)}}
(C.211)
(C.210)
and for xu (t) gilt

1
(x(t) x(t))
F {xu (t)} = F
2

x(t) x(t)
x(t) x(t)
Xu (f ) =
cos 2f t d t j
sin 2f t d t
2
2

uneven function 0
even function

= j
(x(t) x(t)) sin 2f t d t

0
uneven function

= j
x(t) sin 2f t d t
j 2f t
= j Im
x(t) e
dt
(C.212)
(C.213)
(C.214)
(C.215)
(C.216)
= j Im {F {x(t)}}
(C.217)
(b) With eqn. (C.48)

x(t) = xg (t) + xu (t)
(C.218)
122
C EXCERCISES
= xg (t) + xg (t) sgn(t)
(C.219)
since x(0) = 0 also applies the inversion

= xu (t) sgn(t) + xu (t)
X(f ) = Xu (f ) F { sgn(t)} + Xu (f )
(C.220)
(C.221)
mit X(f ) = Xg (f ) + Xu (f ) folgt

Xg (f ) = Xu (f ) F { sgn(t)}
1
= ( j) Xu (f )
t
= ( j) H {Xu (f )}
(C.222)
(C.223)
(C.224)
11. By causality one generally understands the cause-eect principle, i.e. each eect excludes
one or more timely back dated cause(n). An eect can however never occur before the
cause takes place.
The way a regarded LTI system is indicated here is that an initial value y(t1 ) may depend
only on the past input value(s) x(t) with t < t1 and NOT on future or current values19 . If
this condition is fullled, the system is called strictly causal, otherwise not strictly causal.
The signal y(t1 ) must be independent of x(t) for t t1 .

y(t1 ) =
x( ) h(t1 ) d

t1
x( ) h(t1 ) d +
x( ) h(t1 ) d
=
t1

causal part
(C.225)
(C.226)
anti-causal part
I.e. the second term must be equal to zero. Since for the signal x(t) no restrictions are to be
made, the condition is fastened with the impulse response.

x( ) h(t1 ) d

h(t1 ) t1 = 0

h(t)t0 = 0
0=
(C.227)
t1
(C.228)
(C.229)
12. (a) The triangular function triang(x) arises as a result of unique convolution of the rect
function rect(x) with itself.

triang(x) = rect(x) rect(x) =
rect() rect(x ) d
(C.230)
To compute a distinction of cases is necessary

19
Sometimes however dependence on the current input value is also permitted (strictly causal causal)
C.2
123
Solutions
1. Case: x < 1
rect() rect(x ) d = 0
(C.231)
2. Case: 1 x < 0

rect() rect(x ) d =
x+ 21
12
1d = 1 + x
(C.232)
1d = 1 x
(C.233)
3. Case: 0 x < 1

rect() rect(x ) d =

4. Case: 1 < x
1
2
x 21
rect() rect(x ) d = 0
(C.234)
Therewith one gets (see Fig. C.23)
1 + x for
triang(x) =
1t<0
1 x for 0 t < 1
0
(C.235)
otherwise
triang(x)
1
Figure C.23: The triangular function triang(x) = rect(x) rect(x)
To compute the convolution product proceeds in same manner

y(x) = triang(x) rect(x)

=
triang() rect(x ) d
(C.236)
(C.237)
1. Case: x < 32
triang() rect(x ) d = 0
(C.238)
2. Case: 32 x < 12

triang() rect(x ) d =
x+ 12
1
1
3
9
(1 + ) d = x2 + x +
2
2
8
(C.239)
124
C EXCERCISES
3. Case: 12 x <

1
2
4. Case:

x<
5. Case:
1
2
3
2
x+ 12
0
3
4
(C.240)
1
3
9
(1 ) d = x2 x +
2
2
8
x 21
(C.241)
x 12
(1 + ) d +
(1 ) d = x2 +
3
2

triang() rect(x ) d = 0
(C.242)
So one gets (see Fig. C.24)
y(x) =
1 2
x + 32 x +
x2 + 3
1 2
4
3
x
2
9
8
9
8
for
for
for
1
2
3
2
1
2
x < 12
x<
x<
1
2
(C.243)
3
2
otherwise
y(x)
1
2
Figure C.24: Convolution product y(x) = rect(x) rect(x) rect(x)
(b) There are two ways to calculate the spectrum F { triang(t/T )}.
By Fourier transformation of triang(t/T ) explicitly.

triang(t/T ) e j 2f t d t
F { triang(t/T )} =
(C.244)
This method wont be shown here.

We know that
triang(t/t) = rect(t/T ) rect(t/T )
(C.245)
applies, as well as (see Tab. A.3)

rect(t)
si(f )
(t, f dimensionless)
(C.246)
C.2
125
Solutions
F { triang(t/T )}
T1
1
T
Figure C.25: Spectrum of the triangular function F { triang(t/T )}
By exploiting the convolution and scaling properties we get (see Fig. C.25)
F { triang(t/T )} = T si(f T ) T si(f T )
= T 2 si2 (f T )
13. The spectrum is given as
+f
A ff22f
A
H(f ) =
f
A ff22f
for
f2 f < f1
for
f1 f < f1
(C.247)
(C.248)
(C.249)
for f1 f < f2
otherwise
There are (at least) 2 solutions.

With the help of the inverse Fourier transformation, the impulse response h(t) can be
computed directly.

h(t) =
H(f ) ej 2f t d f
(C.250)
da H(f ) = H(f )

=2
H(f ) cos 2f t d f
0
f1
= 2A
(C.251)
f2
cos 2f t d f + 2A
0
f1
f2 f
cos 2f t d f
f2 f1
(C.252)
With the help of the relations

cos ax x sin ax
+
(C.253)
x cos ax d x =
a2
a
xy
x+y
sin
(C.254)
cos x cos y = 2 sin
2
2
gives
f2
f1
f2

A
A
A
f
cos
2f
t
f
sin
2f
t
2

h(t) =
sin 2f t
sin 2f t +
+
t
t f2 f1
t 2(f2 f1 )t
f2 f1 f1
0
f1
(C.255)
126
C EXCERCISES
A cos 2f1 t cos 2f2 t

t
2(f2 f1 )t
= A(f1 + f2 ) si((f1 + f2 )t) si((f2 f1 )t)
(C.256)
(C.257)
Convolution of two rect functions generates the triangle function. The two square pulses
are developed dierently in a trapezoidal process. Thus H(f ) can be also represented
as (fa > fb )

f
f
A
rect
(C.258)
rect
H(f ) =
2fb
2fa
2fb
The corner frequency can be given as
f1 = fa fb
f2 = fa + f b
f1 + f2
2
f2 f1
or fb =
2
or fa =
(C.259)
(C.260)
Since convolution in frequency domain corresponds to a multiplication in time domain,

the impulse response h(t) can be indicated as (see Tab. A.1)

f
f
A
1
1
rect
F
rect
(C.261)
F
h(t) =
f2 f1
f1 + f2
f2 f1
With help of the correspondence of the rect function (see Tab. A.3) and of the scaling
property (see Tab. A.1) nally results in
h(t) = A(f1 + f2 ) si((f1 + f2 )t) si((f2 f1 )t)
(C.262)
The rst solution method always leads (slowly but surely) to the goal, however is at the cost
of cpu time. The second version is clearly faster and more elegant, requires however a certain
amount of experience. Usually only a few correspondences to the Fourier transformation are
necessary in order to solve more complicated problems. Fig. C.26 shows the process of h(t)
with f2 = 2f1 .
14. (a) The output signal y(t) results from the convolution of the signals x(t) and h(t).

x( ) h(t ) d
(C.263)
y(t) = (x h)(t) =

m(,t)
It is now to be demonstrated that the convolution of time-limited signals results in

again a time-limited signal. The signal y(t) can be zero for any time-limited signal x(t)
and h(t), when m(, t) = 0 is valid for any , i.e.
x(),h(), m(, t) = x( ) h(t ) = 0
(C.264)
Therefore x() = 0 or h() = 0. It applies

x( ) = 0 when
h(t ) = 0 when
0 > > T1
0 > t > T2
(C.265)
(C.266)
127
Solutions
3
2.5
2
h(t)/(A f )
C.2
1.5
1
0.5
0
0.5
2
1.5
0.5
0
t/f
0.5
1.5
Figure C.26: Impulse response h(t) of the low-pass with nite area
t T2 < < t
(C.267)
Thereby two coherent intervals develop in each case

x( ) = 0 for (, 0) (T1 , )
h(t ) = 0 for (, t T2 ) (t, )
(C.268)
(C.269)
and m(, t) = 0 Hence by combining (intersection) these intervals

m(, t) = 0 for (, max(0, t T2 )) (min(T1 , t), )
(C.270)
The convolution integral y(t) will be zero when

m(, t) = 0 for (, )
(C.271)
max(0, t T2 ) > min(T1 , t)
(C.272)
t > T1 + T2
t<0
(C.273)
(C.274)
y(t) = 0 for 0 > t > T3 = T1 + T2
(C.275)
gilt, d.h.
This is fullled for
This yields nally

Fig. C.27 should clarify these situations.
(b) The impulse response h(t) is for 2T t T and t = 6T non zero. Thus the signal
y(t) results as follows at time t = 0:

y(0) =
x( ) h( ) d
(C.276)
128
C EXCERCISES
x(t)
h(t)
T1
a) Input signal x(t) and impulse response h(t)
T2
x( ), h(t )
t T2
T1
b) 1. Fall: t < 0 y(t) = 0
x( ), h(t )
t T2
T1
c) 2. Fall: 0 t T3 = T1 + T2
x( ), h(t )
t T2
T1
d) 3. Fall: t > T3 = T1 + T2 y(t) = 0
Figure C.27: Convolution of time-limited signals
2T
= x(6T ) +
x( ) d
(C.277)
Hence it follows that the signal x(t) at the time t = 6T and in the interval [T, 2T ]
must be known, in order to compute the output signal y(t) at time t = 0.
15. (a) The evaluation of convolution shows

y(t) =
A
=
Ts
x( ) h(t ) d
t
A
x( ) d =
Ts
tTS
(C.278)

Ts
x(t ) d
(C.279)
Short-time integrator, moving average, low-pass lter

(b) Since both signals are causal, i.e. x1 (t) = 0 and h(t) = 0 for t < 0, it simplies the
integration limit of the convolution (applies generally)

y(t) = (x1 h)(t) =
x1 ( ) h(t ) d
0
(C.280)
C.2
129
Solutions
Fig. C.28 illustrates the convolution graphically. The calculation yields (see result in
Fig. C.29)

A max(min(t,Ti ),0)
d
(C.281)
y1 (t) =
Ts min(max(0,tTs ),Ti )
h(t )
t TS
t
A
Ts
x1 ( )
t
Ti
Figure C.28: Graphical illustration of the convolution operation
y1 ( )
A TTsi
Ti
a) 1.SFall: Ti < TS
Ti + TS
y1 ( )
A
b) 2.S Fall: Ti = TS
Ti + TS = 2TS
y1 ( )
A
c) 3.SFall: Tii > TS
Ti + TS t
Figure C.29: Result from convolution of two square pulses
Ti TS must be valid for the input impulse x1 (t), in order to get the full amplitude A.
The output signal y1 (t) has a duration of Ti + TS .
(c) See Fig. C.30
(d) From the last subtasks it becomes evident that the input impulse must reach at least
the duration of the impulse response.
TG =
1
2fg
(C.282)
130
C EXCERCISES
y2 (t)
A
0
TS
2TS
3TS
4TS
5TS t
A
Figure C.30: Output signal y2 (t)
Ti TG =
16. The function XT (t) =
+
!
1
= 50 ns
2fg
(C.283)
(t nT ) can be extented as periodic signal with the period T
n=
into Fourier series:

XT (t) =
nt
cn ej 2 T
(C.284)
n=
mit
1
cn =
T
T
2
T2
nt
XT (t) e j 2 T d t

nt
1
(t) e j 2 T d t (Masking property)
=
T
n0
1
= e j 2 T
T
1
=
T
(C.285)
(C.286)
(C.287)
(C.288)
and hence
1 j 2 n t
e T
XT (t) =
T n=
1
n
f
F {XT (t)} =
T n=
T
1
X 1 (f )
T
T
XT (t)
(C.289)
(C.290)
(C.291)
1
X 1 (f )
T
T
(C.292)
17. The voice signal contains speech information in frequency range 0.3 < f < 3.4 kHz (thus
can be regarded as BP signal). However, part of speech, interfering signals and/or noise can
be present above the fundamental frequency fg = 3.4 kHz. Therefore an anti-aliasing lter
is needed before the sampler.
C.2
131
Solutions
(a) minimum sampling rate (Nyquist rate)

fg = 3.4 kHz fs =
1
= 2fg = 6.8 kHz
T
(C.293)
Since LP lters with nite slope are used as anti- aliasing lter, the sampling rate is to
be selected accordingly (for telephone signals fs = 8 kHz is selected (PCM technology)).
(b)
f1 fg , otherwise distortions develop by pass-band droop (linear distortion of the

pass band)
fs fg f2 fs fg + f2 , otherwise distortions occurred due to insucient
image rejection (see also Fig. C.31)
|H(f )|, |Ss (f )|
fg
fs fg
f2
fs
Figure C.31: Signal reconstruction with LP-Filter H(f ) of nite steepness
18. Sampling (time discretization) results images, i.e. the original spectrum is continued periodically. The value discretization (quantization) is left unconsidered.
(a) The spectrum |SV (f )| of the sampled signals (bird voice) is shown in Fig. C.32.
|SV (f )|
-96
-48
48
96
f /kHz
Figure C.32: Amplitude spectrum of bird voice
(b) That connected analog lter of D/A converter is a reconstruction lter and it suppresses
the images (spectral copies) of the former time-discrete signal. Since the original signal
is to be reconstructed here, a low-pass lter with a maximum fundamental frequency
fg = fs /2 = 24 kHz is needed.
(c) The spectrum |SH (f )| of the sampled signal (dog whistle) is shown in Fig. C.33.
|SH (f )|
-96
-48
48
96
f /kHz
Figure C.33: Amplitude spectrum of dog whistle
132
C EXCERCISES
(d) Since no anti-aliasing lter is attached with the A/D converter, aliasing errors may
occur. Exactly what happens here. An image (image frequency) of the dog whistle
fh = 43 kHz lies in the audible range fH = fs fh = 5 kHz. This tone hides the bird
voice due to its loudness. Fig. C.34 shows this superimposition.
|SA (f )|
-48
-24
24
48
f /kHz
Figure C.34: Amplitude spectrum of the superimposed signals (bird voice and dog whistle)
(e) An anti-aliasing lter is needed before the A/D conversion. This is a low-pass lter
with a maximum fundamental frequency fg = fs /2 = 24 kHz.
(f) Since you cannot hear tones above 16 kHz, the whistle may work within the range
16 kHz f < 24 kHz. Since moreover, the range 24 kHz f < 32 kHz drop behind
by aliasing into latter range, the permitted range of the dog whistle is 16 kHz f <
32 kHz.
(g) In principle it is possible to take up the bat voices and can be accurately reproduced.
The LP lter as anti-aliasing or Reconstruction lters becomes now bandpass lters
with the cut-o frequencies fg,u = 144 kHz, fg,o = 168 kHz (the loudspeaker must
be able to work within this range). This procedure is called under-sampling and the
following conditions must be kept: 1) the sampling rate must at least be double of the
signal bandwidth B = fupper flower ,
2) no integral multiple of the half sampling rate may lie in the desired frequency range.
Here both conditions are fullled.
19. (a) The real bandpass signal s(t) can be represented through its quadrature components
uI (t) = Re{u(t)} (in phase component) and uQ (t) = Im{u(t)} (quadrature phase component) (see Fig. C.35).
s(t) = Re{u(t) ej 2fc t }
= Re{(uI (t) + j uQ (t)) (cos 2fc t + j sin 2fc t)}
= uI (t) cos 2fc t uQ (t) sin 2fc t
(C.294)
(C.295)
(C.296)
For the computation of the originating spectrum proven in eqn. (C.99) in context with
1
1
Re{x} = x + x
2
2
(C.297)
is used. Consequently in time domain applies

s(t) = Re{u(t) ej 2fc t }
1
1
= u(t) ej 2fc t + u (t) e j 2fc t
2
2
(C.298)
(C.299)
C.2
133
Solutions
cos 2fc t
Re{u(t)} = uI (t)
Im{u(t)} = uQ (t)
s(t)
sin 2fc t
Figure C.35: Up-mixing of a complex low-pass signal u(t) into a real band-pass lter signal
s(t)
The Fourier transformation of s(t) results with x (t) = X (f ) (see Tab. A.1)
1
1
S(f ) = U (f fc ) + U ((f + fc ))
(C.300)
2
2
Fig. C.36 shows exemplary the spectrum U (f ). In practical applications |fT P,min | =
|fT P,max | is valid, however, more general case will be considered here. The spectra S(f )
U (f ) = URe (f ) + j UIm (f )
fT P,min
fT P,max f
U (f ) = URe (f ) j UIm (f ) U (f ) = URe (f ) j UIm (f )
fT P,min
fT P,max f
fT P,min
fT P,max f
Figure C.36: Real part (solid line) and imaginary part (dotted line) of U (f ), U (f ), U (f )
is yield by right or left shift of the spectra U (f ) or U (f ). That real part of S(f )
is an even function, as well as the imaginary part of S(f ) is an odd function (see
Fig. C.37). The corresponding time function s(t) is therefore (as expected) pure real.
fT P,min < f < fT P,max should be valid for the spectrum U (f ) with fT P,min < 0 and
S(f ) = SRe (f ) + j SIm (f )
1
2 U ((f
1
2 U (f
+ fc ))
fc fc fT P,min
fc + fT P,min
fc )
fc
Figure C.37: Real (solid line) and imaginary part (dotted line) of S(f )
fT P,max > 0. Then applies for the spectra U (f fc ): fc + fT P,min < f < fc + fT P,max or
for U ((f + fc )): fc fT P,max < f < fc fT P,min . Hence these two spectrums are
not overlapping, must consequently applies
(fc + fT P,min ) < fc + fT P,min
fc > |fT P,min |
(C.301)
134
C EXCERCISES
(b) A phase synchronous down-conversion takes place in the receiver (the pertinent lowpass ltering will be considered later)
r(t) = rI (t) + j rQ (t) = s(t) e j 2fc t
(C.302)
From the (Frequency) shifting theorem (see Tab. A.1)we get

r(t)
R(f ) = S(f + fc )
(C.303)
The spectra RI (f ) = F {rI (t)} and RQ (f ) = F {rQ (t)} are yield by utilizing eqn. (C.99,
C.101) similar to
rI (t) = Re{r(t)} =

1
r(t) + r (t)
2
(C.304)

1
S(f + fc ) + S ((f fc ))
2
RI (f ) =
(C.305)
and
rQ (t) = Im{r(t)} =
S(f ) =
(C.306)

1
S(f + fc ) S ((f fc ))
2j
(C.307)

1
U (f fc ) + U ((f + fc ))
2
(C.308)
RQ (f ) =
(c) With

1
r(t) r (t)
2j
yields
1
U (f ) + U ((f + 2fc ))
2)
*
1

1 1
RI (f ) =
U (f ) + U (f ) + U (f 2fc ) + U ((f + 2fc ))
2 2

2
(C.309)
R(f ) =
UI (f )=F {uI (t)}
1

1 1
RQ (f ) =
U (f ) U (f )
U (f 2fc ) U ((f + 2fc ))
2 2j
2j

UQ (f )=F {uQ (t)}
(C.310)
*
(C.311)
By superimposition of the two spectrums U (f ) and U (f ), is it not possible to reconstruct the entire complex signal from rI (t) or rQ (t) (see Fig. C.38). For the reconstruction of u(t) one needs r(t). This can be done via a simple low-pass lter with a
fundamental frequency of fg = fc . Filtering of rI (t) or rQ (t) leads to the reconstruction
of uI (t) or uQ (t). In communications the principle described in this task is called I/Q
demodulation (see Fig. C.39).
20. (a) The receipt signal can be disassembled into its quadrature components rI (t) and rQ (t).

j(2fc t+0 )
r(t) = sBP (t) e
(C.312)

|f |<fc
C.2
135
Solutions
R(f ) = RRe (f ) + j RIm (f )
2fc
2fc
2fc
2fc
RI (f )
2fc
RQ (f )
2fc
Figure C.38: Real (solid line) and imaginary part (dotted line) of the spectra R(f ), RI (f ),
RQ (f )
cos 2fc t
rI (t)
TP
1
u (t)
2 I
= 12 Re{u(t)}
1
u (t)
2 Q
= 12 Im{u(t)}
fg = fc
s(t)
rQ (t)
TP
sin 2fc t
Figure C.39: phase synchronous down-mixture I/Q-Demodulator

= (sI (t) cos 2fc t sQ (t) sin 2fc t)(cos(2fc t + 0 ) j sin(2fc t + 0 ))
|f |<fc
(C.313)
rI (t) = Re{r(t)}

= sI (t) cos(2fc t) cos(2fc t + 0 ) sQ (t) sin(2fc t) cos(2fc t + 0 )
(C.314)
|f |<fc
(C.315)
rQ (t) = Im{r(t)}

= sI (t) cos(2fc t) sin(2fc t + 0 ) + sQ (t) sin(2fc t) sin(2fc t + 0 )
(C.316)
|f |<fc
(C.317)
136
C EXCERCISES
Hence yields
and
1
1
rI (t) = sI (t) cos 0 + sQ (t) sin 0
2
2
(C.318)
1
1
rQ (t) = sI (t) sin 0 + sQ (t) cos 0
2
2
(C.319)
21. (a) In two-sideband amplitude modulation (DSB) the following condition must be kept(A >
0)
s(t) + A 0 t
(C.320)
i.e. the information of the signal s(t) is exclusively contained in the envelope of the
modulated signal m(t). This is equivalent to the following condition
=
max(|s(t)|)
1
A
(C.321)
Hence
max(|a(cos 2f1 t + cos 4f1 t)|)
1
A
2a
1
A
1
a
A
2
(C.322)
(C.323)
(C.324)
(b) The modulated signal results as

m(t) = (a1 cos 2f1 t + a2 cos 2f2 t + A) cos 2fc t
= a(cos 2f1 t + cos 4f1 t + 4) cos 20f1 t
Fig. C.40 shows the spectrum M (f )
correspondences from Tab. A.2. It gives
(C.325)
(C.326)
m(t), which can be easily indicated with
a
(f + 8f1 ) + (f + 9f1 ) + (f + 11f1 ) + (f + 12f1 ) + (f 8f1 )
4

+ (f 9f1 ) + (f 11f1 ) + (f 12f1 ) +2a (f + 10f1 ) + (f 10f1 ) (C.327)
M (f ) =
M (f )
2a
a
4
10f1
10f1
Figure C.40: Spectrum M (f ) of the transmitted signal m(t)
C.2
137
Solutions
(c) As it was presupposed here that no over modulation takes place, i.e. 1 can be used
|m(t)| = | (s(t) + A) cos 2fc t|

(C.328)
>0
= (s(t) + A) | cos 2fc t|

(C.329)
"
#

(1)n
2
cos 22nfc t
(C.330)
= (s(t) + A) 1 2
(2n
1)(2n
+
1)
n=1
"
#

cos
22nf
t
cos
22nf
t
2
c
c
(1)n+1
(1)n+1
=
A + s(t) + 2A
+ 2s(t)
21
21
4n
4n
n=1
n=1
(C.331)
By Fourier transformation yields
MB (f ) = F {|m(t)|}
"

(f + 2nfc ) + (f 2nfc )
2
=
A(f ) + S(f ) + A
(1)n+1
4n2 1
n=1
#

n+1 (f + 2nfc ) + (f 2nfc )
(1)
+S(f )
4n2 1
n=1
"

2
(f + 2nfc ) + (f 2nfc )
=
(1)n+1
A(f ) + S(f ) + A
4n2 1
n=1
#

n+1 S(f + 2nfc ) + S(f 2nfc )
+
(1)
4n2 1
n=1
(C.332)
(C.333)
(C.334)
(C.335)
(C.336)
The amplitude spectrum of the rectied signal is can be visualized from Fig. C.41.
|MB (f )|
8a
1
n2
20f1
20f1
Figure C.41: Amplitude spectrum |MB (f )| of the rectied signal |m(t)|
22. (a) We get the following signal by mixing

g(t) = m(t) cos 2fM t
(C.337)
138
C EXCERCISES
= (A1 + s1 (t)) cos 2fc t cos 2fM t + (A2 + s2 (t)) cos 2fc,S t cos 2fM t
(C.338)
Using addition theorem

1
cos cos = (cos( ) + cos( + ))
2
(C.339)
originates
A1 + s1 (t)
(cos(2(fc fM )t) + cos(2(fc + fM )t))
2
A2 + s2 (t)
+
(cos(2(fc,S fM )t) + cos(2(fc,S + fM )t))
2
g(t) =
(C.340)
With the stipulations y1 (t) = A1 + s1 (t) and y2 (t) = A2 + s2 (t), as well as Y1 (f ) =

F {y1 (t)} and Y2 (f ) = F {y2 (t)} gives the spectrum G(f ) at the output of the mixer
as
1
G(f ) = (Y1 (f + (fc fM )) + Y1 (f (fc fM )) + Y1 (f + (fc + fM )) + Y1 (f (fc + fM ))
4
+ Y2 (f + (fc,S fM )) + Y2 (f (fc,S fM )) + Y2 (f + (fc,S + fM )) + Y2 (f (fc,S + fM )))
(C.341)
Fig. C.42 shows exemplarily the spectrum G(f ) at the output of the mixer, where only
used-signal channel 1 is shown for simplication. The inuence of the interference-signal
channel 2 will be considered later with image frequency fc,S .
|G(f )|
IF-BP
(fc + fM )
fM fc
IF-BP
(fc fM ) (fc fM )
fc
fM
fc + f M
Figure C.42: Receipt spectrum |G(f )| at the mixer output with fM > fc , fZF = fM fc while
considering the simplied form y2 (t) = A2 + s2 (t) = 0
Thus one receives two dierent possible intermediate frequencies fZF , of which one lies
below and one above the carrier frequency fc .
fZF,1 = |fM + fc |
fZF,2 = |fM fc |
(C.342)
(C.343)
Naturally one selects fZF < fc . For a selected intermediate frequency fZF also two
dierent heterodyne frequencies are possible
fM,1 = |fc + fZF |
fM,2 = |fc fZF |
(C.344)
(C.345)
As will be shown later that the heterodyne frequency fM is selected mostly above the
carrier frequency fM > fc .
C.2
139
Solutions
(b) The antenna is usually interpreted as highly broadband and receives therefore (intended) the entire MW range. At the mixer input therefore not only the utilizable
channel (here indicated as s1 (t)), but also all adjacent channels would rest. If the so
called image frequency fc,S lies with same power for such a interference channel (here
marked as s2 (t)), both channels at the output of the mixer would be superimposed.
In order to clarify this eect, the spectrum at the output of the mixer (eqn. C.341)
will be considered. A superposition of the spectra of Y1 () and Y2 () occurs if due to
|G(f )|
fc,SfM fc
fZF
fZF
fc fM fc,S
Figure C.43: Receipt spectrum |G(f )| at the Mixer output with fM > fc , fZF = fM fc < fc
and interference channel with image frequency fc,S = 2fM fc , both channels
are superimposed at fZF and are not separable any more
the symmetrical spectrum one of the two possible intermediate frequency of signal 1
corresponds with one of the two possible intermediate frequency of signal 2.
fZF,Signal 1 = fZF,Signal 2
(C.346)
fZF = |fM fc | = |fM fc,S |
(C.347)
From eqn. (C.342, C.343)

Here a distinction of cases with fM , fZF , fc , fc,S > 0 and fc = fc,S must be accomplished.
It becomes evident that the frequency pair fM + fZF and |fM fZF | are always formed
from the carrier and the corresponding image frequency.
fc,S = 2fM fc
fc,S = 2fM + fc
for fM > fZF

for fM < fZF
(C.348)
(C.349)
From Fig . C.43 it follows that the unwanted superposition and distortions comes
from intermediate frequencies. This disturbance can be avoided, if a bandpass lter is
attached to the mixer around fc .
(c) Hence the image frequency fc,S lies always outside of the utilizable region, so that the
frequency pair must belong together
fM + fZF
and |fM fZF |
(C.350)
with the dierence

|fc fc,S | = 2fZF
|fc fc,S | = 2fM
for fM > fZF

for fM < fZF
(C.351)
(C.352)
be larger than the utilizable region fc,max fc,min . Since a down-mixture always taken
place in the receiver, the interesting case is fM > fZF and therewith
fZF >
fc,max fc,min
2
(C.353)
140
C EXCERCISES
MW:
UKW:
0.5 MHz fc 1.5 MHz

88 MHz fc 108 MHz
fZF > 0.5 MHz

fZF > 10 MHz
(d) MW: fZF = 0.5 MHz (in practice mostly fZF = 0.455 MHz is selected Standard
lter)
fM,min = fc,min fZF = 0
fM,max = fc,max fZF = 1 MHz
(C.354)
(C.355)
fM,min = fc,min + fZF = 1 MHz

fM,max = fc,max + fZF = 2 MHz
(C.356)
(C.357)
or
Here only the adjustable region fM [1 MHz, 2 MHz] is important.

USW: fZF = 10 MHz (in practice mostly fZF = 10.7 MHz is selected Standard lter)
78 MHz fM 98 MHz
98 MHz fM 118 MHz
(C.358)
(C.359)
Here both adjustable regions are realizable. As the ration of variation fM,max /fM,min
should be kept as small as possible, usually
fM > fc
(C.360)
is selected.
(e) The bandpass lter (around fc ) prior to the mixer must suppress the image frequency
signals suciently, meanwhile bandpass lter (around fZF ) following the mixer accomplishes the actual channel splitting. So
fg = 5 kHz
B = 2fg = 10 kHz
(C.361)
The ratio of bandwidth and center frequency of the bandpass lter determines its
necessary quality. The smaller the ratio the better is the quality. Therefore as small an
intermediate frequency as possible is aimed at in order to keep the quality of necessary
lter small.
23. From h(t)
H(f ) we get

h(nT ) =
H(f ) ej 2nT f d f

k
k
H f+
ej 2nT (f + T ) d f
=
1
T
k= 2T

1
2T
k
k
T df
H f+
ej 2nT f ej 2nT
=

1
T
2T
k=
1

1
2T
Hp (f )
(C.362)
(C.363)
(C.364)
C.2
141
Solutions

=
1
2T
Hp (f ) ej 2nT f d f
1
2T
(C.365)
As Hp (f ) is a periodic function of 1/T , it can be expanded in a Fourier series
Hp (f ) =
cm ej 2mT f
(C.366)
m=
cm = T
From eqn. (C.365) follows

cm =
1
2T
1
2T
Hp (f ) e j 2mT f d f
(C.367)
T h(0) for m = 0
0
(C.368)
otherwise
And hence (see Fig. C.44)

n
Hp (f ) =
H f
= T h(0) = const.
T
n=
(C.369)
Such an impulse h(t) after eqn. (C.69) s designated also as Nyquist-1-pulse. A lter having
impulse response which satisfy this condition is called Interpolator lter.

H f Tn
Hp (f )
T h(0)
1
2T
1
2T
Figure C.44: 1. Nyquist condition in frequency domain
24. (a) see Fig. C.45

A0
s(t)
{0}
2T
T
A0
{1}
{1}
{0}
{1}
{0}
{0}
{1}
2T
3T
4T
5T
6T
7T
8T
9T t
Figure C.45: Transmitted signal s(t)
142
C EXCERCISES
DACTransmitting lterChannel Receiving lter ADC

rE {k}
rE (t)
s(t)
r(t)
d{k} b{k} b(t)
hT (t)
hR (t)
Detector
d{k}
kT
Figure C.46: Block diagram of Data transmission system
(b) see Fig. C.46

(c) As r(t) = s(t) is valid, it is possible also to write
rE (t) = sE (t) = r(t) hR (t) = s(t) hR (t)

7
b{k} T hT (t kT ) hR (t)
=
=
(C.370)
(C.371)
k=0
7

b{k} T h(t kT )
(C.372)
k=0
with
h(t) = hT (t) hR (t)

t
A0
triang
=
T
T
(C.373)
(C.374)
gives
rE (t) = sE (t) = A0
7

b{k} triang(t/T k)
(C.375)
k=0
See sketch in Fig. C.47

rE (t) = sE (t)
A0
{0}
2T
{1}
{1}
{0}
{1}
{0}
{0}
{1}
2T
3T
4T
5T
6T
7T
A0
8T
9T t
Figure C.47: Filtered receipt signal rE (t) = sE (t) = s(t) hR (t)
(d)
PsE
N 1
1 2
=
s {k}
N k=0 E
(C.376)
= A20
(C.377)
C.2
143
Solutions
DACTransmitting lterChannel Receiving lter ADC

rE {k}
rE (t)
s(t)
r(t)
d{k} b{k} b(t)
hT (t)
+
hR (t)
Detector
d{k}
kT
n(t)
Figure C.48: Block diagram of Data transmission system
(e) see Fig. C.48

(f)
SnE ,nE (f ) = |HR (f )|2 Sn,n (f )
= |F {hR (t)} |2 N0 /2

2
N0

t
1

= F
rect
2
T
T
N0 2
si (f T )
=
2
(g)

PnE =
(C.378)
(C.379)
(C.380)
(C.381)
SnE ,nE (f ) d f

N0 2
si (f T ) d f
=
2
(C.382)
(C.383)
by substituting a = T and with the help of eqn. (C.83)

PnE =
N0
N0
=
2 T
2T
(C.384)
(h)
Puseful-signal
Ps
dB = 10 lg E dB
Pnoise
PnE
2
2T A0
= 10 lg
dB
N0
SN R = 10 lg
(C.385)
(C.386)
25. (a) see Fig. C.49

(b) see Fig. C.49
(c) In BPSK the imaginary axis is the separation function between the two decision ares
d{k}
= d0 and d{k}
= d1 , only the sign of real part of the receipt value rE (kTs ) =
Re {rE (kTs )} is needed for the decision. The transmitter phase diagram is rotated by the
non-ideal phase synchronization in reference to that of the receiver. Thus the resulting
amplitude of the information signal is reduced with unchanged noise amplitude at the
receiver input (see Fig. C.50).

(C.387)
s0 = Re s0 ej = s0 cos
4
144
C EXCERCISES
Im {rE }
d{k}
= d1
Im {rE }
d{k}
= d0
s1
s0
d{k}
= d1
d{k}
= d0
s
1
Re {rE }
frE |d1 (rE |d1 ) frE |d0 (rE |d0 )
s0 Re {rE }
frE |d1 (rE |d1 ) frE |d0 (rE |d0 )

1
2
P (d{k}
= d0 |d{k} = d1 )
s1
s0
rE
s1
s0
rE
Figure C.49: Phase diagram at the detector

input (above) and amplitude density function

frE (rE ) with rE = Re rE (below) for ideal noise free transmission (left) and
for additive noise (right)
1
= s0
2
1
s1 = s1
2
(C.388)
(C.389)
This corresponds to a decrease of the signal-to-noise ratio by factor 2

Im {rE }

s1
s1
s0
s0 Re {rE }
Figure C.50: Phase diagram with a phase oset =

C.2
145
Solutions
s0 2
s20
=
2
2 2
SN R
=
2
SN R =
(C.390)
(C.391)
or about 3 dB reduction in logarithmic scale (see Fig. C.51).

SN R = SN R 10 lg 2 dB SN R 3 dB
(C.392)
BER
100
101
102
103
104
15
3 dB
10
Figure C.51: BER with a phase oset of =
10
10 lg(SN R) dB
(Ideal curve for = 0 is shown as dotted.)
(d) From (see eqn. (8.48))

1
BER = erfc
2
s
0
2

(C.393)
and
erfc(x) = 1 erf(x)
follows
An MATLAB example

1 1
s0
BER = erf
2 2
2

1
1
BER
s0 = 2 erf
2
2
= 2 0.5 V erf 1 (2(0.5 0.1))

= 0.6408 V = 640.8 mV
20
(C.394)
(C.395)
(C.396)
(C.397)
(C.398)
% Communications: Digital Modulation -- BPSK

% Solution with MATLAB lsg_bpsk.m
%
% Default standard deviation
20
For further studies see Proakis, J. G. and Masoud S.: Contemporary Communications Systems Using MATLAB, Boston: PWS 1998
146
C EXCERCISES
% in V
sigma = 0.5 ;
% Default bit error rate
BER
= 0.1 ;
% inverse error function
s_0
= sqrt(2)*sigma*erfinv(2*(0.5-BER)); % in V
%
% graphical presentation
%
% Default range
r_E
= -2:0.05:2;
% Amplitude density function
f
= 1/(sqrt(2*pi)*sigma)*exp(-(r_E-s_0).^2/(2*sigma^2));
% Draw
plot(r_E,f);
147
D
D.1
Exams
Exam SS 2006
Problem 1: Signal Theory and LTI Systems

Note: Question a) ,b) and c) can be solved independently from each other.
(a) (7 Points) A LTI system is characterized by the following equation
h(t) =
N2

(n + 1) (t nT ) , N1 < N2
n=N1
(a1) What is h(t) and its Fourier transform H(f ) usually refered to?
(a2) Calculate H(f ) !
(a3) Sketch h(t) for N1 = 3 and N2 = 4 !
(a4) The system is excited with an arbitrary signal s(t). Calculate the output signal y(t)
depending on s(t) !
(a5) Now N1 = 0 and N2 = 1. Sketch the output signal y(t) for
1
rect( 2Tt ) !
s(t) = 2T
(Note: It is helpful sketching s(t) rst.)
(b) (3 points) The gure below shows the block diagram of a radio transmission chain.
PN1 = 0.1 pW
LA
Sender
LT
LR
PT
PR
SNR1
PN 2 = 90 pW
SNR2
Demodulation
The channel attenuation LA is assumed to

LA = 100dB + 20 log10 (d/[km])dB
where d stands for the distance between transmitter and receiver.
(b1) Calculate the receive power level LR in dBm with d = 10km and a transmit power
PT = 1 W?
(b2) How large can be the distance d chosen maximally, when the transmit power PT is
degraded to 10mW and a receive power level LR of 90dBm is needed?
(c) (5 points) The following considerations on the SNR are related to the same transmission
chain as used in part b).
148
EXAMS
(c1) Let the SNR in front of the amplier be SN R1 = 20 dB. Calculate the needed amplication v (in dB), to ensure an SNR before the demodulation of at least SN R2 10 dB.
(c2) The SNR before the amplier(SN R1 ) is now 20 dB. Calculate the theoretically maximum value of SN R2 after the amplier! Give reasons for your answer! (Note: The
amplier can be adjusted between 100 dB and +100 dB.)
Notes:
log(a b) = log(a) + log(b) ; log(a/b) = log(a) log(b); log(ab ) = b log(a)
x
log10 (x)
0.1 0.25 1 2
5 10 100 1000
1 0.6 0 0.3 0.7 1
2
3
Question 2: Analog-digital conversion

(a) (5 Punkte) Consider the following spectrum S(f ) of a real-valued, analog signal s(t):
Lowpass
This signal is now sampled with sampling frequency fs = 6 kHz and then reconstructed with
a low pass lter.
(a1) Draw the spectrum Sa (f ) of the signal after sampling in the range 15 kHz f
15 kHz! Note: Make sure you label the axes of the diagram correctly!
Now assume that the signal s(t) is disturbed through a real-valued, analog 11 kHz cosine
signal.
(a2) Why is it now impossible to reconstruct the original signal s(t) from sa (t)?
(a3) Draw the spectrum of the signal s(t) after the low pass lter in the range 3 kHz
f 3 kHz.
(a4) Due to technical reasons, the sampling frequency can be chosen within the range
4 kHz fs 10 kHz. How should fs be chosen, so that s(t) yields a perfect reconstruction of s(t)?
(a5) How should the above sampling circuit be extended, to allow a perfect reconstruction
of s(t) out of sa (t) while keeping the sampling frequency xed to fs = 6 kHz?
Now the time-discrete signal x[k] is supposed to be quantized with 3 bit. For this, a linear mid-rise
quantizer is used. The input range of the quantizer is dened as 4s xin 4s, where s is
the step size of the quantizer.
(b) (5 Punkte)
149
D.1 Exam SS 2006
(b1) Draw the output signal xout as a function of the input signal xin of the quantizer! Make
sure the axes of the diagram are labelled correctly.
(b2) The SNR shall now be increased by 6 dB. The following parameters can be adjusted:
- Number of quantization bits b.
- Sampling frequency fs .
- Cut-o frequency of the reconstruction low pass fr .
Decide for each stated parameter, whether it is suitable for the desired SNR improvement, and to which value the parameter would have to be adjusted! Please justify your
answer.
Question 3: Analog modulation

(a) (5 points) A signal corresponding to
x(t) = cos(2f1 t) +
1
cos(2f2 t)
2
with f2 = 2f1 , is transmitted via double-side band amplitude modulation (DSB-AM)

s(t) = (A + x(t)) cos(2fc t)
where A = 4 and fc = 10f1 .
(a1) Calculate the modulation index of the above modulation system.
(a2) Draw the spectrum S(f ) of s(t).
An envelope demodulator (see diagram below) is the simplest way to demodulate a doubleside band amplitude modulated signal
s(t)
x(t)
Rectifier
Lowpass
Highpass
(a3) Within which range should the modulation index be chosen, to enable a correct
envelope demodulation?
(a4) Explain the function of the high pass lter in the envelope demodulator!
150
x1(t)
Lowpass
|f |< fg
x(t)
2cos(2fgt)
EXAMS
2cos(2fct)
s(t)
x2(t) Lowpass
|f |< fg
2sin(2fgt)
fc>> fg
2sin(2fct)
X(f)
S(f)
2
1
-fg
fg
-fc -fc+fg
fc -fg fc
(b) (6 points) The following gure shows a transmit system for single-side band amplitude
modulation (SSB-AM) with a suppressed carrier. Advantages of single-side band modulation
compared to double-side band modulation are that only half of the transmission bandwidth
is required, and that the suppression of the carrier leads to a more ecient system. Below,
the frequency domain representations of the input signal, X(f ), and the output signal, S(f ),
are given.
(b1) Draw a block diagram of a suitable receiver structure (including quantities such as
frequencies etc.), which can obtain x(t) out of s(t) again.
(b2) Draw the spectra X1 (f ) and
X2 (f )
j
of the signals x1 (t) and x2 (t).
Note: cos(2fc t) 21 [(f fc ) + (f + fc )]
sin(2fc t) j
[(f fc ) (f + fc )]
2
Question 4: Digital modulation

(a) (6 points) Consider the time-domain representation of the following two digitally modulated
signals x1 (t) and x2 (t). In both cases, all possible symbols are shown.
151
D.1 Exam SS 2006

2
1
x2(t)
x1(t)
0
1
5
0
10
20
30
40 50
t in ms
60
70
80
10
20
30
40 50
t in ms
60
70
80
(a1) Which digital modulation schemes were used?

(a2) Draw the constellation diagram (phase diagram) of the modulation scheme 4-ASK.
(a3) Which data rate (in bit/s) can be achieved if 4-ASK with a symbol length of 0.5ms is
employed?
(a4) Which modulation scheme could be used to achieve twice the data rate of 4-ASK while
maintaining the same symbol length?
(a5) We assume that the transmitted signal is disturbed through additive white Gaussian
noise, so that some symbols will be detected wrongly. Find a suitable mapping of data
bits to the constellation points in the 4-ASK constellation diagram that minimizes the
bit error rate.
(b) (8 points) We now observe a transmitter that uses the modulation scheme 4-QAM. Groups
of two data bits b(2k) {0, 1}, b(2k + 1) {0, 1} are mapped onto time-discrete symbols
sI (k), sQ (k) and then transformed into time-continuous information signals mI (t), mQ (t).
Transmitter
b(2k)
0
1
cos(2fct)
2 sI(k)
2
mI(t)
1
Ts
xI(t)
Ts
-sin(2fct)
b(2k+1)
0
1
2 sQ(k)
2
Mapper
mQ(t)
1
Ts
xQ(t)
n(t)
x(t)
y(t)
Channel
Ts
Puls shaper Upconversion
(b1) Draw a brief sketch of a suitable receiver for this modulation scheme and name the
chosen components! Assume here that there is no phase shift between the oscillators of
the transmitter and receiver.
QAM-systems are often described by their equivalent, complex-valued baseband signals
s(k) = sI (k) + j sQ (k). We assume here that we have the signal
r(k) = s(k) + n(k)
at the input of the detector, where n(k) is complex Gaussian noise with one-dimensional
variance 2 .
152
EXAMS
(b2) Draw the constellation diagram (phase diagram) of the input signal r(k) of the detector
for 2 = 0 with the corresponding bit values b(2k), b(2k + 1). Add suitable decision
borders such that the bit error rate is minimized, assuming that the data bits are
equally probable, i.e. l : P (b(l) = 0) = P (b(l) = 1) = 12 .
The bit error rate (BER) of such a 4-QAM transmission system can be calculated through
the following equation, where Es = E{|s(k)|2 } is the average signal power of the transmitted
symbols s(k), and the complementary error function erfc(x) is given in the following gure.
1
BER = erfc
2
%
Es
4 2
Complementary error function
10
erfc(x)
10
10
10
0.5
1
x
1.5
(b3) Calculate the average signal power Es of the symbols s(k).

(b4) Calculate the bit error rate (BER) of the 4-QAM-system for 2 = 1.
(b5) The symbol error rate (SER) of a 4-QAM-system corresponds to the probability that
one of the two bits (or both) has been detected wrongly. Derive an equation that allows
to calculate the symbol error rate for a given bit error rate, and then calculate the
symbol error rate for BER = 0.1.
153
D.2 Solutions: Exam SS 2006
D.2
Solutions: Exam SS 2006
Question 1: Signal theory and LTI Systems

(a) (a1)
(1 point)
h(t) Impulsantwort (impulse response)
H(f ) Ubertragungsfunktion
(transfer function)
(a2)
(1 point)
F{(t nT )} = ej2f nT
H(f ) = F{h(t)} =
N2

(n + 1)F{(t nT )} =
n=N1
N2

(n + 1)ej2f nT
n=N1
(a3) See gure below

qualitatively correct:
quantitatively correct:
(1 point)
(1 point)
h(t)
5
4
3
2
-3
-2
-1
1
t/T
-1
-2
-3
(a4)
(1 point)

y(t) =
N2

n=N1
N2

s(t )
N2

(n + 1)( nT )d
n=N1
(n + 1)
s(t )( nT )d
(n + 1)s(t nT )
n=N1
Use shifting property of the Dirac impulse

(a5) y(t) is obtained through the superposition of two scaled and time-shifted rectangular
impulses of width 2T and height T1
154
EXAMS
y(t)
3
2T
1
T
1
2T
-T
2T
qualitatively correct:
quantitatively correct:
(1 point)
(1 point)
(b) (b1)
(1 point)
L R = LT L A
PT
d
)dBm 100dB 20log10 (
)dB
mW
km
= 30dBm 100dB 20dB = 90dBm
LR = 10log10 (
LR
(b2)
PT
d
)dBm 100dB 20log10 (
)dB
mW
km
d
90dBm = 10log10 (10)dBm 100dB 20log10 (
)dB
km
d
)dB
90dBm = 10dBm 100dB 20log10 (
km
d
)dB = 0dB
20log10 (
km
d = 100 km = 1km
LR = 90dBm = 10log10 (
approach correct
result correct
if an alternative approach is chosen, points are distributed accordingly
(c) (c1)
(1 point)
(1 point)
(3 points)
SN R1 = 20dB = 10log10(
PR
)dB
PN 1
PR = 100PN 1
SN R2 = 10dB = 10log10 (
vlinear PR
vlinear PN 1 + PN 2
10PN 2
10PN 2
=
=
PR 10PN 1
100PN 1 10PN 1
10 =
vlinear
vlinear PR
)dB
vlinear PN 1 + PN 2
155
10 90 1012 W
= 100
90 1013 W
= 10log10 (vlinear )dB = 20dB
vlinear =
vdB
1 point for the correct calculation of the noise before demodulation: PN = vlinear PN 1 +
PN 2
1 point for the correct calculation of PR = 100PN 1 = 10pW
1 point for the result vdB = 20dB
if another approach is chosen or the noise term is not mentioned explicitly, but the
result is correct, both points are obtained
(c2)
(2 point)
SN R2 = 10log10 (
PR
vlinear PR
)dB
)dB = 10log10 (
vlinear PN 1 + PN 2
PN 1 + v PN 2
linear
SN R2 is maximized, if the denominator is minimized; the denominator is minimized,

if the amplication is maximized
vdB = +100dB
PN 2
0
vlinear
PR
= 10log10 (
) = SN R1 = 20dB
PN 1
vlinear = 1010
SN R2,max
1 point for result and 1 point for calculation or explanation

Question 2: Analog-digital conversion
(a) (a1) Diagram qualitatively and quantitatively correct
(1 point)
(a2) Due to the sampling we observe the eect of aliasing.
(1 point)
(a3) Diagramm qualitatively and quantitatively correct
(1 point)
156
(a4) The sampling frequency must be chosen in the range 7 kHz fs 8 kHz.
Any choice within this range valid.
EXAMS
(1 point)
(a5) Anti-aliasing low pass before sampling with the cut-o frequency 2 kHz < |fg |
point)
(b) (b1) See gure
(1
(1 point qualitative, 1 point quantitative)

7
2
5
2
3
xout
s
2
s
xin
2
4 s 3s 2 s
fs
2
2 s
3s
4 s
s
2
5
s
2
7
s
2
(b2) Following parameters are suitable for increasing the SN R by 6 dB:

- Increase number of bits b by 1
See calculation of the SINR as a function of the number or quantization bits b, as
derived in lecture.
(1 point)
- Increase sampling frequency fs by a factor of 4
The quantization noise power remains constant and can be assumed as distributed
over the frequency range f2s < f f2s . Thus, increasing fs by a factor of 4
reduces the quantization noise density by a factor of 4, corresponding to a 6dB
SNR improvement.
(1 point)
- The cut-o frequency of the reconstruction low pass is not a suitable parameter,
as the in-band noise power cannot be reduced by ltering, without also aecting
the desired signal.
(1 point)
Question 3: Analog modulation
(a) (a1) Maximum value of |x(t)|:
1
x(t) = cos(2f1 t) + 2 cos2 (2f1 t) 1
2
*2
)
1
3
= cos(2f1 t) +
2
4
where 1 cos(2f1 t) 1 max[|x(t)|] =
3
2
157
Modulation index :
(1 point)
max[|x(t)|]
A
3
=
8
(a2) Spectrum according to following gure.

qualitatively correct
quantitatively correct
(1 point)
(1 point)
S(f)
1
1/2
1/4
-12f1 -11f1 -10f1 -9f1 -8f1
8f1
9f1 10f1 11f1 12f1
(a3) (0 <) < 1
(1 point)
(a4) The high pass lter removes the DC component from the carrier signal.
(1 point)
(b) (b1) Receiver according to following diagram.

qualitatively correct: Down converter and low pass lter
Lowpass
|f |< fg
s(t)
(each 1 point)
(1 point)
x(t)
cos(2(fc-fg)t)
(b2) Spectra according to following diagram.

qualitatively correct
(each 1 point)
(1 point)
158
X1(f)
X2(f) / j
1
-fc -fg
EXAMS
-fc -fc +fg
fc -fg fc fc +fg
fg
-2fg
-fg
2fg
-1
Problem 4: Digital Modulation

(a) (a1) x1 (t): 8-ASK (1 point), x2 (t): 2-FSK (1 point)
(2 points)
(a2) 4-ASK has 4 symbol amplitudes either on the I branch or on the Q branch possible,
e.g.
Q
I
(1 Punkt)
(a3) 4-ASK allows 2bit/symbol, i.e. R = 2 2000 = 4000bit/s
(1 point)
(a4) correct are 16-QAM, 16-ASK, 16-PSK, 16-FSK etc. (1 answer is enough)
(1 point)
(a5) neighbouring symbols only divers in 1 bit, e.g.

00
01
Q
11
10
I
(1 point)
(b) (b1) sketch should be similar to the gure below:

Empfnger
2cos(2fct)
Ts
n(t)
y(t)
x(t)
Kanal
kTs
^
b(l)
kTs
^b(l+1)
( )dt
0
-2sin(2fct)
Ts
( )dt
0
Integrierer
Abwrtsmischer
Komponenten vorhanden
order and names correct
Abtaster
Entscheider
(1 point)
(1 point)
159
(b2) constellation diagram:

01
11
2
I
Entscheidungsgrenzen
00
10
constellation correct:
decision borders correct:
(1 point)
(1 point)
(b3) Es = ( 2)2 + ( 2)2 = 4
(1 point)
(b4) BER = 21 erf c(1) 0.07 (A range of 0.06 ... 0.08 is accepted)
(1 point)
(b5) Ps = 1 (1 Pb )2
Ps = 1 (0.9)2 = 0.19 (A range of 0.18 ... 0.2 is accepted)
(1 point)
(1 point)
160
D.3
EXAMS
Exam WS 2006/2007

Note: Question a), b) and c) can be solved independently from each other. Please note the hints
at the end of this problem on the next page (back side).
(a) (3 Points) A System is characterized by the following relation between the input signal x(t)
and output signal y(t):
y(t) = [x(t) + x(t + 2)]2 + 3
(a1) Is the system linear?
(a2) Is the system time invariant?
(a3) The system is not causal. How could the above relationship be modied in order to
ensure causality?
(b) (4 Points) The gure below shows the block diagram of a radio transmission link between a
mobile and a base station.
PN1 = 1013 W
LA
Mobile
LT
LR
PT
PR
Demodulation
The channel attenuation LA (in dB) is assumed to be given by

LA = 100dB + 10 log10 (d/km)dB
,
where d stands for the distance between transmitter and receiver and denotes the pathloss
coecient, which depends strongly on the characteristics of the environment.
(b1) The mobile typically has a transmit power of 1 W. Calculate the received power in W
at a base station located at a distance of 3 km! ( = 4)
(b2) Calculate the signal-to-noise ratio (SNR) at the input of the amplier!
(b3) To ensure reliable transmission, the SNR at the output of the amplier must be SN R
0 dB. Calculate the maximum allowable distance between the mobile and the base
station!
Note: The pathloss coecient is now = 3!
(c) (8 Points) A lter is characterized by the following impulse response:
h(t) = a1 (t) + a2 (t T )
161
D.3 Exam WS 2006/2007
(c1) Draw h(t) for a1 = 1 and a2 = 0.5 !

Note: Make sure you label the axes of the diagram correctly!
(c2) Calculate |H(f )| and draw it in the range T1 < f < T1 for a1 = a2 = 1 !
1
1
< f < 2T
are of interest. What is the char(c3) Now, only the frequencies between 2T
acteristic of the lter in this frequency range (high pass, low pass, band pass, band
stop)?
(c4) The lter is excited with one half-wave of the sine function:
sin( t ) if 0 < t < 2T

2T
x(t) =
0
else
Draw x(t) !
(c5) Sketch the output signal for a1 = 1 und a2 = 0.5 !
Hints:
cos2 x = 12 (1 + cos(2x)) ;
log(a b) = log(a) + log(b) ; log(a/b) = log(a) log(b);
log(ab ) = b log(a)
x
log10 (x)
0.1 0.25 1 2
3
5 10 100 1000
1 0.6 0 0.3 0.5 0.7 1
2
3
Problem 2: Analog-Digital Conversion

(a) (5 Points) The real-valued, analog signal s(t) with the spectrum S(f ) is sampled and afterwards reconstructed.
S(f)
- fg
fg
(a1) Sketch the arrangement needed for the distortion-free sampling and reconstruction and
name all components!
(a2) What is the minimal sampling frequency fs,min which ensures a distortion-free reconstruction of the signal s(t)?
(a3) Which eect arises when the sampling frequency is chosen to be fs < fs,min , and hence
a distortion-free reconstruction of s(t) is not possible any more?
(a4) Sketch the spectrum of the signal s(t), sampled with a sampling frequency fs < fs,min !
Mark the eect described in question (a3) in the diagram!
162
EXAMS
xout
3
s
2
s
2
2 s
2s
xin
s
2
a) Characteristics of the 4 step

mid-rise quantizer
b) probability density function

of the amplitude of xin (k)
(b) (5 Punkte) The discrete-time signal xin [k] is now quantized by a linear, 4 step mid-rise
quantizer with the following characteristics.
(b1) What is the signal-to-noise ratio (SNR) at the output of the quantizer, when the amplitude of the input signal xin [k] is uniformly distributed with xmax = 2s (see gure
above)!
Note: Equation is sucient!
(b2) Give one advantage and one disadvantage of the linear mid-rise quantizer!
Now xin [k] is still uniformly distributed between [xmax , xmax ] (see gure above), but now
xmax s.
(b3) Calculate the signal power Pout = E{x2out [k]} of the signal xout [k] at the output of the
quantizer, where E{.} denotes the statistical expectation!
Note: Draw rst the probability density function of the amplitude of the output signal
xout [k]!
Problem 3: Analog Modulation
(a) (6 Points) A transmitter based on amplitude modulation is transmitting the band pass signal
sBP (t) = (1 + x(t)) cos(2fc t) ,
where x(t) is the actual, real-valued information signal to be transmitted (with the property
t, |x(t)| < 1) and fc is the carrier frequency. The spectrum of x(t) is shown in the following
plot:
1
-fg
|X(f)|
fg < f c
fg
163
D.3 Exam WS 2006/2007
(a1) Derive an equation for the spectrum SBP (f ) of the generated band pass signal, and
plot the absolute value of this spectrum! Make sure the axes are labelled correctly!
(a2) Now draw a block diagram of an envelope demodulator, which is able to obtain the
signal x(t) from the signal sBP (t)! Please label all components and briey state their
function!
(a3) In our example, a DC component is added to the information signal before upconversion.
Why is this necessary or helpful, if
an envelope demodulator
a synchronous demodulator
is used, respectively?
(b) (5 Points) We assume that the signal sBP (t) from question (a) is transmitted without any attenuation, and the receiver uses the following synchronous demodulator. As you can see, the
received signal is downconverted with g(t) = cos(2fc t+), i.e. the oscillators of transmitter
and receiver have a carrier phase oset of .
sBP(t)
x(t)
Ideale Band pass
(surpression of
adjacent channels)
Ideale low pass

|f| fg
g(t)=cos(2 fc t+ )
Ideale high pass

|f| > 0
(b1) Derive the signal x(t) at the output of the synchronous demodulator! Here, the relation
cos(a)cos(b) = 1/2(cos(a b) + cos(a + b)) might be useful.
(b2) A measure for the quality of transmission is the signal-to-noise-ratio (SNR) after transmission and demodulation. We assume that the signal x(t) has an average power of
Es = E{|x(t)|2 }, we have a perfect transmission without signal attenuation, and measure an additive white noise with power En at the output of the demodulator. Derive
the SNR at the output of the demodulator as a function of !
(b3) We now assume that the synchronous demodulator downconverts the received signal
with g(t) = cos(2(fc + f )t), i.e. transmitter and receiver use slightly dierent carrier
frequencies. What is the consequence for the signal x(t)? (Short comments sucient;
please name at least two consequences)
(a) (5 Points) A digital transmission system has a symbol duration TS = 1 ms, where the data
rate is 3000 bits/s.
(a1) How many bits are represented by one symbol?
(a2) How many symbols contains the symbol alphabet? Give a modulation scheme which
fulllls this requirement!
164
EXAMS
(a3) Now the data rate shall be doubled while keeping the symbol duration constant. How
many bits must be represented by one symbol now? Give an appropriate modulation
scheme!
(b) (7 Points)
(b1) Sketch the constellation diagram of the modulation scheme 8-PSK!
(b2) Write in the diagram of question (b1) a possible mapping of the bits to the constellation
points (symbols)!
Now we consider the modulation scheme 4-PSK. To enable the use of a given receiver, the
real axis should be the decision threshold forthe rst bit and the imaginary axis for the
second bit. The symbols have a magnitude of 2.
(b3) Draw a constellation diagram which fulllls these requirements! Write in the mapping
of the bit pairs to the symbols.
If the transmission is corrupted by additive Gaussian noise with noise power 2 2 , the bit
error rate Pb (BER) in such a QPSK transmission system can be calculated by the following
equation:
1
Pb = erfc
2
1
2 2

.
The complementary Gaussian error function erfc(x) is depicted in the gure below.
0
10
10
10
10
10
erfc(x)
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
165
D.3 Exam WS 2006/2007
(b3) Calculate Pb for 2 = 1/8 !

(b4) Calculate 2 for Pb = 1/4! Give also the derivations!
(c) (2 Points) The following three constellation diagrams of binary digital modulations are given:
1)
2)
3)
-1
(c1) Which contellation is best suitable for the data transmission in presence of additive
gaussian noise, when the bit error rate shall be kept as low as possible?
Decide for exactly one answer, and give reasons for your decision! Apply additional
criteria if necessary.
166
D.4
EXAMS
Solutions: Exam WS 2006/2007
Problem 1: Signal Theory und LTI Systeme

(a) (a1)
(1 Point)
nonlinear
(a2)
(1 Point)
time invariant
(a3)
(1 Point)
function could be shifted by 2 in positive temporal direction:
y(t) = [x(t 2) + x(t)]2 + 3
(b) (b1)
(1 Point)
PT
dBm = 30dBm
mW
= LT LA = 30dBm 100dB 40log10 3dB
= 30dBm 100dB 20dB = 90dBm
= 109 mW = 1012 W
LT = 10log10
LR
LR
PR
(b2)
(1 Point)
PN
dBm = 100dBm
mW
SN R = LR LN = 90dBm (100dBm)
SN R = 10dB
LN = 10log10
(b3)
(2 Point)
The SNR is not inuenced by the amplier: SN Rin = SN Rout
SN R = LR LN = 0dB
LR = LN = 100dBm
LR = 100dBm = LT LA = 30dBm 100dB 30log10
30dBm = 30log10
d
dB
km
d
dB
km
d = 10km
1 Point if SN Rin = SN Rout
1 Point for correct distance d
(c) (c1)
(1 Point)
167
D.4 Solutions: Exam WS 2006/2007
h(t)
0.5
-1
t/T
qualitative and quantitative correct: 1 Point

(c2)
(3 Point)
rst H(f ) and |H(f )| must be calculated:
H(f ) = 1 + ej2f T

|H(f )| =
(1 + cos(2f T ))2 + sin2 (2f T )

|H(f )| =
1 + 2cos(2f T ) + cos2 (2f T ) + sin2 (2f T )

|H(f )| =
2 + 2cos(2f T )
%
1
4( (1 + cos(2f T )))
|H(f )| =
2
|H(f )| = 2|cos(f T )|
|H(f)|
-1
T
-1
2T
1
2T
1
T
1 Point for |H(f )|

1 Point: Sketch qualitatively correct
1 Point: Sketch quantitatively correct
(c3)
(1 Point)
Low pass
(c4)
(1 Point)
168
EXAMS
x(t)
1/2
-1
(c5)
t/T
(2 Point)
The output signal is the superposition of 2 sine half-waves, where the second half-wave
is shifted by T and scaled by 1/2.
y(t)
1/2
-1
t/T
2 half-waves correct: 1 Point

superposition correct: 1 Point
(a) (a1) Arrangement (with or w/o anti-aliasing lter) correct
Names of components correct
LP
AntiAliasing
Filter
(1 Point)
(1 Point)
LP
Sampling
Reconstruction
low pass
(a2) fs,min > 2 fg
(1 Point)
(a3) Aliasing
(1 Point)
(a4) Diagram qualitatively correct and spectral overlapping due to aliasing correctly marked
(1 Point)
169

Aliasing
Aliasing
Sa(f)
...
...
- fg
- fs
- 2f s
fg
2fs f
fs
(b) (b1) Number of quantisation bits b = 2

SN R 6 dB b = 12 dB
(1 Point)
(b2) Advantage: symmetric characteristic

Disadvantage: No quantization step for xout [k] = 0
Noise at output of quantizer even if xin [k] = 0
(1 Point)
(1 Point)
(b3) Amplitude distribution of xout [k] see gure

2 1 s 2 s2
Pout = 12 s
+2 2
= 4
2
(1 Point)
(1 Point)

(a) (a1) SBP (f ) = 12 (f fc ) + 12 (f + fc ) + 2 X(f fc ) + 2 X(f + fc )
0.5
0.5
-fc-fg
-fc
-fc+fg
|SBP (f)|
fc-fg
fc
Equation correct
Sketch and descriptions correct
fc+fg f
(1 Point)
(1 Point)
(a2) Magnitude demodulator should be sketched as follows:
Band pass
Rectifier
(Surpression of (Reconstruction of
adjacent channels)
envelope)
Low pass
(Cancellation
of carrier)
High pass
(Cancellation of
DC offset)
constellation correct (bandpass optional):

Description and explanation of components correct:
(1 Point)
(1 Point)
(a3) The DC oset servers in the magnitude demodulator to avoid phase leaps in the band
pass signal due to zero-crossings of the base band signal.
(1 Point)
170
EXAMS
The DC oset serves in synchron demodulators for detection of the carrier signal.
(1 Point)
(b) (b1) The signal at the output of the demodulator can be calculated by:
x(t) = (1 + x(t)) cos(2fc t) cos(2fc t + )|0<|f |fg

= 12 (1 + x(t)) (cos(4fc t + ) + cos)0<|f |fg

= 12 (1 + x(t)) cos0<|f |
=
cos
x(t)
2
Derivation correct:
Result correct:
(b2) SNR =
E{|
x|2 }
En
(1 Point)
(1 Point)
2 cos2 Es
4En
(1 Point)
(b3) Each correct answer one point, maximum 2 points total:
(max. 2 Points)
The spectrum of the information signal x(t) is shifted in comparison to x(t).

The DC oset in the base band signal can not be surpressed.
High frequency components of the information signal are surpressed partly by ltering with |f | fg .
(a) (a1) 2 Bits/Symbol
(1 Point)
(a2) 22 = 4 Symbols
Example: 4-PSK
(1 Point)
(1 Point)
(a3) 4 Bits/Symbol
Example: 16-FSK
(1 Point)
(1 Point)
(b) (b1) Sketch can be like this:

Q
171
Qualitatively correct
(1 Point)
(b2) Every mapping of all possible binary triples is correct.
(1 Point)
(b3) Drawing can be like this:

Q
10
11
1
-1
-1
00
01
Qualitatively correct
Quantitatively correct
Bit mapping correct
(1 Point)
(1 Point)
(1 Point)
(b4) Pb 2,3 103 , are considered as correct 2 103 Pb 3 103
(1 Point)
(b5) For Pb = 1/4:

1
2
1
2
4
2
Derivation and solution:

= erfc

2 2
2 2
2 2
2
(1 Point)
(c) Two answers are possible:

The constellation 1) is best suited, because it has the symbol distance 2 and a lower mean
signal power than constellation 3).
The constellation 3) is best suited, because it has the symbol distance 2 and it can be used
with simpler transmitter and receiver circuits 3).
Solution:
Explanation:
(1 Point)
(1 Point)
172
D.5
EXAMS
Exam SS 2007

Note: Questions a), b) and c) can be solved independently from each other. Please note the hints
at the end of this problem on the next page (back side).
(a) (2 Points) A system is characterized by the following relation between input signal x(t) and
output signal y(t):
y(t) = |x(t + 1)|.
Which of the following properties has the system:
linear,
causal,
time invariant?
(b) (6 Points) The gure below shows the impulse response h(t) of a LTI-system.
h(t)
-2
-1
(b1) Is the system causal?

(b2) Express h(t) using the triangular function triang(t)!
(b3) Determine the transfer function H(f ) of this LTI-system!
(b4) The system is excited by the signal x(t) = (t) + (t 3). Draw the output signal of
the LTI system! Make sure you label the axes of the diagram correctly!

Notes: triang(t) =
1 |x| f
ur 1 < x < 1
0
f
ur
sonst
F{triang(t)} = si2 (f )
(c) (7 Points) The signal arriving at a radio receiver with the receiving level LR = 60dBm
has to be amplied before demodulation. Considering the 2 available ampliers, which of
them are connected in an amplier chain? For each amplication a certain amount of noise
is added to the signal by the amplier itself. The resulting block diagram can be found in
the gure below.
173
D.5 Exam SS 2007
PN ,LNA = 9 * 10 10 W
LR = 60dBm
PN ,PA = 10 7 W
v LNA
v PA
v LNA = 10dB
v PA = 30dB
Low Noise Amplifier
Power Amplifier
PP
LP
Demodulation
The 2 ampliers dierentiate showed in the picture in their gain and in the noise power
they cause. One of them is a low noise amplier (LNA) with a low noise power of PN,LN A =
9 1010 W and a low gain of vLN A = 10dB. The other one is a power amplier (PA) with a
strong noise power of PN,P A = 107 W and high gain of vP A = 30dB.
(c1) How large is the signal-to-noise ratio (SNR) before the power amplier?
(c2) The received signal must have a power of at least PP = 0.1mW before demodulation.
Is this with the given ampliers possible? If not, to which value has the gain of the
power amplier vP A (in dB) to be changed?
(c3) How large is the SNR before demodulation for a power amplier gain of vP A = 30dB?
(c4) The SNR before demodulation has to become maximum. For this reason the order of
the ampliers in the amplier chain can be exchanged. Which oder is the best one?
Give Reasons! For this, look at the SNR before demdodulation in case that the power
amplier is located in front of the low noise amplier in the amplier chain!
Notes:
log(ab ) = b log(a)
x
log10 (x)
0.1 0.25 1 2
3
5 10 100 1000
1 0.6 0 0.3 0.48 0.7 1
2
3

(a) (2 Points) Digital signals can be attained from analog signals by means of analog-digitalconversion. Specify the dierence between digital signals and analog signals!
(b) (4 Points) Consider the analog signal s(t) with the frequency spectrum S(f ) shown in gure
D.1. During an analog-digital conversion s(t) is perfectly sampled with sampling frequency
fA = 25 kHz. The signal resulting from this sampling is denoted by sA (t). Assume that there
is
no
anti-aliasing
lter employed.
(b3) Draw the frequency spectrum SA (f ) of the signal sA (t) in the range of
60 kHz f 60 kHz! Pay attention to correct axis labeling!
(b4) Can the analog signal s(t) be recovered from the signal sA (t) without any error? Give
reasons and refer to the given sampling frequency fA !
174

Figure D.1: Frequency spectrum S(f )

EXAMS

Figure D.2: Frequency spectrum N (f )
(b5) Now, there is an additive noise signal n(t), that interferes the analog signal s(t) before
the sampling. The frequency spectrum N (f ) of the additive noise signal n(t) is shown
in Figure D.2. What is the minimum sampling frequency fA,min that can be allowed, if
we just want to recover the signal s(t) without any error using a low-pass lter after
the sampling?
(c) (4 Points) Consider the quantizer of an analog-digital converter with the quantization characteristic shown in gure D.3. x(t) denotes the input signal and xq (t) denotes the output
signal of the quantizer. The quantization step size is x.
"
# %$
!
'
# %$
.&%$
.%$
' )(
- # %$
&%$ !
, +* %$
Figure D.3: Quantization characteristic
175
D.5 Exam SS 2007
(c1) Is it a mid-rise or mid-tread quantizer? Give reasons!

(c2) Draw the absolute value of the quantization error signal |e(t)| as a function of the input
signal x(t) in the range of 2 x x(t) 2 x! Pay attention to correct axis labeling!
(c3) Assume the input signal of the quantizer to be x(t) = 3 cos(2f t). Furthermore, assume x = 1.5. Determine the signal-to-noise-ratio SNR resulting at the output of the
quantizer in this case!
(a) (7 points) A transmitter which is based on amplitude modulation generates the band pass
signal
sBP (t) = (1 + x(t)) cos(2fc t)
where x(t) is the actual, real-valued information signal to be transmitted, and fc is the
carrier frequency. The frequency spectrum X(f ) of the information signal x(t) is given as
follows:
1
X(f)
fg << fc
fg
-fg
(a1) Compute the frequency spectrum SBP (f ) of the band pass signal and plot this spectrum!
Please make sure that the axes are correctly assigned!
(a2) We now assume that the information signal is x(t) = sin(2fi t). How large can the
factor be maximally chosen, if an envelope demodulator is supposed to be used at
the receiver side? Please state briey, why an appropriate choice of is important in
this case.
(a3) A disadvantage of this kind of amplitude modulation is that only a portion of the
transmitted power is actually connected to the information signal x(t). Please compute
- assuming that x(t) = sin(2fi t) and = 0.5 - the ratio of the power of the scaled
information signal x(t) over the power of the entire band pass signal sBP (t)!
(b) (4 points) We now assume that the signal sBP (t) from problem (a) is transmitted error-free,
and the following demodulator is used at the receiver side:
xI (t )
sBP (t )
Idealer Tiefpa
|f| / fg
cos ( 2 f c t + )
Idealer Hochpa
|f| > 0
Phasenschtzung
Idealer Bandpa
(Nachbarkanalunterdrckung)
xQ (t )
Idealer Tiefpa
|f| / fg
sin ( 2 f c t + )
Idealer Hochpa
|f| > 0
176
EXAMS
(b1) Compute the signals xI (t) and xQ (t) at the output of the demodulator as a function of
(Note: cos(a)cos(b) = 1/2(cos(a b) + cos(a + b)) and cos(a)sin(b) = 1/2(sin(a
b) + sin(a + b))).
(b2) We now assume that can be estimated at the receiver side through . Please
extend the demodulator architecture in such a way that the original information signal
x(t) can be reconstructed from xI (t), xQ (t) and as accurately as possible (Note:
: cos2 + sin2 = 1). Please ensure that all used components and signals are
adequately labelled.
(a) (6 Points)
0.5
BP,2
(t)
sBP,1(t)
The following two diagrams show two digital modulated band-pass signal sBP,1 (t), sBP,2 (t)
in time domain. In both cases, all possible symbols are shown.
4
0
0.5
4
t/T
Figure D.4: sBP,1 (t)
1
0
4
t/T
Figure D.5: sBP,2 (t)
(a1) Which digital modulation schemes were used? Give the names and their order.
(a2) Draw the I/Q diagram (constellation diagram) of the modulation scheme of sBP,1 (t)!
(a3) Sketch the impulse response of the pulse shape lter, used for the modulation of sBP,1 (t)!
(a4) Which data rate R1 (in bit/s) can be achieved with the modulation scheme of sBP,1 (t)
when a symbol length T = 2 ms is employed?
(a5) For doubling the data rate (R3 = 2 R1 ), without changing the modulation scheme, the
symbol length T must be adapted. How T must be changed to achieve the doubled
data rate?
(b) (6 Points)
In gure D.6 and D.7 the block diagram of a QPSK-Modulators is given with the correspond

ing symbol mapping. The impulse former may have the impulse response (t) = rect Tt 12 .
The data stream d[k] = [01, 10, 11, 11] is transmitted.
177
D.5 Exam SS 2007
Symbolmapper
d[k] a[k] = I[k] + j Q[k]

00
1
j
01
11
1
j
10
Figure D.7: Symbolmapper
Figure D.6: QPSK-Modulator
(b1) Draw the signals I(t) and Q(t) in the time range 0 t < 4 T !
Note: Pay attention to complete axes labels!
(b2) Draw the resulting bandpass signal sBP (t) in the time range 0 t < 4 T for the given
data stream, where the carrier frequency is given by fc T = 2!
Note: Pay attention to complete axes labels!
(b3) Draw the I/Q-diagram (constellation diagram) of the used modulation scheme and also
the decision thresholds!
(c) (2 Points)
The bit error rate (BER) of such a QPSK transmission can be computed through the following Equation:
% E
1
s
BER = erfc
2
4 2

where Es = E |S[k]|2 denotes the average signal power of transmitted symbols S[k] and
2 denotes the variance of the channel noise. The complementary error function erfc(x) is
given in the following gure.
(c1) Compute the bit error rate (BER) with an average symbol energy Es = 1 and a noise
1
!
variance 2 = 16
(c2) How will change the bit error rate (BER) of the QPSK modulation scheme if there is
a is a unknown phase oset between the transmit and receive oscillator? Give reasons!
178
10
10
10
10
10
erfc(x)
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
EXAMS
179
D.6

(a) The system is:
non-linear (computation of magnitude is a nonlinear operation)
non-causal (e.g. for computation of y(0), knowledge of x(1) is necessary)
time invariant (time shift of x(t) by results in a time shift of y(t) by , too)
2 properties correct:
all 3 properties correct:
(1 point)
(2 points)
(b) (b1)
(1 point)
non causal (in the negative time h(t) has values which are not zero)
(b2)
(1 point)
Superposition of 3 time shifted triangles or one large triangle minus one small triangle
h(t) = triang(t) + triang(t 1) + triang(t + 1) = 2triang(t/2) triang(t)
(b3) Use of linearity and time shifting of fourier transformation:

H(f ) = si2 (f ) + si2 (f )ej2f + si2 (f )ej2f = si2 (f )(1 + 2cos(2f ))
or
H(f ) = 4si2 (2f ) si2 (f )
The 2 solutions are equivalent as it can be seen below:
sin2 (2f ) sin2 (f )
(2sin(f )cos(f ))2 sin2 (f )
=
4 2 f 2
2f 2
2f 2
sin2 (f )
sin2 (f )
1
2
(4cos
(f
)
1)
=
(4 (1 + cos(2f )) 1)
=
2
2
(f )
(f )
2
2
sin (f )
(2 + 2cos(2f ) 1) = si2 (f )(1 + 2cos(2f ))
=
(f )2
4si2 (2f ) si2 (f ) = 4
H(f ) is the fourier transfomred of h(t)

result correct
(1 point)
(1 point)
(b4) 2 shifted trapezoids superimpose to resulting signal

y(t)
1
-2
-1
Partial trapezoids quantitative and qualitative correct:

Complete signal quantitative and qualitative correct:
(1 point)
(1 point)
180
EXAMS
(c) (c1)
(1 point)
LN,LN A =
LN,LN A =
SN R =
SN R =
9 1010 W
10lg(
)dBm = (10lg(9) + 10lg(107 ))dBm
3
10 W
(9.6 70)dBm = 60.4dBm
LP + vLN A LN,LN A = 60dBm + 10dB (60.4dBm)
10.4dB
(c2)
(2 points)
104
)dBm = LR + vLN A + vP A
103
= LP LR vLN A = 10dBm (60dBm) 10dB = 40dB
LP = 10lg(
vP A
The given ampliers are not enough. The power amplier has to be enhanced to a gain
of 40dB.
(c3)
(2 points)
Power level of received signal in front of demodulation:
Lp = LR + vLN A + vP A = 60 dBm + 10dB + 30dB = 20dBm
Power level
of rst noise
signal nLN A (t) in front of demodulation:

PN,LN A
Ln,1 = 10 lg 1 mW + vP A = (4.8 + 4.8 70)dBm + 30dB = 30.4dBm Signal power
of rst noise signal in front of demodulation:
Pn,1 = 10Ln,1 /10 mW = 9 107 W
Total signal power of the noise signals nLN A (t) and nP A (t) before demodulation:
Pn,ges = Pn,1 + Pn,P A = 106 W
SNR before demodulation:

n,ges
= 10dB
SN R2 = Lp 10 lg P1mW
(c4)
(2 points)
vLN A vP A PR
)dB
vLN A PN,P A + PLN A
105 W
= 10lg( 6
)dB
10 W + 9 1010 W
SN R2 = 10lg(
SN R2
Compared to SN R1 from c3) here the term 9 1010 is added to the denominator.
Thus, the SNR drops under the 10dB from c3). Therefore, it is better to use low noise
amplier rst and then power amplier in the amplier chain.
(a) Analog signals: continous in time and value
Digital signals: discrete in time and value
(1 Point)
(b) (b1) Spectrum SA (f ) as shown in gure below6

Diagram qualitatively correct - curve
Diagram quantitatively correct - axis labeling
(1 Point)
(1 Point)
181

01234
6
5
...
<@; <== <=; <?= <?; <>= <>; <6= <6; <5= <5; <=
...
5; 5= 6; 6= >; >= ?; ?= =; == @; 3 7 89:
(b2) Yes, s(t) can be recovered from sA (t) without any error.
Reasons: The sampling freqeuncy fA = 25 kHz is greater than twice the upper critical
frequency fG,s = 10 kHz of s(t), i.e. fA > 2 fG,s . Consequently, there is no aliasing. (1
Point)
(b3) The frequency spectra S(f ) and N (f ) do not overlap. After sampling the resulting
frequency spectra SA (f ) and NA (f ) must not overlap, too. Aliasing within NA (f ) is
irrelevant (see gure).
ABCDEFGBCDE
I
H
...
OSN OPP OPN ORP ORN OQP OQN OIP OIN OHP OHN OP
...
HN HP IN IP QN QP RN RP PN PP SN D J KLM
If fG,s and fG,n denote the upper critical frequencies of s(t) and n(t), respectively, than:
fA fG,s + fG,n = 10 kHz + 20 kHz
Hence, the minimum sampling frequency that can be allowed is fA,min = 30 kHz.
Point)
(1
(c) (c1) It is a mid-rise quantizer.

Reasons: xq (t) x
(1,
3,
.
.
.)
, i.e. there is no quantization step xq (t) = 0 in the
2
quantization characteristic.
(1 Punkt)
(c2) |e(t)| = |xq (t) x(t)| as a function of x(t) as shown in gure below
(1 Point)
(1 Point)
TUVWXT
]
\[
^
_Z\[
_\[
\[
Z\[ YVWX
(c3) With 2 x = 3 and x(t) = 3 cos(2f t) the full range of 4 quantization steps is used,
which corresponds to a resolution of b = 2 Bits. For a sine signal the signal-to-noise-ratio
at the output of the quantizer is given by:
SNR = (1.76 + 6.02 b/Bits) dB
Subsequently: SNR = 13.8 dB.
(1 Point)
182
EXAMS

0.5
0.5
-fc-fg
-fc
-fc+fg
|SBP (f)|
fc-fg
fc
fc+fg f
Equation correct:
Plot qualitive correct:
Plot quantitive correct:
(a2) t : 1 + x(t) > 0 < max|x(t)| < 1
One of the following aspects is mentioned:
(1 point)
(1 point)
(1 point)
(1 point)
(1 point)
Information signal may only be contained in the envelope of sBP (t)

Phase jumps in sBP (t) must be avoided
(a3) Power of the information signal in the band pass is si,BP (t) = x(t) cos(2fc t):
1
Pi,BP = 2 Px 21 = 0.52 12 12 = 16
Total power of the band pass signal sBP (t):
9
Ps,BP = 2 Px 21 + 12 = 16
Pi,BP
Power ratio: Ps,BP
= 19
Correct approach:
(1 point)
Correct result:
(1 point)
(b) (b1) The signals at the output of the demodulator can be calculated as
xI (t)
= (1 + x(t)) cos(2fc t) cos(2fc t + )|0<|f |fg

= 21 (1 + x(t)) (cos(4fc t + ) + cos)0<|f |fg

= 12 (1 + x(t)) cos0<|f | = cos
x(t)
2
xQ (t)
= (1 + x(t)) cos(2fc t) (-sin(2fc t + ))|0<|f |fg

= 12 (1 + x(t)) (sin(4fc t + ) sin())0<|f |fg

= 12 (1 + x(t)) sin0<|f | = sin
x(t)
2
Correct approach:
Correct result:
(b2) The following solution is possible:
(1 point)
(1 point)
183
x I (t )
x(t )
cos ( )
xQ (t )
sin ( )
Correct principle:
Correct annotation:
(1 point)
(1 point)

(a) (a1) left: 8-ASK
rigth: 4-FSK
(a2) I/Q-Diagram
(1 point)
(1 point)
Q[k]
I[k]
qualitive und quantitive correct
(1 point)
(a3) Impulse response of the impulse former (t) = rect
t
T
1
2

(1 point)
(a4) 8-ASK: Order

M =8
Bits
ld(M ) = 3 Bits/Symbol
1 Symbol
= 1500Bits/s
R1 = ld(M ) Symbol T
s
(1 point)
(a5) The symbol length must be divided in half. See problem a4)!
(1 point)
(b) (b1) Diagrams of I(t) and Q(t):

1
1
-1
-1
184
every diagram qualitive and quantitive correct

(b2) Diagram qualitive correct
quantitve correct
EXAMS
(1 point)
(1 point)
(1 point)
sBP(t)
0.5
0.5
1
0
2
t/T
(b3) I/Q-Diagram
Q[k] Decision
thresholds
I[k]
qualitive and quantitive correct

decision thresholds correct
(c) (c1) BER = 12 erfc(2)
erfc(2) = 4 103 ...6 103
BER = 2 103 ...3 103
(1 point)
(1 point)
(1 point)
(c2) The bit error rate increases.

Reason:
The phase oset between the transmit and receive oscillator leads to a rotation of
the I/Q-diagram in the receiver with unknown angle. By this the constellation points
approximate to the decision thresholds or even cross them. Additionally due to the
channel noise the sampled values in the receiver vary around the constellation points,
leading to so called noise clouds. Therefore the decreasing distance of the constellation
points to the decision thresholds results in more wrong decisions of the decider and
thus the bit error rate increases.
(1 point)
185
D.7 Exam WS 2007/2008
D.7
Exam WS 2007/2008

Note: Questions a), b) and c) can be solved independently from each other. Please note the hints
at the end of this problem on the back side.
(a) (5 Points) The rst digital cellular network in Germany was built in 1992. The maximal
transmit power of the mobiles is PT x = 2 W. The path loss LA is dened by:
d
dB .
LA = 130 dB + 30 log10
km
(a1) Compute the maximal transmit power level in dBm of the mobiles!
(a2) The mobile transmits to the base station with a constraint of having a minimal received power level of LRx = 106 dbm at the antenna of the base station. What is the
maximum allowed distance dmax between the mobile and the base station?
The received signal form the mobile at the base station is composed by the attenuated
transmitted signal and the thermal noise, which is added by the receiver front-end. The
noise power level is Ln = 107 dBm (See gure below). Furthermore, in order to have
correct demodulation of the received signal, the signal to noise ratio at the front-end of the
demodulator must have a minimum SNR of 10 dB.
PTx
LA
PRx
Noise
Modulator
Demodulator
Mobile
Base station
(a3) What is the maximum distance which can be allowed between the mobile and the base
station?
(b) (4 Points) A LTI system with the impulse response h(t) and the transfer function H(f ) is
given as shown in the gure below:
x(t)
LTI
y(t)
(b1) Give the relation between the input signal x(t) and the output signal y(t) in time
domain and between the related signal spectrum X(f ) and Y (f ) in frequency domain?
(b2) An amplier has the relation between the input signal x(t) and the output signal y(t):
y(t) = x(t) 0.2 x2 (t) 0.01 x3 (t). Decide whether the amplier is a LTI system! Give
reasons!
(b3) A LTI-System have the impulse response h(t) as shown in the gure below:
186
EXAMS
1
0.8
h(t)
0.6
0.4
0.2
0
0.2
0.4
0
t [s]
Decide and give reasons whether the LTI system can be realized in practise or how h(t)
must be modied, in order to implement the LTI system in practise!
(c) (6 Punkte) Two low pass lters, TP1 and TP2, are given with the transfer functions H1 (f )
and H2 (f ), respectively:
f
f
,
H2 (f ) = rect
H1 (f ) = rect
B1
B2
where B1 = 10 kHz and B2 = 20 kHz.
(c1) Draw the transfer function H1 (f ) of the low pass lter TP1 within the range 30 kHz <
f < 30 kHz! Make sure you label the axes of the diagram correctly!
(c2) Give the Fourier integral for the calculation of the impulse response h1 (t) of the low
pass lter TP1!
(c3) Calculate the real- and imaginary part of h1 (t) by solving the Fourier integral!
Now, a band pass lter should be constructed, having the lower cut frequency fg,u = 5 kHz
and the upper cut frequency fg,o = 10 kHz. For the construction of this band pass lter
the two described low pass lters, TP1 and TP2, and adding and subtracting elements are
available.
(c4) Sketch a suitable construction of the band pass lter, using the described lters and
elements.
Notes:
log(ab ) = b log(a)
x
log10 (x)
0.1 0.25 1 2
3
5 10 100 1000
1 0.6 0 0.3 0.48 0.7 1
2
3

(a) (2 points) Analog signals can be translated to digital signals by means of analog-digital
conversion. This enables digital signal processing. Give two advantages of digital signal
processing over analog signal processing!
187
D.7 Exam WS 2007/2008
(b) (4 points) Analog-digital converters with very high sampling rate typically have only low
quantization resolution. The analog-digital converter depicted in the following picture has
the sampling rate fA = 1/TA and a quantization resolution of b = 1 bit.
s a(t )
Sampling
x (t )
Quantization
s d(t )
Analog-Digital Converter
The sampling is carried out perfectly at time instances t = n TA with n Z and can
be expressed by a Dirac pulse train. The quantization characteristic of the analog-digital
converter is given by

q(x(t)) =
1 for x(t) 0
0 for x(t) < 0,
where x(t) denotes the continuous-valued signal before quantization.

(b1) Give the mathematical expression including the sampling rate fA and the quantization
characteristic q() that describes the relation between the input signal sa (t) and the
output signal sd (t) of the analog-digital converter!
(b2) Assume that the input signal of the analog-digital converter is sa (t) = cos(2 f2A t)! Draw
the related output signal sd (t) in the range of 0 t 8 TA ! Pay attention to the correct
axis labeling!
(c) (4 points) The analog low-pass signal sTP (t) has a cut-o frequency of fG = 10 kHz and
values in the range of 10 sTP (t) 10. To digitize sTP (t), an analog-digital converter is
required.
(c1) What is the minimum sampling rate fA,min that the analog-digital converter has to
provide to enable analog-digital conversion of sTP (t) without aliasing?
(c2) How can be avoided, that arbitrary interfering signals with frequencies f > 10 kHz lead
to aliasing errors, when sTP (t) is analog-digital converted using the minimum sampling
rate fA,min ?
(c3) Determine the quantization step size s, when the analog-digital converter employs a
linear mid-rise quantizer with a resolution of b = 3 bit and a dynamic input range that
is equal to the range of values of sTP (t)!
(c4) What is the signal-to-noise ratio SNR that results from the analog-digital conversion of
sTP (t) due to the quantization resolution of b = 3 bit? Assume sTP (t) to have a uniform
amplitude density over its whole range of values!
188
EXAMS
Problem 3: Analoge Modulation

(a) (7 Points) A transmitter based on amplitude modulation transmits the band pass signal
sBP (t) = (1 + x(t)) sin(2fc t),
where x(t) is the real-valued information signal to be transmitted, and fc denotes the carrier
frequency. Please note that the information signal is here upconverted with a
sinus - as opposed to the approach stated in the lecture!. The spectrum X(f ) of the
information signal x(t) is shown here - separated into its real and imaginary component:
1
0,5
-fg
Re{X(f)}
fg
fg << fc
1
0,5
f
Im{X(f)}
-fg
fg
(a1) Please derive the spectrum SBP (f ) of the band pass signal as a function of arbitrary
X(f ).
(a2) Draw the spectrum of SBP (f ) for given X(f ) and = 1! Please make sure that the axes
are appropriately annotated and real and imaginary signal components are separated!
(a3) Assume the information signal is now x(t) = 12 cos(2fi t). How large can be maximally
chosen, if an envelope demodulator is supposed to be used at the receiver side? Briey
argue why an appropriate choice of is important for this kind of receiver.
(a4) Please state an advantage and a disadvantage of this modulation scheme.
(b) (4 Points) We now assume that the band pass signal sBP (t) = xI (t) cos(2fc t) xQ (t)
sin(2fc t) is received at the antenna of the following demodulator:
xI (t )
sBP(t)
cos(2` fct+a)
Ideal low pass

|f| fg
Ideal band pass

(Suppression of
neighboring
channels)
xQ (t )
-sin(2` fct+a)
Ideal low pass

|f| fg
(b1) Derive the signals xI (t) and xQ (t) at the output of the demodulator as a function of
and arbitrary real-valued information signals xI (t) and xQ (t). Note:
cos(a)cos(b) = 1/2(cos(a + b) + cos(a b))
sin(a)sin(b) = 1/2(cos(a b) cos(a + b))
cos(a)sin(b) = 1/2(sin(a + b) sin(a b))
(b2) If the receiver knows the phase shift , the following extension to the demodulator
can be used to reconstruct the information signal xI (t). Please draw a sketch of a
corresponding extension of the demodulator that can reconstruct xQ (t). Make sure
your sketch is appropriately annotated.
189
D.7 Exam WS 2007/2008
x I (t )
2 cos ( )
xI ( t )
xQ (t )
2sin ( )

(a) (3 Points)
(a1) Why is digital modulation necessary?
(a2) Sketch the block diagram of a digital transmission line beginning at the send bit series
up to the received, decoded bit series!
(b) (6 Punkte)
(b1) Sketch one constellation-(I/Q)-diagram for each of the following modulation schemes:
8-Phase Shift Keying (PSK) and On-O Keying (OOK)!
(b2) Draw the decision thresholds into the I/Q-diagrams of problem b1)!
(b3) The symbol duration of both modulation schemes (OOK and 8-PSK) shall be 2ms.
How large is the achievable data rate in bit/s at each modulation scheme?
(c) (5 Punkte) A bit series {bk }, k N, bk {0, 1} shall be transmitted across a channel with
additive white Gaussian noise (AWGN-channel). The bandpass signal s(t), which is send
after digital modulation, is superimposed by the noise n(t). As modulation scheme binary
phase shift keying (BPSK) is used. The symbols are given by sk = 1 2bk . A rectangular
impulse shaping in applied.
(c1) The bit series bk = {0110} , k = 0, 1, 2, 3 shall be transmitted. Draw the corresponding
send bandpass signal s(t) in the interval from 0 to 4T! The bandpass signal must have
the following parameters: signal amplitude A, symbol duration T, carrier frequency
fc = T2 . Pay attention to completely labeled axis!
(c2) At the receiver the send bandpass signal is superimposed by noise. Calculate the bit
error rate of the received signal! At the receiver there is an SNR of 6dB and all send
symbols have the same probability. (Also be aware of the hints at the end of this
problem.)
(c3) A bit error rate of 1.5 103 must be achieved. Calculate the necessary SNR (in dB!)
at the receiver entrance!
(c4) The channel is no simple AWGN-channel anymore. Additional to the noise every symbol
experiences independently from each other a random phase shift. How large is the bit
error rate now with and without noise?
Hint: If the transmission is corrupted by additive Gaussian noise, the bit error rate Pb
(BER) in such a BPSK transmission system can be calculated by the following equation:
"%
#
1
SN Rlinear
BER = erfc
2
2
190
EXAMS
The complementary Gaussian error function erfc(x) is given in the following draw.
0
10
10
10
10
10
erfc(x)
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
191
D.8

(a) Power level calculation
(a1) Power level of the transmitted signal
P
Tx
dBm
1 mW
2 1000 mW
= 10 log10
dBm
1 mW
= (3 + 30) dBm = 33 dBm
LT x = 10 log10
Result correct
(1 Point)
(a2) Maximal distance

Maximal path loss: LA = LT x LRx = 33, dBm (106 dBm) = 139 dB
and thus:
d
LA = 139 dB = 130 dB + 30 log10
dB
km
d
9
dB = log10
dB
30
km
d
100.3 =
km
dmax = 2 km
approach correct
result correct
(1 Point)
(1 Point)
(a3) Required minimal receive power level:

LRx,min = SN R + Ln = 10 dB + (107 dBm) = 97dBm
Maximal path loss: LA,max = LT x LRx,min = 33 dBm (97 dBm) = 130 dB
and thus:
d
130 dB = 130 dB + 30 log10
dB
km
d
0 = log10
km
d = 1 km
approach correct
result correct
(1 Point)
(1 Point)
(b) (b1) LTI System:

Time domain: y(t) = h x (t)

Frequency domain: F y(t) = Y (f ) = F h(t) F x(t) = H(f ) X(f )
Both equations correct
(1 Point)
192
EXAMS
(b2) The amplier is not a LTI system, because his characteristic contains quadratic and
cubic terms.
(1 Point)
(b3) The LTI system can not be implemented in pratice, because h(t) = 0 for t < 0 and thus
= h(t t0 )
is not causal. By shifting the impulse response by t0 3 s to the right: h(t)
the causal impulse response h(t) can be derived.

Correct decision (yes/no)
(1 Point)
(1 Point)
Reson and modication (Shifting of h(t) by t0 3 s) correct
(c) (c1) See gure below
H1(f)
1
-B1/2
B1/2
gure correct
(1 Point)
(c2) Inverse Fourier integral:

h1 (t) =
H1 (f ) ej2f t d f
Equation correct
(1 Point)
(c3)

h1 (t) =
rect
Re{h1 (t)} =
f j2f t
e
df =
B1
+ sin(2f t) ,B1 /2
B1
2

cos 2f t + j sin 2f t d f
B1
2
2t
B /2
2B1 t 1
sin
2
=2
= B1 sinc(B1 t) = B1 si(B1 t)
2t
+ cos(2f t) ,B1 /2
=0
Re{h1 (t)} =
2t
B1 /2
Approach correct
Result correct
(2 Points)
(1 Point)
(c4) Bandpass lter
H2(f)
+
-
H1(f)
Constellation correct
(1 Point)
193

(a) Advantages of digital signal processing are (among others):
- independent of technology and temperature
- easier to implement compared to analog signal processing
- simple to parameterize and recongure
- some signal modications can not be done by analog signal processing
- repeatable results
- simple signal storage
(b) (b1) Mathematical expression as given in equation below:
Sampling using Dirac pulse train correct
Quantization using quantization characteristic q() correct
(2 points)
(1 point)
(1 point)

!
!
sd (t) = q sa (t)
(t/TA n) = q sa (t)
(t fA n)
n=
n=
(b2) Signal sd (t) as shown in gure below:

(1 point)
(1 point)
s d(t )
1
0
...
0
t /TA
(c) (c1) fA,min > 2 fG = 20 kHz
(1 point)
(c2) To avoid aliasing errors caused by interfering signals with frequencies f > 10 kHz, an
anti-aliasing lter, i.e. a low-pass lter with cut-o frequency fG = 10 kHz has to be
applied before the analog-digital converter.
(1 point)
(c3) The dynamic input range of the linear mid-rise quantizer is symmetric w.r.t 0. The
width of the dynamic input range is equal to the width of the range of values of sTP (t),
which yields max(sTP (t)) min(sTP (t)) = 20. The total number of quantization steps
is given by q = 2b/bit = 23 . Consequently, the quantization steps size results in:
s =
max(sTP (t)) min(sTP (t))

20
= 3 = 2.5
q
2
(1 point)
(c4) The dynamic input range of the quantizer of the analog-digital converter is equal to the
range of values of sTP (t). For an input signal with uniform amplitude density over the
whole dynamic input range, the signal-to-noise ratio SNR at the output of the quantizer
is given by:
SNR = 6.02 dB b/bit
194
EXAMS
Consequently, the signal-to-noise ratio SNR that results from the analog-digitalconversion of sTP (t) due to the quantization resolution of b = 3 bit yields:
SNR = 18.06 dB 18 dB
(1 point)

Im{X(f)}
-fc-fg
SBP (f)
j/2
0.25
-fc
-fc+fg
Re{X(f)}
fc-fg
fc
fc+fg f
-j/2
Equation correct:
Plot qualitive correct:
Plot quantitive correct:
(a2) t : 1 + x(t) > 0 < max|x(t)| < 1
One of the following aspects is mentioned:
(1 point)
(1 point)
(1 point)
(1 point)
(1 point)
Information signal may only be contained in the envelope of sBP (t)

Phase jumps in sBP (t) must be avoided
(a3) Power of the information signal in the band pass is si,BP (t) = x(t) cos(2fc t):
1
Pi,BP = 2 Px 21 = 0.52 12 12 = 16
Total power of the band pass signal sBP (t):
9
Ps,BP = 2 Px 21 + 12 = 16
Pi,BP
= 19
Power ratio: Ps,BP
Correct approach:
(1 point)
Correct result:
(1 point)
(b) (b1) The signals at the output of the demodulator can be calculated as
xI (t)
= (1 + x(t)) cos(2fc t) cos(2fc t + )|0<|f |fg

= 21 (1 + x(t)) (cos(4fc t + ) + cos)0<|f |fg

= 12 (1 + x(t)) cos0<|f | = cos
x(t)
2
xQ (t)
= (1 + x(t)) cos(2fc t) (-sin(2fc t + ))|0<|f |fg

= 12 (1 + x(t)) (sin(4fc t + ) sin())0<|f |fg

= 12 (1 + x(t)) sin0<|f | = sin
x(t)
2
Correct approach:
Correct result:
(b2) The following solution is possible:
(1 point)
(1 point)
195

x1 (t )
x1 (t )
2 cos ( )
xI (t )
x2 (t )
2sin ( )
xQ (t )
x2 (t )
2sin ( )
2 cos ( )
Correct principle:
Correct annotation:
(1 point)
(1 point)

(a) (a1) Digital modulation is necessary in order to transmit digital signals across an analog
channel.
(1 Punkt)
(a2)
bs {k }
impulse
shaping
filter
symbol
mapper
up
conversion
channel
bR {k }
symbol
demapper
decision
maker
sampling
Rreceiving
filter/
Matched
filter/
Integrator
down
conversion
at least 3 components correct:

3 more components correct:
(1 point)
(1 point)
(b) (b1)
Q
8-PSK
OOK
-A
-A
per diagram:
(1 point)
(b2)
196
D
Q
8-PSK
EXAMS
OOK
-A
-A
decision thresholds
per diagram:
(1 point)
(b3) Data rate D = bit/symbol * symbol/second

8-PSK: 2ms result in 500 symbols per second; 1 symbol transmits 3 bits
D = 3bit/symbol 500symbols/s
D = 1500bit/s
result correct
(1 point)
OOK: 2ms result in 500 symbols per second; 1 symbol transmits 1 bit
D = 1bit/symbol 500symbols/s
D = 500bit/s
result correct
(1 point)
(c) (c1)
A/2
-A/2
-A
T
2T
3T
BPSK modulated bandpass signal
4T
diagram qualitatively correct: 1 point

diagram quantitatively correct: 1 point
(c2) The logarithmic SNR must be transformed into linear SNR.
SN Rdb = 6dB = 3dB + 3dB = 10lg(SN R1 ) + 10lg(SN R1 )
197
SN Rlin = SN R1 SN R1 = 2 2 = 4
"%
#
1
SN Rlin
erfc
BER =
2
2

1
erfc
=
2
2
1
=
erfc (1.4)
2
In the diagram an x = 1.4 results in erfc = 5 102
BER =
1
5 102 = 2.5 102
2
correct is: 2 102 Pb 3 102
(1 point)
(c3)
BER = 1.5 103
3 103
"%
#
SN Rlin
1
erfc
=
2
2
"%
#
SN Rlin
= erfc
2
In the diagram a erfc(x) = 3 103 results in x = 2.1

%
SN Rlin
= 2.1
2
SN Rlin
= 4.41
2
SN Rlin = 8.82 9
SN Rdb = 10lg(9)dB 10dB
(1 point)
(c4) Because of the random phase shift detection at BPSK becomes impossible. In the
I/Q-diagram it can be seen independently from noise half of the symbols slip onto the
other side of decision threshold. Noise does not play any role anymore. Therefore, BER
(with and without noise) is:
BER = 0.5 = 50%
(1 point)
198
D.9
EXAMS
Exam SS 2008

Hint: Parts (a), (b) and (c) can be solved independently.
(a) (5 points)
Given the following signal:
x(t) =
N2

(1)n n (t nT ) , N1 < N2
n=N1
(a1) Draw x(t) for N1 = 3 and N2 = 3! Pay attention to correct axis labeling!
(a2) Consider now the more general case when N1 = N2 , where N2 > 0. Answer the
following questions!
i. Can the signal x(t) be a valid impulse response of a causal system?
ii. Is the signal x(t) odd?
iii. Is the spectrum X(f ) of the signal x(t) real valued?
(b) (5 points)
Two LTI systems with impulse response h1 (t) = rect(t 1/2) and h2 (t) = rect(t) are
serially connected, resulting in an overall system with impulse response h(t), as shown in
the following gure.
x(t)
y(t)
h1 (t)
h2 (t)
h(t)
(b1) Draw the impulse response h(t) of the overall system! Pay attention to correct axis
labeling!
(b2) Determine the transfer function H(f ) = F{h(t)}!
(b3) Draw the output signal y(t), when the input signal x(t) has the following shape:
199
D.9 Exam SS 2008
x(t)
2
1
1
3
t
2
1
2
3
(c) (5 points)
The picture bellow shows the block diagram of a radio transmission link.
LA
LT
PN 1
LR
PN 2
SN R1
SN R2
Transmitter
Demodulator
PT
PR
The power attenuation LA over a distance d can be approximated by the following equation:
LA = 95 dB + 25 log10
d
dB
km
The power of the transmitted signal is PT = 1 W. The noise powers at the receiver are
PN 1 = 0.1 pW and PN 2 = 90 pW. The amplier has a gain of v = 20 dB.
(c1) What is the power level LR in dBm for a distance of 4 km?
(c2) Determine the signal-to-noise ratio SN R1 at the input of the amplier as a function of
d!
(c3) What is the maximum distance between transmitter and receiver if the signal-to-noise
ratio SN R2 at the input of the demodulator should be at least 10 dB?
Hints:
x
log10 (x)
0.1 0.25 1 1.5

2
3
5 10 100 1000
1 0.6 0 0.18 0.3 0.48 0.7 1
2
3
200
EXAMS
Problem 2: Analog-to-Digital Conversion

(a) (7 points)
A mono audio signal s(t) shall be digitized, stored and reconstructed later on with the equipment given below. Unfortunately, the signal s(t) is superimposed by an interference signal
i(t) at the microphone. The associated signal spectra S(f ) = F{s(t)} and I(f ) = F{i(t)}
are depicted in the gure below. The sampling frequency is fs = 40 kHz. At rst, quantization errors shall be neglected. The reconstruction low pass lter TPR shall be considered as
ideal, having a cuto frequency of fR = 10kHz.
s(t) + i(t)
Sampling
sd (t)
Quantization
sq (t)
Memory
fs
fs
TPR
fR
a) Block diagram of the audio recorder/player
1.2
1.0
I(f )
40 38
10
S(f )
10
I(f )
38 40 f /kHz
b) Spectra S(f ) and I(f )
(a1) Derive an equation that relates the ideally sampled signal sd (t) to the analog signals
s(t) and i(t). Derive also the equation of the spectrum Sd (f ) = F{sd (t)} in function of
S(f ) and I(f )!
(a2) Draw the spectrum Sd (f ) in the range of 60 kHz f 60 kHz! Pay attention to
correct axis labeling!
(a3) Now, the sampling frequency is changed to fs = 28 kHz. Draw the spectrum Sd (f ) in
the range of 42 kHz f 42 kHz!Pay attention to correct axis labeling!
(a4) Due to this reduced sampling frequency aliasing is avoided. Determine the minimum
sampling frequency fs,min to avoid aliasing.
(b) (3 Points)
Assume the amplitudes of the sampled audio signal sd (t) of problem (a) to be uniformly
distributed in the range of Amax sd (t) Amax . Now, sd (t) shall be quantized using
a linear mid-tread quantizer with a resolution of 3 bits and a symmetrical dynamic range
[Amax , Amax ]. Note that one quantization level remains unused.
(b1) Sketch the quantization characteristic of the mid-tread quantizer described above!
(b2) For the quantization of sd (t) a linear mid-rise quantizer with a resolution of 3 bits
and a symmetrical dynamic input range [Amax , Amax ] can also be used. Which of the
described quantizers causes less quantization noise power? Give reasons!
201
D.9 Exam SS 2008

(a) (3 points)
Analog modulation is understood as imposing an analog source signal x(t) on a carrier signal
sc (t) = A cos(2fc t + ).
(a1) Explain why analog modulation is required to transmit signals in communications systems!
(a2) Which analog modulation schemes exist? Denominate the modulation schemes and
specify the parameters of sc (t) that are modulated.
(b) (4 points)
sm (t)
The following gure shows the plot of an analog modulated signal sm (t), where the analog
source signal is given by x(t) = Ax cos(2fx t) and the carrier signal is given by sc (t) =
cos(2fc t).
5
4
3
2
1
0
1
2
3
4
5
0
4
t/s
(b1) What is the value of the frequency fc of the carrier signal sc (t)?
(b2) What is the value of the frequency fx of the analog source signal x(t)?
(b3) Compute the modulation index and decide, whether an envelope detector can be used
to recover the source signal x(t) from the modulated signal sm (t)!
(c) (4 points)
A single sideband modulated received signal sBP (t) shall be demodulated with a heterodyne
receiver. The magnitude spectrum |SBP (f )| of sBP (t) is shown in the gure below.
202
EXAMS
|SBP (f )|
2
1
-600
-590
590
600
f / MHz
The next gure shows the structure of the heterodyne receiver. The frequency of the rst
mixer is given by fLO = 450MHz and the frequency of the second mixer is given by fIF =
150MHz.
sBP (t )
IF bandpass filter
Lowpass filter
z (t )
100MHz<|f |< 200MHz
y(t )
|f |< 15MHz
cos(2fIF t )
cos(2fLO t )
(c1) Draw the magnitude spectrum |Z(f )| of the IF signal z(t) at the input of the second
mixer in the range of 250MHz f 250MHz! Pay attention to correct axis labeling!
(c2) Draw the magnitude spectrum |Y (f )| of the demodulated received signal y(t) in the
range of 50MHz f 50MHz! Pay attention to correct axis labeling!
A customer requested an order to you in order to develop a wireless digital communications
system. For this purpose, you can choose between two dierent modulation schemes: QPSK and
4-ASK. Due to source and channel coding schemes, all symbols can be assumed to be transmitted
with the same probability. The symbol amplitudes of QPSK and 4-ASK are given in the following
constellation (IQ) diagrams, where A is a parameter:
Q
4-ASK
QPSK
jA
-A
A
-j A
(a) (6 points)
2A
3A
203
D.9 Exam SS 2008
(a1) With QPSK or 4-ASK, more than 1 bit can be transmitted per symbol. By that, the
bits can be assigned to the symbols in such a way that the bit error rate is minimized
at a constant symbol error rate. Add mapping schemes of bits to symbols to the given
constellation diagrams of QPSK and 4-ASK which minimize the bit error rates!
(a2) Add the decision thresholds of the detector to the constellation diagrams of QPSK and
4-ASK!
(a3) Compute the average symbol energy ES of both modulation schemes assuming that
all symbols are transmitted with equal probability! (Hint: ES = E{|dk |2 }, dk is the
amplitude of each symbol.)
(b) (4 points)
In order to realize the wireless digital communications line you will need transmitters and
receivers. The block diagram of a QPSK transmitter is shown below:
cos(2 fC t )
QPSK-Mapper
bl
TS
mI (t )
s( t )
TS
Pulse Shaping
Filter
mQ (t )
sin(2 fC t )
bl denotes the bit stream to be transmitted, TS is the symbol duration, s(t) is the transmitted
bandpass signal and mI (t) and mQ (t) denote the inphase and quadrature phase components
of the transmitted signal s(t) in the baseband. The pulse shaping lter has an rectangular
impulse response: rect( TtS 12 ).
(b1) Design and sketch a suitable QPSK receiver for the given QPSK transmitter and name
its components!
(b2) What is the basic dierence between a 4-ASK transmitter and the given QPSK transmitter?
(b3) Baseband signals with zero mean value are preferred in communications engineering.
Do the baseband signals of the given QPSK and 4-ASK have a zero mean value with
the given pulse shaping lter?
(c) (2 points)
Consider the 4-ASK. There are conditional probability density functions of the amplitudes
of the sampled receive signal rk at the input of the detector at the receiver. These probability
density functions are denoted by p(rk |dk ). dk denotes the k th 4-ASK symbol which is dened
by the constellation diagram and can take the following values: dk = {0, A, 2A, 3A}.
(c1) Sketch (qualitatively) the conditional probability density functions p(rk |dk ) of all 4
symbols of the 4-ASK in one diagram for a channel without noise.
204
EXAMS
(c2) In your communications systeme there is an AWGN (Additive White Gaussian Noise)
channel present. Sketch (qualitatively) the resulting conditional probability density
distributions p(rk |dk ) of all 4 symbols of the 4-ASK in one diagram for an AWGN
channel.
(d) (2 points)
Consider the QPSK. The bit error rate (BER) of a QPSK transmission over an AWGN
channel can be computed via the following formula:
"%
#
1
ES
BER = erfc
2
4 2
ES denotes the average symbol energy. 2 is the noise variance of the AWGN channel given
by 2 = 1. The complementary Gaussian error function erfc(x) is given by the diagram
below.
0
10
10
erfc(x)
10
10
10
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
(d1) What is the value of the amplitude A of the symbols in your QPSK transmission system
that is required to achieve a bit error rate not greater than 1,5% if the transmission is
done over an AWGN channel?
205
D.10

(a) (a1)
x(t)
3
2
1
2
3
1
1
3
2
t/T
1
2
3
Positions and absolute values of the pulses correct:

All signs correct:
(a2)
(1 point)
(1 point)
i. No, the system would not be causal, since x(t) = 0 for t < 0 is not satised.
(1 point)
ii. Yes, the signal is odd, since x(t) = x(t).
(1 point)
iii. No, X(f ) is imaginary valued (and odd), since x(t) is real valued and odd. (1 point)
(b) (b1)
h(t)
1
1/2
1/2
Shape correct:
Axis labels correct:
3/2
(1 point)
(1 point)
(b2) From
h(t) = rect(t 1/2) rect(t)
follows
H(f ) = F{rect(t 1/2)} F{rect(t)} = ejf si(f ) si(f ) = ejf si2 (f )
(1 point)
206
EXAMS
(b3)
y(t)
0.5
1.5
2.5
4.5
3.5
1
2
3
Individual triangles correct:

Superposition correct:
(1 point)
(1 point)
(c) (c1)

1000mW
dBm 95 dB 25 log10 (2 2) dB
LR = LT LA = 10 log10
1 mW
= 30 dBm 95 dB 2.5(3 + 3) dB = 80 dBm
Result correct:
(1 point)
(c2) General power level at the receiver:

LR = 65 dBm 25 log10
d
km

dB
Noise power level at the amplier input:

10
10 mW
dBm = 100 dBm
LN 1 = 10 log10
mW
This gives:

SN R1 = LR LN 1 = 35 dB 25 log10
Calculation method correct:
Result correct:
d
km

dB
(1 point)
(1 point)
207
(c3) Overall power level at the demodulator input:

Ltotal = LR + v = LR + 20 dB
Overall noise power level at the demodulator input:
LN total

PN 1 vlinear + PN 2
0.1 pW 100 + 90 pW
dBm = 10 log10
dBm
= 10 log10
mW
mW

= 10 log10 107 dBm = 70 dBm
This gives:

SN R2 = 10 dB = Ltotal LN total = 65 dBm 25 log10

15 dB
d
10 log10
dB =
= 6 dB
km
2.5
d = 4 km
d
km

dB + 20 dB (70) dBm
Calculation method correct:

Result correct:
(1 point)
(1 point)

(a) (a1) Equation for s(t), i(t) and sd (t) in time domain:
sd (t) =
1
fs

!
s(t) + i(t) t k/fs
(1 point)
k=
Equation for S(f ), I(f ) and Sd (f ) in frequency domain:

!
f nfs
Sd (f ) = S(f ) + I(f )
(1 point)
n=
(a2) Graphic qualitatively correct:

Graphic quantitatively correct:
(1 point)
(1 point)
Sd (f )
1.2
1.0
-50
-42 -38
-40
-30
-10
-2 0 2
10
(a3) Graphic qualitatively and quantitatively correct:
30
38
42
50
f /kHz
40
(1 point)
208
EXAMS
Sd (f )
1.2
1.0
-40 -38
-28
-18
-16 -10
-12
10 12 16 18
38 40
28
f /kHz
(a4) In order to avoid aliasing the following conditions must be fullled:

Sampling theorem: fs 2fg
|fu + mfs | > fR , m = 1, 2, ...
|fo + nfs | > fR , n = 1, 2, ...
where fu = 38 kHz and fo = 40 kHz denote the lower and upper corner frequency of i(t).
m = 1 leads to fs fu fR and n = 2 leads to fs
fo +fR
.
2
Thus, the minimum sampling frequency to avoid aliasing is fs,min = 25 kHz.

Approach and solution correct:
(2 points)
(b) (b1) Graphic qualitatively correct
(1 point)
sq (t) s
s
Amax
Amax
(b2) Decision:
The mid-rise quantizer causes less quantization noise power.
sd (t)
(1 point)
Explanation:
(1 point)
The quantization step size of the mid-rise quantizer is lower than that of the mid-tread
quantizer.
209
Detailed Explanation:
Due to its symmetric dynamic input range the mid-tread quantizer comprises only
q = 2b 1 = 7
quantization levels and the quantization step size is s =
quantizer comprises
q = 2b = 8
2Amax
q
= 27 Amax . The mid-rise
quantization levels and the quantization step size is s = 2Amax

= 28 Amax .
q
For input signals the amplitudes of which are uniformly distributed within the dynamic
input range of the quantizer, the quantization noise power Pe is given by (see script):
Pe =
1 2
s
12
Thus, the linear mid-rise quantizer causes less quantization noise power than the midtread quantizer.

(a) (a1) Main reasons for analog modulation:
(1 point)
1) Matching the source signal to the propagation channel.

2) Transmitting dierent analog source signals using the same propagation channel,
e.g. by frequency multiplexing.
Points: At least one correct answer required!
(a2) Amplitude modulation (AM): A A(x(t))
Phase modulation (PM): (x(t))
Frequency modulation (FM): fc f (x(t))
(2 points)
Points: 1 point for modulation schemes, 1 point for modulated parameters.

(b) (b1) fc = 4/1s = 4Hz
(b2) fx = 1/2s = 0.5Hz
max |x(t)|
, where
A0
max |x(t)| = Ax = 2 and A0 = 3 follows: = 2/3 = 0.666.
(b3) With sm (t) = [A0 + x(t)] cos(2fc t) and =
(1 point)
(1 point)
(1 point)
Yes, an envelope detector can be used, since the source signal x(t) is completely (1 point)
represented by the envelope of sm (t) ( < 1).
(c) (c1) Magnitude spectrum |Z(f )| as shown in gure below.
(2 points)
210
EXAMS
|Z (f )|
1
0.5
-250
-150
-140
140
150
250 f / MHz
Points: 1 point for correct shape, 1 point for correct axis labeling
(c2) Magnitude spectrum |Y (f )| as shown in gure below.
(2 points)
|Y (f )|
0.5
-50
-40
-30
-20
-10
10
20
30
40
50 f / MHz
Points: 1 point for correct shape, 1 point for correct axis labeling
(a) (a1) The following mappings are possible.
Important is to change only one bit between neighboring symbols.
(2 points)
Q
QPSK
4-ASK
01
11
00
00
10
11
10
10
Per correct mapping: 1 point

(a2) The following decision thresholds are correct.
(2 points)
Q
QPSK
4-ASK
Decision Thresholds
Per qualitatively correct diagram: 1 point

(a3) All symbols have the same probability: probability of one symbol is 14 .
sk is the amplitude of each of the 4 possible symbols.
1
1
1
1
ES = E{|sk |2 } = |s0 |2 + |s1 |2 + |s2 |2 + |s3 |2
4
4
4
4
(2 points)
211
1
(|s0 |2 + |s1 |2 + |s2 |2 + |s3 |2 )
4
1
1
=
(|A|2 + |jA|2 + | jA|2 + | A|2 ) = 4A2 = A2
4
4
2
1
A
=
(|0|2 + |A|2 + |2A|2 + |3A|2 ) =
(0 + 1 + 4 + 9) = 3.5A2
4
4
ES =
QP SK : ES
4 ASK : ES
(b) (b1) Basic receiver block diagram:
(2 points)
cos(2 fC t )
Demapper
hR (t ) = hS ( t )
r (t )
kTS
hR (t ) = hS ( t )
sin(2 fC t )
Receive Filter,
Integrator,
Matched Filter
Sampling
b(k )
Detector
(b2) The quadrature phase branch (or inphase branch) of the QPSK transmitter (1 point)
(and also receiver) can be removed. Transmission and processing is only carried out
in the quadrature phase (or inphase).
(b3) QPSK is free of a mean value. Due to the same probability of all symbols,
positive and negative parts are of same size and nullify in average.
(1 point)
The given 4-ASK is not free of a mean value, because all symbols are positive and
do not nullify each other in average.
(c) (c1) Conditional amplitude probability density function without any noise:
1
p(rR | d 0 )
d0 = 0
p(rR | d 1)
p(rR | d 2 )
d1 = A
d2 = 2A
p(rR | d 3 )
rR
d3 = 3A
(c2) Conditional amplitude probability density function with AWGN:

p(rR | d 0 )
d0 = 0
p(rR | d 1)
d1 = A
p(rR | d 2 )
d2 = 2A
(1 point)
(1 point)
p(rR | d 3 )
rR
d3 = 3A
(d) (d1)
(2 points)
1
0.015 =
erfc
2
"%
ES
4 2
212
"%
0.03 = erfc
ES
4 2
EXAMS
From the diagram it can be found that x = 1.5 for erfc(x) = 0.03 = 3 102 .
%
ES
1.5 =
4 2
ES
2.25 =
4 2
9 = ES
ES = E{|sk |2 } = E{|A|2 }
Due to the fact that all QPSK symbols have the same absolute amplitude A, the average
symbol amplitude is the same as the amplitude of any arbitrary QPSK symbol.

A = ES = 3
Correct x from the complementary Gaussian error function: 1 point
Correct symbol amplitude: 1 point
213
D.11 Exam WS 2008/2009
D.11
Exam WS 2008/2009

Hint: Parts (a), (b) and (c) can be solved independently.
(a) (4 points)
In communications, LTI sytems are often considered, for which there exist specic relationships between input signals x(t) and output signals y(t). The mappings f : x(t) y(t) of
LTI systems therefore have to fulll certain properties.
x(t)
y(t)
H
(a1) Which properties do linear systems have to fulll with respect to their mappings?
Give also a counter example of a mapping f : x(t) y(t) that is impossible for a linear
system!
(a2) Which properties do source free systems have to fulll with respect to their mappings?
Give also a counter example of a mapping f : x(t) y(t) that is impossible for a source
free system!
(b) (6 points)
Consider the signal
x(t) = t (t) ,
1
with (t) = 1/2
if t > 0
if t = 0
otherwise
and two positive constants t0 and T .

(b1) Draw the signal x
axis labeling!
t+t
T
within the range 2 t0 < t < 2 t0 ! Pay attention to correct
214
EXAMS
(b2) Show by means of a drawing how the signal

y(t) = triang(t/T )
can be expressed as a superposition of several shifted and scaled instances of the signal
x(t)! Pay attention to correct axis labeling! Give also a formula for this relationship!
Hint:

triang(t) =
1 |t|
if 1 < t < 1
otherwise
(b3) Determine the Fourier transform Y (f ) = F{triang(t/T )}!

(c) (5 points)
The following gure shows a device for adding the power of two incoming signals se,1 (t) und
se,2 (t). The power Pa of the output signal sa (t) is equal to the sum of the powers Pe,1 and
Pe,2 of the input signals.
n1 (t)
se,1 (t)
s1 (t)
Power Adder
se,2 (t)
s2 (t)
sa (t)
Pa = Pe,1 +
Pe,2
n2 (t)
(c1) Consider rst the case that two signals s1 (t) and s2 (t) with powers P1 = 200 mW and
P2 = 800 mW are combined noisefree, i.e. n1 (t) = n2 (t) = 0. Determine the power levels
L1 , L2 and La of the signals s1 (t), s2 (t) and sa (t) in dBm!
(c2) Now let the signals s1 (t) and s2 (t) from (c1) be disturbed by additive noise n1 (t)
and n2 (t) with powers Pn1 = 100 mW and Pn2 = 300 mW, respectively (see gure).
Determine the signal-to-noise ratio SN Ra at the output of the device in dB!
Hints:
x
log10 (x)
0.1 0.25 1 1.5

2
3
5 10 100 1000
1 0.6 0 0.18 0.3 0.48 0.7 1
2
3
215
D.11 Exam WS 2008/2009
Promblem 2: Analog-to-Digital Conversion

(a) (2 points)
Analog signals can be converted to digital signals by means of analog-to-digital conversion.
What are the advantages of representing analog signals in the digital domain. Give at least
two advantages!
Consider the conguration of an analog-to-digital converter depicted in gure (A). At the input
of the analog-to-digital converter, the desired signal s(t) is disturbed by an interfering signal i(t).
The respective frequency spectra S(f ) und I(f ) are shown in gure (B).
s(t )+i(t )
Sampling
z a(t )
Quantizer
z q(t )
Analog-to-Digital-Converter
(A)
I(f )
-35 -30 -25 -20 -15 -10 -5
S(f )
I(f )
10 15 20 25 30 35 f / MHz
(B)
(b) (4 points)
(b1) Draw the frequency spectrum Za (f ) of the signal za (t) resulting after sampling within
the range 35 HMz f 35 MHz assuming a sampling frequency of fa = 25 MHz!
Pay attention to correct axis labeling!
(b2) For technical reasons, the sampling frequency can not be higher than 50 MHz. Explain
two technically dierent solutions that nevertheless allow for recovering the desired
signal s(t) from the sampled signal za (t) without errors! Give reasons for both solutions!
Hint: An additional component can be used.
(c) (4 points)
(c1) Draw the quantization characteristic of the linear mid-rise quantizer, the input range
of which is [4, 4] and which has a quantization resolution of b = 3 bits. Pay attention
to correct axis labeling!
(c2) The signal za (t) at the input of the quantizer takes only values within the range 2
za (t) 2 with uninform amplitude distriubution. Determine the signal-to-noise ratio
SN R at the output of the quantizer with the quantization characteristic given in (c1)!
216
EXAMS

(a) (4 points)
(a1) Which are the symmetry properties of the real and imaginary part of the frequency
spectrum of a real-valued time domain signal?
Figure (A) shows the magnitude spectra A1 (f ) = |S1 (f )| and A2 (f ) = |S2 (f )| of the two
time domain signals s1 (t) and s2 (t) with frequency spectra S1 (f ) = F{s1 (t)} and S2 (f ) =
F{s2 (t)}.
A1 (f ) = |S1 (f )|
A2 (f ) = |S2 (f )|
(A)
(a2) Based on A1 (f ), decide whether s1 (t) is a real-valued time domain signal! Give reasons!
(a3) Based on A2 (f ), decide whether s2 (t) can denitely be identied as real-valued time
domain signal!
(b) (7 points)
The amplitude modulated baseband signal s3 (t) of a radio station is received and demodulated by the receiver depicted in Figure (B).
Demodulator
s3 (t)
m(t)
|f |>180 kHz
|f |<220 kHz
2 cos(2fLO t)
|f | < 40 kHz
2 cos(2fZF t)
(B)
The received signal is given by s3 (t) = (A + x3 (t)) cos(2fc t), where the carrier frequency
A
is fc = 1 MHz and the modulation index is = | max{x
< 1. The frequency spectrum
3 (t)|}
S3 (f ) = F{s3 (t)} is shown in Figure (C).
S3 (f )
I(f )
4
2
1
2
1
f /MHz
1.02 1.0 0.98
0.98 1.0 1.02
(C)
f /kHz
620 600 580
580 600 620
(D)
217
D.11 Exam WS 2008/2009
The receiver down-converts the received signal to the intermediate frequency fZF = 200 kHz.
(b1) Which two oscillator frequencies fLO of the rst mixer stage are feasible to down-convert
s3 (t) to fZF ?
(b2) Draw the frequency spectrum M (f ) = F{m(t)} of the signal after the rst mixer stage
for the case of fLO < fc within the range 250 kHz f 250 kHz! Pay attention to
the correct axis labeling!
Due to a defect, the amateur radio transceiver of your neighbor transmits an interfering
signal i(t) to the antenna of your receiver. The corresponding frequency spectrum I(f ) is
shown in Figure (D).
(b3) Sketch the spectrum M (f ), resulting from the superimposed reception of s3 (t) and i(t)
for the case of fLO < fc within the range 250 kHz f 250 kHz!
(b4) What kind of lter (high-pass, low-pass, band-pass, band-stop) with which cut-o frequencies has to be employed between the antenna and the rst mixer in order to assure
an interference-free demodulation of s3 (t) with the interfering signal i(t) at the antenna?
(b5) The demodulation of m(t) can also be done by a dierent method than shown in Figure
(B). Sketch a dierent type of demodulator that is suitable for the demodulation of m(t)
and name its components!
A wireless communications system can be used with two dierent modulation schemes: either with
8-PSK or with 4-QAM. Dependening on the quality of the channel and the resulting bit error rate
one of the modulation schemes is chosen in such a way that the data rate is maximized while
reliability of data transmission is ensured.
(a) (4 points)
(a1) Sketch an IQ-diagram (constellation-diagram) of the 8-PSK modulation!
(a2) Add a possible assignment of bits to the symbols in the IQ-diagram of the 8-PSK which
minimizes the bit error rate at constant symbol error rate! What is the name of this
kind of mapping?
(a3) Add the decision thresholds of the detector to the IQ-diagram of the 8-PSK assuming
that each of the symbols is sent with the same probability!
(b) (2 points)
It is possible to switch from the 4-QAM mode to the 8-PSK mode if the bit error rate
of the 4-QAM transmission across an AWGN channel falls below 5 103 . For a 4-QAM
transmission across an AWGN channel, the bit error rate (BER) can be computed via the
following formula:
#
"%
ES
ES
1
BER = erfc
SNR[dB] = 10 log(SNR)
SNR =
2
4N0
N0
218
EXAMS
ES denotes the average symbol energy. N0 is the noise power of the expected AWGN channel.
The complementary Gaussian error function erfc(x) is given in the diagram below.
(b1) At the receiver input there is a signal-to-noise ratio of SNR = 12 dB. Is it possible to
switch to the 8-PSK mode in this case?
Hint: Make sure not to mix the linear and the logarithmic SNR!
0
10
10
10
10
10
erfc(x)
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
(c) (8 points)
The gure below shows the block diagram of a receiver for 4-QAM transmission and the
IQ-diagram the detector uses for bit detection including the bit to symbol assignments.
Q
cos(2 fC t + )
`10
( k + 1)T S
s(t )
() d t
k TS
kTS
( k + 1)T S
d(2k )
r1(kTS )
() d t
d(2k + 1)
r2 (kTS )
sin(2 fC t + )
Integrator
-C
`11
k TS
Sampling
Detector,
Decider
jC
`00
C
-jC
`01
Demapper
s(t) denotes the sent and received bandpass signal, fC is the carrier frequency, TS is the
symbol duration and denotes the phase oset between the transmitter and receiver oscillator. fC is related to TS by fC = T5S . In the IQ-diagram, C denotes a constant factor that
is not specifed in more detail.
219
D.11 Exam WS 2008/2009
(c1) Sketch a suitable 4-QAM transmitter that corresponds to the given 4-QAM receiver
starting at the bit series d(k) and name the components!
Now, assume the signal s(t) = A cos(2fC t) A sin(2fC t) to be transmitted.
(c2) Compute the signals r1 (kTS ) and r2 (kTS ) in the inphase and quadrature phase branch
of the receiver after integration under the condition = 0 for the transmitted signal
s(t)! See also the hints given below.
(c3) Which 2 bits will be decoded in case of the transmitted signal s(t) according to the
IQ-diagram of the detector?
(c4) Which 2 bits will be decoded if there is a phase oset of = 90 between the transmitter and receiver oscillator. Give reasons for your decision!
Hints:
sin(x) cos(y) = 12 (sin(x y) + sin(x + y))
sin(x) sin(y) = 12 (cos(x y) cos(x + y))
cos(x) cos(y) = 12 (cos(x y) + cos(x + y))
cos(x) = sin( 2 x)
sin(x) = cos( 2 x)
220
D.12
EXAMS

(a) (a1) If x1 (t) y1 (t) and x2 (t) y2 (t) the following relations must be satised (linearity):
x1 (t) + x2 (t) y1 (t) + y2 (t) and a x1 (t) a y1 (t) , a R
(1 point)
Counter example: x(t) y(t) = |x(t)|.
(1 point)
(a2) If x(t) = 0 for all t < t0 (source freeness): y(t) = 0 must hold for all t < t0 .
Counter example: x(t) y(t) = 2x(t)
(1 point)
(1 point)
(b) (b1)
x
t+t0
T
t0 /T
t0
Shape and labeling correct:
(2 Points)
(b2)
y(t)
x1 (t)
x3 (t)
x2 (t)
Shape and labeling correct:
(2 Points)
y(t) = x1 (t) + x2 (t) + x3 (t) = x
t+T
T

t
tT
2x
+x
T
T
(1 Point)
221
(b3)
(1 point)
Y (f ) = T si2 (f T )
(c) (c1)
(3 points)

2 100mW
L1 = 10 log10
dBm = 20 dBm + 3 dB = 23 dBm
1 mW
P2 = 4P1 L2 = L1 + 6 dB = 29 dBm
Pa = P1 + P2 = 5P1 = 1/2 10 P1 La = L1 3 dB + 10 dB = 30 dBm
In general we have:

La = 10 log10 10L1 /10 + 10L2 /10
(c2)
(2 points)

P1 + P2
SN Ra = 10 log10
dB
Pn1 + Pn2

4 100mW
dBm = 30 dBm 20 dBm + 6 dB) = 4 dB
= 30 dBm 10 log10
1 mW
(a) Advantages: Signal processing becomes independent of integration technology and
temperature; storage of digital signals is easier; digital signal processing allows for
additional / some special signal manipulations
(2 points)
(b) (b1) Frequency spectrum Za (f ) according to diagram below
Diagram qualitatively correct:
Diagram quantitatively correct (axis labeling):
(1 point)
(1 point)
2
1
-35 -30 -25 -20 -15 -10 -5
10 15 20 25 30 35 f / MHz
(b2) Applying an anti-aliasing / low-pass lter at the input of the analog-to-digital

converter with cut-o frequency 10 MHz < fG < 25 MHz, such that the interfering
signal is removed before the analog-to-digital conversion
(1 point)
Dedicated sampling frequency: 45 MHz < fa < 50 MHz, such that the frequency
images of the interfering signal do not superimpose with the frequency spectrum of
the desired signal.
(1 point)
(c) (c1) Quantization characteristic accordig to diagram below
Diagram quantitatively correct (axis labeling):
(1 point)
(1 point)
222
EXAMS
zq(t )
3.5
2.5
1.5
0.5
-4
-3
-2
-0.5 1
-1
za(t )
-1.5
-2.5
-3.5
(c2) Only half of the input range of the quantizer is used by the input signal za (t). Hence,
only 4 out of 2b/bit = 8 quantization levels are used. This corresponds to an eective
quantization resolution of b = 2 bits.
(1 point)
In case of an input signal with uniform amplitude distribution the signal-to-noise
ratio at the quantizer output is given by
SN R = 6.02 dB b /Bit
Thus, the signal-to-noise ratio resulting from the quantization of za (t) derives as
SN R 12 dB.
(1 point)
Solution 3: Analog Modulation
(a) (a1) The real part of the frequency spectrum of a real-valued time domain signal is axial
symmetric to f = 0 (even function) and the imaginary part is point symmetric to
f = 0 (odd function).
(1 point)
(a2) The signal s1 (t) is complex-valued. Due to the symmetry conditions described in (a1)
for frequency spectra of real-valued time domain signals, the magnitude spectrum of
any real-valued time domian signal is axial symmetric to f = 0.
Decision correct:
(1 point)
Reasoning correct:
(1 point)
(a3) It can not clearly be concluded, whether s2 (t) is real- or complex-valued. As explained
in (a2), the magnitude spectrum of a real-valued signal is axial symmetric. Hence,
s2 (t) could be real-valued. However, the real and imaginary part of the frequency spectrum S2 (f ) = F{s2 (t)} could also be as depicted in the following gure, which would
correspond to a complex-valued time domain signal.
Im{S2 (f )}
Re{S2 (f )}
223
Decision correct:
(1 point)
(b) (b1) Oscillator frequency of the rst mixer stage:

1. possibility: fLO = fc + fZF = 1.2 MHz
2. possibility: fLO = fc fZF = 0.8 MHz
Calculation and values correct:
(1 point)
(b2) Frequency spectrum M (f ) according to gure below

M (f )
4
2
1
f /kHz
220 200 180
180
200
220

Axis labeling correct:
(1 point)
(1 point)
(b3) Frequency spectrum M (f ) according to gure below

M (f )
4
2
1
f /kHz
220 200 180
180
200
220
(1 point)
(b4) Either a band-stop lter with lower cut-o frequency fg,u = 1.38 MHz and upper
cut-o frequency fg,o = 1.42 MHz or a band-pass or low-pass lter with upper
cut-o frequency fg,o = 1.38 MHz has to be inserted between the antenna and the
rst mixer.
(1 point)
(b5) An envelope demodulator can be applied:
Band pass
Rectifier
(Surpression of (Reconstruction of
adjacent channels)
envelope)
Low pass
(Cancellation
of carrier)
Conguration correct:
Labeling and description of the components correct:
High pass
(Cancellation of
DC offset)
(1 point)
(1 point)
224
EXAMS

(a) (a1) IQ-diagram as shown below
It is important that there are 8 constellation points arranged on a circle.
(1 point)
(a2) Assignment as shown below

It is important that adjacent symbols do only dier by one bit.
This kind of mapping is called GRAY mapping.
(1 point)
(1 point)
'111'
'101'
'110'
'100'
'000'
I
'010'
'001'
'011'
(a3) Decision thresholds as shown below

It is important that all thresholds lead through the origin of ordinates.
(1 point)
Decision
Thresholds
I
(b) (b1) Remember that 3 dB corresponds to 2 in the linear domain and 12dB = 4 3dB.
SNR[db] = 12dB = 4 3dB = 4 10 0.3dB
0.3 = log10 (2) (see also table of logarithms in problem 1)
4100.3
10
= (100.3 )4 = 24 = 16
"%
#
1
ES
BER =
erfc
2
4N0
"%
#
1
SNR
BER =
erfc
2
4
SNR = 10
225
1
BER =
erfc
2
BER =
"%
16
4
1
erfc (2)
2
In the diagram x = 2 corresponds to erfc(2) = 5 103 . Hence, we get:

BER =
1
5 103 = 2.5 103
2
This is smaller than 5 103 . So, the communciations system can switch to the 8-PSK
mode.
Linear SNR correct:
(1 point)
Result correct:
(1 point)
(c) (c1) General transmitter architecture as shown below
(2 points)
cos(2 fC t )
4-QAMMapper
d (k )
TS
m1(t )
s(t )
TS
m2 (t )
sin(2 fC t )
Puls shaping
filter
Up conversion
(c2)

(k+1)TS
r1 (kTS ) =
(A cos(2fC t) A sin(2fC t)) (cos(2fC t))dt
kTS
r1 (kTS ) = A
r1 (kTS ) = A
(k+1)TS
kTS
(k+1)TS

r1 (kTS ) = A
kTS
(k+1)TS
kTS

cos(2fC t) cos(2fC t)dt A
(k+1)TS
sin(2fC t) cos(2fC t)dt

kTS
(k+1)TS
1
1
(cos(0) + cos(4fC t))dt A
(sin(0) + sin(4fC t))dt
2
2
kTS
(k+1)TS
1
1
(1 + cos(4fC t))dt A
(sin(4fC t))dt
2
2
kTS
According to fC = T5S , sinus and cosinus are integrated over over full periods. Therefore,
these integrals become zero.
(k+1)TS
A
ATS
A
r1 (kTS ) =
dt = ((k + 1)TS kTS ) =
2
2
2
kTS
Similar to the computation of r1 (kTS ):

(k+1)TS
r2 (kTS ) =
(A cos(2fC t) A sin(2fC t)) (sin(2fC t))dt
kTS
226

r2 (kTS ) = A
(k+1)TS
cos(2fC t) sin(2fC t)dt + A

kTS
(k+1)TS
r2 (kTS ) = A

kTS
(k+1)TS
r2 (kTS ) = A

(k+1)TS
kTS
(k+1)TS
r2 (kTS ) =
kTS
EXAMS
sin(2fC t) sin(2fC t)dt

kTS
(k+1)TS
1
1
(sin(0) + sin(4fC t))dt + A
(cos(0) cos(4fC t))dt
2
2
kTS
(k+1)TS
1
1
sin(4fC t)dt + A
(1 cos(4fC t))dt
2
2
kTS
A
A
ATS
dt = ((k + 1)TS kTS ) =
2
2
2
Approach via the two integrals correct:

Integrals over full sinus and cosinus periods become zero:
Results for r1 (kTS ) and r2 (kTS ) correct:
(1 point)
(1 point)
(1 point)
(c3) The results from the previous solution can be reused. r1 (kTS ) is the inphase
component and negative. r2 (kTS ) is the quadrature phase component and positive.
This corresponds to the symbol in the upper left quadrant of the IQ-diagram which
will be decoded as 10.
(1 point)
(c4) Simple way: A phase oset of = 90 = 2 results in a virtual rotation of the
IQ-diagram at the detector by 90 in mathematically negative direction (clockwise).
So, over the received symbol, which lies in the upper left quadrant of the transmitter
IQ-diagram, there is the rotated demapper IQ-diagram (see also diagrams below). In
the rotated demapper IQ-diagram the received signal can be found in the lower left
quadrant and is recognized as AT2 S j AT2 S , which is decoded as 11.
Transmitter side
Receiver side
Virtual turn
of ReceiverIQ-diagram
by -90
1
I
Complex solution (computation):

(k+1)TS
r1 (kTS ) =

kTS
(k+1)TS
r2 (kTS ) =
kTS
(A cos(2fC t) A sin(2fC t)) (cos(2fC t
(A cos(2fC t) A sin(2fC t)) (sin(2fC t
sin(x ) = sin((x )) = sin( x) = cos(x)

2
2
2
cos(x ) = cos((x )) = cos( x) = sin(x)

2
2
2
))dt
2
))dt
2
227
(k+1)TS
r1 (kTS ) =
kTS
(k+1)TS
r2 (kTS ) =
(A cos(2fC t) A sin(2fC t)) sin(2fC t)dt

(A cos(2fC t) A sin(2fC t)) cos(2fC t)dt
kTS
After multiplying out, applying addition theorems and integration (partially over full
periods of sinus and cosinus like in (c2)):
ATS
2
ATS
r2 (kTS ) =
2
r1 (kTS ) =
The received symbol is thus in the lower left quadrant of the I/Q diagram and decoded
as 11.
Correct bit combination:
(1 point)
Reason for decision (explanation, diagram or computation):
(1 point)
228
D.13
EXAMS
Exam SS 2009
Problem 1: Signal Levels and LTI Systems

Note: Questions (a), (b) and (c) can be solved independently of one another.
(a) (7 points)
In LTI systems, which are often used in communications technology, the output signal y(t)
is distinctly related to the input signal x(t). We assume the following LTI system
x(t)
h(t)
with the impulse response

h(t) =
y(t)
eat
,t 0
, otherwise.
(a1) Give the general relation between x(t), y(t) and h(t).
(a2) Draw a sketch of h(t)!
(a3) Calculate the transfer function H(f ) for the given impulse response h(t).
We now assume the input signal to be:
x(t) = a1 sin(2f1 t) a2 cos(2f2 t)
(a4) Draw a sketch of the magnitude spectrum |X(f )| for a1 < a2 and f1 < f2 !
(a5) Calculate and draw the magnitude spectrum |Y (f )| of the output signal
for the following
parameter values: a = 4, a1 = 10, a2 = 11, f1 = 3/(2) and f2 = 105/(2)! Pay
attention to correct axis labeling!
(b) (5 points)
In wireless communications systems, data transmission between a transmitter and a receiver
is often disturbed by signals of other transmitters that send at the same time using the
same carrier frequency. In this case, the data signal is impaired by both, the thermal noise
at the receiver and the signals of other transmitters (termed interference). We consider the
following situation where transmitter 1 transmits the desired data to the receiver, while the
signals of transmitter 2 cause interference to the data transmission.
LS1
Receiver
PN
Transmitter 1
d1
Demodulator
d2
Transmitter 2
LS2
LE
LE
Lg = 20dB
229
D.13 Exam SS 2009
The attenuation of the radio channel is given as

LA = 20dB + 10 log10 (d/m)4 dB,
where d is the distance between transmitter and receiver.
(b1) Calculate the power level LS1 of transmitter 1 assuming that its transmit power is
PS1 = 250mW!
(b2) Calculate the power level LE1 of the received signal from transmitter 1 at the input of
the demodulator given that the power level of transmitter 1 is LS1 = 30dBm and the
distance between transmitter 1 and receiver is d1 = 20m!
(b3) How long is the distance d2 between transmitter 2 and receiver, if both transmitters use
the same transmit power, the ratio of signal power and interference power (SIR, signalto-interference-ratio) at the receiver is known to be 12 dB and the distance between
transmitter 1 and the receiver is d1 = 57m?
Note:
x
log10 (x)
0.1 0.25 1 1.5 2

3
5 10 100 1000
1 0.6 0 0.2 0.3 0.5 0.7 1
2
3
(c) (3 points)
The same scenario as in problem (b) is assumed.
(c1) What is the degradation in dB of the signal-to-disturbance ratio (ratio of the signal
power to the power of the disturbance), if the interference of transmitter 2 is considered,
as opposed to the case where the thermal noise at the receiver is the only considered
disturbance?
Calculate the degradation for the assumption that the power level of the received
signal of transmitter 1 is L E1 = 66dBm, the ratio of signal power to interference
power (signal-to-interference-ratio, SIR) at the receiver is given by 9dB and the power
of the thermal noise is PN = 0.01nW (and thus the noise power level LN = 80dBm)!
Note: The degradation equals the dierence of the SINR (signal-to-interferenceand-noise-ratio) and the SNR (signal-to-noise-ratio) in dB.
(a) (1 point)
Numerous communications systems use analog-to-digital conversion to transform analog
signals to digital signals. Name one of the main reasons!
(b) (4 points)
Assume the signal s(t) at the input of an analog-to-digital converter. The real-valued spectrum S(f ) of the signal s(t) is shown in the following gure.
230
EXAMS
efc g
h
ji
jh
c d
kHz
(b1) What is the minimum sampling frequency fs for the signal s(t) to be recoverable without
errors?
(b2) Name the eect which prevents recovering the signal s(t) without errors when the
sampling frequency fs is chosen too low!
(b3) Now, the sampling frequency is fs = 4kHz. Draw the spectrum Ss (f ) of the sampled
signal ss (t) in the interval f [8kHz, 8kHz]! Pay attention to correct axis labeling!
(c) (5 points)
The sampled signal ss (t) is quantized. For this, a linear mid-rise quantizer is used with an
input range dened as Ulower ss (t) Uupper with Ulower = 4V und Uupper = 4V. The step
size of the quantizer is s.
(c1) Compute s, in case the resolution of the quantizer is 3 bits! Draw the quantization
characteristic in the intervall [4s, 4s]! Pay attention to correct axis labeling!
(c2) We now assume the quantizer to have a resolution of 5 Bits. Compute the maximum
absolute quantization error emax .
(c3) What are the two options to increase the SN R (signal-to-noise ratio) of the quantization
by +12dB?
231
D.13 Exam SS 2009

(a) (3 points)
Shaping the waveform of a carrier signal by an analog source signal is known as analog
modulation.
(a1) Why is analog modulation used for transmitting analog signals in communications?
(a2) Compare amplitude modulation to frequency modulation in terms of the required ampliers and the occupied bandwidth when the same analog source signal is transmitted.
(b) (4 points)
Consider a signal s(t) that is transmitted by means of amplitude modulation. The modulated
band-pass signal is given as sBP (t) = (A + s(t)) cos(2fc t). The carrier frequency is fc =
100 kHz.
(b1) Draw sBP (t) in the range 0 s t 80 s for the case of A = 2 and s(t) = cos(2f1 t)
with f1 = 25 kHz! Pay attention to correct axis labeling!
(b2) Consider the signal s(t) to be given as s(t) = cos(2f1 t) + 41 cos(2f2 t), where f1 =
25 kHz and f2 = 12.5 kHz. Which value of A is required to allow for demodulation of
s(t) with an envelope detector at the receiver and to spend as little transmit power as
possible for the carrier signal?
(c) (4 points)
A modulated single-sideband signal sESB (t) shall be demodulated with a simple direct conversion receiver. The gure below shows the magnitude spectrum |SESB (f )| of sESB (t).
|SESB (f )|
1.0
0.5
-120
-110
100
-100
110
120
f / MHz
The next gure shows the block diagram of the direct conversion receiver. The mixer frequency is fLO = 120 MHz.
sESB (t )
sBB (t )
Low-pass filter
y (t )
|f |< fg
cos(2fLO t )
232
EXAMS
(c1) Give an equation that relates the frequency spectrum SBB (f ) of the signal sBB (t) after
the mixer to SESB (f )! Draw the magnitude spectrum |SBB (f )| in the range 40 MHz
f 40 MHz! Pay attention to correct axis labeling!
(c2) Consider a signal i(t) that interferes with sESB (t) at the receiver input. The gure below
shows the magnitude spectrum |I(f )| of i(t). What is the maximum cut-o frequency
fg of the low-pass lter at the receiver that can be allowed to remove i(t) from the
received signal?
|I (f )|
1.0
0.5
-95 -90
90 95
f / MHz

Consider two binary modulation schemes (A) and (B). Their constellation diagrams (IQ-diagrams),
including the mapping of the input bits, are illustrated in the following gure:
(A)
(B)
Q
A
0
1
(a) (4 points)
(a1) Name two further common binary modulation schemes!
(a2) Now the bit sequence {bk } = {0, 1, 1, 0} is transmitted. The symbol duration is Ts and a
rectangular function with hTs (t) = rect(t/Ts 1/2) is used for pulse shaping. Draw the
inphase and quadrature phase components I(t) and Q(t) of the modulated baseband
signals s(t) for the cases (A) and (B)! Pay attention to correct axis labeling!
(b) (4 points)
(b1) One of the two modulation schemes (A) and (B) is going to be used for data transmission
at a carrier frequency fc . The real valued bandpass signal at the receiver input is rBP (t).
Sketch a general receiver architecture that can be used for both modulation schemes
(A) and (B)! Name its components!
(b2) Draw the decision boundaries into the constellation diagrams of (A) and (B) in such a
way, that the bit error probability is minimized when the transmitted symbols occur
with equal probability.
233
D.13 Exam SS 2009
(c) (6 points)
For achieving a higher data rate, a higher order modulation scheme (C) is considered, having
the following constellation diagram:
Q
(C)
2A
2A
2A
2A
(c1) There are dierent ways of mapping the input bits to the symbols. Gray mapping
denotes a mapping rule that minimizes the bit error probability at a constant symbol
error probability.
Which property characterizes Gray mapping in general? Insert such a bit mapping into
the constellation diagram (C) given above!
(c2) Now the average energies E S,A and E S,B are considered for the modulation schemes (A)
and (B), respectively. In both cases the same amplitude A is used and the transmitted
symbols are assumed to occur with equal probability. Determine the ratio of the average
energies E S,A / E S,B !
(c3) The symbol error probability is usually dominated by the pairwise error probability
between symbols of lowest distance dmin . For the AWGN channel the pairwise error
probability of two symbols with distance d in the constellation diagram is given by
#
"%
1
d2
PS = erfc
,
2
8 2
where 2 denotes the variance of the noise and the complementary error function erfc(x)
is given in the gure below.
How large does the amplitude A need to be for case (B) to achieve a symbol error
probability of 103 for = 1/2?
234
D
0
10
10
10
10
10
erfc(x)
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
EXAMS
235
D.14

(a) (a1) y(t) = x(t) h(t) =
x(t )h( )d =
x( )h(t )d
(1 point)
(a2) Sketch - see gure below.

Sketch qualitatively correct:
(1 point)
h(t)
1
(a3) Correct transfer function:

j2f t
H(f ) =
x(t)e
dt =
eat ej2f t dt
0

t(a+j2f )
1
et(a+j2f ) dt =
=
e
a + j2f
0
1
=
a + j2f
(1 point)
(a4) Sketch - see gure below.

Sketch qualitatively correct:
(1 point)
|X(f)|
a 2/2
a1 /2
-f 2
(a5) Magnitude spectrum correct:

|H(f )| =
-f1
f1
f2
(1 point)
1
a2 + (2f )2
236
EXAMS
1
1
=
5
42 + 32
1
1
1
|H(f2 )| =|H(f2 )| =
=
=
11
42 + 105
121

a

a1
2
|Y (f )| = |H(f1 )| (f f1 ) + (f + f1 ) + |H(f2 )| (f f2 ) + (f + f2 )
2
2
1

(f f2 ) + (f + f2 )
= (f f1 ) + (f f1 ) +
2
|H(f1 )| =|H(f1 )| =
Plot - see gure below.

Qualitatively correct:
Quantitatively correct (axis labeling):
(1 point)
(1 point)
|Y(f)|
1
1/2
-f 2
-f1
f1
f2
(b) (b1) Result and unit correct:
(1 point)
LS1 = 10 log10 (250)dBm = 10 log10 (1000 0.25)dBm = (30 6)dBm = 24dBm

(b2) Channel attenuation correct:
(1 point)
LA1 = 20dB + 10 log10 (204 )dB = 20dB + 40 log10 (2 10)dB

= (20 + 52)dB = 72dB
Result and unit correct:
(1 point)
LE1 = LS1 LA1 + Lg = 30dBm 72dB + 20dB = 22dBm

(b3) Equations correct:
Result and unit correct:
(1 point)
(1 point)
SIR = 12dB = LS LA1 + Lg (LS LA2 + Lg )

signal power
interference power
= LA 2 LA 1
= 20dB + 40 log10 (d2 ) 20dB 40 log10 (57)

d2
3dB = 10 log10
57
d2 = 2 57m = 114m
237
(c) (c1) Solution approach correct:
(1 point)
SINR
P E1 /(PN + P E2 )
PN
PN
=
=
=

SNR
P E1 /(PN )
PN + P E2
PN I
dierence in dB:
SINR[dB] SNR[dB] = LN LN I
Combined disturbance power correct:
(1 point)
L E2 = L S1 SIR = 75dBm = 80dBm + 5dB P E2 = 3 108 mW

PN I = PN + P E2 = 4 108 mW
Degradation in decibels correct:
(1 point)
LN I = 10 log10 (PN I )dBm = (80 + 3 + 3)dBm = 74dBm

LN LN I = 80dBm (74dBm) = 6dB
Answer : The ratio of signal power to disturbance power degrades by 6dB, due to the
interference.
Problem 2: Analog-Digital-Conversion
(a) Advantages of digital signals:
(1 point)
Small sensitivity to distortion and disruption; simple storage of signals;
simple signal processing and manipulation; independence of technology and temperature
(b) (b1) Nyquist criterion: fs 2fg = 6kHz
(1 point)
(b2) Aliasing
(1 Point)
(b3) Spectrum - see gure below

Diagram qualitatively correct (S(f ) recuring):
Diagram quantitatively correct (labeling of the axes):
(1 point)
(1 point)
S s(f )
2
1
-7
-5
-3
-1
f / kHz
(c) (c1) Given the size of the input range as u = Uupper Ulower , the number of steps q and
the quantization resolution b in bits, the results are
238
q = 2b q = 2 3 = 8
u
Uupper Ulower
s =
= 1V
s =
q
8
EXAMS
(1 point)
Quantization characteristic - see gure below

Diagram quantitatively correct (labeling of the axes):
(1 point)
s q(t )
Ulower
ko
kn
km
l
k
s a(t )
km
kn
Uupper
(c2) Given the maximum absolute quantization error as emax = s/2 the results are
q = 2b q = 25 = 32
u
Uupper Ulower
s =
s =
= 250mV
q
32
250mV
= 125mV
emax =
2
(1 point)
(c3) Increase the number of bits used for quantization by 2;

(2 points)
increase the sampling frequency fs of the input signal s(t) by a factor of 16
(a) (a1) Main reasons for analog modulation:
(1 point)
1) Matching the analog source signal to the propagation medium, i.e. the transmission
channel.
2) Transmitting dierent analog source signals using the same propagation channel,
e.g. by frequency multiplexing.
Points: At least one correct answer required.
(a2) AM amplier for AM needs to be linear,
FM amplier can be non-linear
(1 point)
AM bandwidth is equal to the bandwidth of the analog source signal (2 fg ), (1 point)

FM bandwidth is larger than AM bandwidth (2 fg + 2 F , Carson rule)
239
(b) (b1) Signal sBP (t) - see gure below.

(1 point)
(1 point)
3
2
BP
s (t)
1
0
1
2
3
0
20
40
t / s
60
80
(b2) To allow for an envelope detector at the receiver the modulation index needs to be
=
A
1
max |s(t)|
To spend as little as possible energy for the carrier signal, A needs to be a small
point)
as possible, i.e.:
=
(1
A
!
=1
max |s(t)|
From this requirement and with max |s(t)| = 54 , for e.g. t = 0, follows:
(1 point)
5
A = max |s(t)| = .
4
(c) (c1) Equation for frequency spectrum SBB (f ):
SBB (f ) =
(1 point)
1
1
SESB (f fLO ) + SESB (f + fLO )
2
2
Magnitude spectrum |SBB (f )| - see gure below.

(1 point)
(1 point)
|SBB (f )|
1.0
0.5
-20
-10
10
20
f / MHz
240
EXAMS
(c2) Down-conversion with fLO = 120 MHz leads to the following frequency spectrum at the
input of the low-pass lter:
|SBB (f )|
1.0
0.5
-30 -25 -20
-10
10
20 25 30
f / MHz
To remove the down-converted interfering signal with a frequency band lower-bounded

by fu = 25 MHz, the cut-o frequency fg of the low-pass lter must not be greater
than fu , i.e.:
(1 point)
fg 25 MHz
(a) (a1) BPSK, 2-FSK
(1 point)
(a2) s(t) = I(t) + j Q(t)

(A) qualitatively and quantitatively correct:
(B) qualitatively correct:
(B) quantitatively correct:
(1 point)
(1 point)
(1 point)
(B)
(A)
I(t)
I(t)
3T
Q(t)
3T
Q(t)
A
(b) (b1) Basic receiver structure:
3T
4T
(2 points)
241

kT
cos(2fc t)
rI (t)
rI (k)
hT (t)
bk
rBP (t)
kT
rQ (t)
rQ (k)
hT (t)
sin(2fc t)
Receiver lter
Integrator
Matched lter
Sampling
Detector
242
(b2) (A) correct:

(B) correct:
EXAMS
(1 point)
(1 point)
(A)
(B)
Q
A
0
1
(c) (c1) Gray-mapping is characterized by the property that neighboring symbols onlx dier
(1 Point)
in a single bit. This way it is ensured that the most likely symbol errors cause only
a single bit error.
One possible mapping:
(1 Point)
Q
2A 011
010
001
110
2A
000
A
A
A
111
2A
100
2A 101
(c2) Approach correct:

Result correct:
E S,A = 12 (0 + A2 ) =
(1 point)
(1 point)
A2
2
E S,B = 12 (A2 + A2 ) = A2
(c3) Approach correct:

E S,A
E S,B
1
2
(1 point)
243
Result correct:
(1 point)
"%
2 PS = 2 10
= erfc
From the curve one obtains with d2 = 2 A2 :

%
%
d2
2 A2
=
2.2
8 2
2
d2
8 2
A 2.2
244
D.15
EXAMS
Exam WS 2009/2010
Problem 1: LTI-Systems und Signal Theory
Note: Question a), b) and c) can be solved independently from each other. Please also note the
hints at the end of this problem.
(a) (8 points)
Consider the following LTI-system
x(t)
h(t)
with the impulse response

h(t) = rect
y(t)
1
t
T
2

.
(a1) Give the general relation between x(t), y(t) and h(t) in frequency domain.
(a2) Is the given system causal? Give a reason for your answer!
(a3) Calculate the transfer function H(f ) and draw its magnitude spectrum in the range
T3 f T3 . Pay attention to correct axis labeling!
(a4) Let x(t) be
x(t)

x(t) =
A
T
t + A T t 0
otherwise.
-T
Determine the output signal y(t) and give its equation.
245
D.15 Exam WS 2009/2010
(b) (3 points)
The following gure depicts the radio communication link between a transmitter and a
receiver.
Receiver
PN
LS
Transmitter
Demodulator
LE
d
LD
Lg = 20dB
The transmitted signal is subject to attenuation on the radio channel. This attenuation is
directly related to the distance d (in meters) between transmitter and receiver. The relationship is given as follows

LA = 20dB + 10 log10 (d/m)4 dB.
At the receiver, the received signal is subject to additive noise (with power Pn ), is amplied
and reaches the demodulator which tries to reconstruct the transmitted information from
the received signal.
(b1) Calculate the transmit power PS (in Watts), if the transmitter uses LS = 46dBm for
transmission.
(b2) Calculate the received power level LD at the input of the demodulator if the transmit
power level is LS = 43dBm, Pn = 0 and the receiver is located at a distance of d = 250m
from the transmitter.
(c) (4 points)
We consider the same radio communication link as considered in the problem above.
(c1) Many practical systems use antennas that have a certain directional characteristic (referred to as antenna pattern). This means that transmit signals are amplied or attenuated depending on the transmit direction. We now assume the following positioning
of transmitter and receiver.
Receiver
Transmitter
Furthermore, we assume the transmitter uses an antenna with a directional amplication that depends on the angle (angle between the main direction of the transmit
antenna and the receiver) in the following way:
Lant = 24 cos() dB.
246
EXAMS
Calculate the signal-to-noise-ratio (SNR) at the input of the demodulator for this
setup, given the following parameters: the transmitter uses a transmission power of
LS = 20dBm, the angle between the main antenna direction of the transmitter and the
receiver is = 60 , the radio channel has an attenuation of LA = 122dB and the noise
level at the receiver is given with Ln = 10 log10 (Pn ) = 104dBm.
Hint: cos 0 = 1, cos 6 = 12 3, cos 4 = 12 2, cos 3 = 12 , cos 2 = 0

(c2) The noise power at the receiver is directly related to the bandwidth of the system.
It is often assumed that the noise density is -174dBm/Hz. The density units in the
logarithmic domain have the following relation to the units in the linear domain:
10 log10 (P in mW/Hz) = L in dBm/Hz. Given the above mentioned noise density, what
is the maximum system bandwidth B, if the received power is LE = 98dBm and the
minimum required signal-to-noise-ratio (SNR) is 3dB?
Hint:
x
log10 (x)
0.1 0.25 1 1.5 2

3
5 10 100 1000
1 0.6 0 0.2 0.3 0.5 0.7 1
2
3
Aufgabe 2: Analog-digital conversion

(a) (1 Point)
In analog-digital (AD) conversion, an analog signal s(t) is sampled with 48kHz and quantized
with a resolution of b = 3 Bit. Which data rate is achieved at the output of the AD
conversion?
(b) (4 Points)
The following gure shows the real-valued spectrum S(f ) of the desired signal s(t). The
signal is sampled and stored. Afterwards, it is reconstructed again.
S(f )
1
0,5
-3
-1
f / kHz
(b1) Which minimal frequency fs,min has to be used for sampling in order to reconstruct the
signal s(t) without errors?
(b2) The signal s(t) is fed into the sampler with the frequency fs fs,min . What is the name
of the eect which prevents reconstructing s(t) without errors in this case?
(b3) We assume the sampling frequency fs = 4kHz. Draw the resulting spectrum Ss (f ) of
the sampled signal s(t) in the interval f [7kHz, 7kHz]! Pay attention to correct axis
labeling!
247
D.15 Exam WS 2009/2010
(c) (5 Points)
The following gure presents the quantization characteristics of an analog-digital converter.
s q(t )
twr
sqr
pqr
uqr
vx
vt
vs
vp
vuqr p
s(t )
vpqr
vsqr
vtwr
(c1) The signal s(t) is quantized and sq (t) denotes the output signal of the quantizer. Which
type of quantizer is shown in the gure above? Give reasons for your answer!
(c2) s is the distance between two adjacent quantization steps. Draw the absolute value
of the quantization error |e(t)| as a function of the input signal s(t) in the interval
4s s(t) 4s assuming the given quantization characteristics!
(c3) We assume a constant amplitude density for s(t). The signal is quantized with a resolution of 3 bits. What is the general equation of the signal-to-quantization-noise ratio
SN R occurring with analog-digital conversion? Calculate the corresponding SN R!
248
EXAMS

(a) (4 points)
An analog signal s(t) having a bandwidth of 48 kHz (cut-o frequency fG = 24 kHz) shall
be transmitted over a radio link. You can choose between amplitude modulation (AM) and
frequency modulation (FM). In case of frequency modulation, the peak frequency deviation
is F = 5 kHz.
(a1) The radio channel allows a bandwidth of 50 kHz. Which modulation scheme can be
used (AM, FM or both)? Give reasons!
(a2) Now, consider the case when the bandwidth of the channel does not matter. Which of
the two modulation schemes (AM or FM) is better suited to allow for a high-quality signal transmission that is robust to distance variations between transmitter and receiver?
Give reasons!
(b) (5 points)
The analog signal s(t) = 0.5 + sin(2f1 t) shall be transmitted by means of amplitude modulation, where f1 = 10 MHz. The modulated bandpass signal is sBP (t) = (A+s(t)) cos(2fc t).
(b1) Give an equation for the modulation index !
(b2) Consider a receiver with an envelope detector that has certain tolerances and therefore
requires a modulation index 0.8. Which value of A is then required?
(b3) Calculate the frequency spectrum SBP (f ) of the modulated bandpass signal sBP (t) in
terms of f1 , fc und A! Note: sin(a) cos(b) = 0.5 (sin(a b) + sin(a + b)).
(b4) What is the minimum frequency fc of the carrier signal cos(2fc t) that is required to
transmit an arbitrary signal s(t) with a bandwidth of 10 MHz without distortions?
(c) (2 points)
A modulated single-sideband signal sESB (t) shall be demodulated with a heterodyne receiver.
The gure below shows the magnitude spectrum |SESB (f )| of sESB (t).
|SESB (f )|
2.0
1.0
-1200
-1150
1100
-1100
1150
1200
f / MHz
The next gure shows the block diagram of the heterodyne receiver. The frequency of the
two mixers is fLO = fZF = 650 MHz.
sESB (t )
Bandpass filter
sZF (t )
350 MHz < |f |< 650 MHz

cos(2fLO t )
Lowpass filter
|f |< 250 MHz

cos(2fZF t )
y (t )
249
D.15 Exam WS 2009/2010
(c1) Draw the magnitude spectrum |Y (f )| of the demodulated signal y(t) in the range
400 MHz f 400 MHz! Pay attention to correct axis labeling!
(a) (6 Points) Given is a segment of the time-domain representation of a digitally modulated signal sBP (t) in the bandpass domain, as shown in the following gure. All possible transmitted
symbols occur within this segment.
3
2
BP
(t)
1
2
3
0.5
1.5
2
t/T
2.5
3.5
(a1) Specify the modulation scheme that is used, including the modulation order!
(a2) Draw the corresponding constellation diagram (I/Q-diagram)! Pay attention to correct
axis labeling! Draw a bit assignment into the diagram that satises the Gray mapping
rule! What is the bit sequence that corresponds to the depicted signal segment?
(a3) In general the signal sBP (t) can be written as
sBP (t) = Ts
dk hT (t kTs ) cos(2fc t) .
k=
Specify the impulse response hT (t) of the used transmit lter and the carrier frequency
fc !
(b) (4 Points) Consider now the three modulation schemes (A), (B), and (C) of order four,
having the following constellation diagrams (IQ-diagrams):
(A)
(B)
(C)
A2
A1
A1
A1
A3
A1
A2
A3
A3
A3
250
EXAMS
(b1) Given is the following receiver structure:

cos(2fc t)
rI (t)
kT
rI {k}
hT (t)
dI {k}
rBP (t)
Symbol
Demapper
kT
rQ (t)
rQ {k}
hT (t)
b{n}
dQ {k}
sin(2fc t)
The detectors in the inphase- and quadrature branches make a decision based on the
rule

1
if rI/Q {k} > 0
.
dI/Q {k} =
1 otherwise.
Specify for which of the three modulation schemes (A), (B), and (C) the minimum
possible bit error rate can be achieved with this receiver structure if all transmit symbols
occur with equal probability (multiple selections are possible)! Give reasons for your
answer!
(b2) Assume now that an optimal receiver with ideal decision boundaries is used. For this
case the modulation schemes (A) and (B) shall be compared, given that all transmit
symbols again occur with equal probability.
How large does the amplitude A2 have to be chosen if A1 = 1 and the same bit error
probability is desired for (A) and (B)? Which of the modulation schemes requires in
this case a larger average signal energy ES ? Give reasons for your answer!
(c) (4 Points) Consider now a BPSK transmission, were the constellation appears rotated by
the angle at the receiver due to a phase shift.
Q
For an optimal receiver the bit error rate is given by

"%
#
1
d2
BER = erfc
,
2
8 2
where 2 denotes the noise variance and the complementary error function erfc(x) is given in
the gure below. But we only have access to a conventional BPSK receiver, which is designed
for the case = 0 and unable to perform a phase correction.
251
D.15 Exam WS 2009/2010
(c1) Draw the decision boundary of the receiver designed for = 0 into the depicted constellation diagram!
(c2) Give an expression for the resulting bit error rate for /2 < < /2!
(c3) How large does d have to be chosen such that a bit error rate of 0.5% is achieved for
= 45 and = 0.5?
10
10
erfc(x)
10
10
10
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
252
D.16
EXAMS
Problem 1: LTI-Systems and Signal Theory

(a) (a1) Y (f ) = X(f ) H(f )
(1 Point)
(a2) The system is causal, because h(t) = 0 for t < 0.

(a3) H(f ) = F {h(t)} = F rect Tt 12
(1 Point)
From the correspondance table: F {rect(t)} = si(f )

= T si(T f )
Applying the scaling property: F rect Tt
&
T '
t
= T si(T f )ejf T
With the time shifting property: H(f ) = F rect T 2
(1 Point)
Drawing as follows:
|H(f)|
0
-3/T
-2/T
-1/T
0
f
1/T
2/T
Drawing qualitatively correct:

Correct axis labeling:
3/T
(1 Point)
(1 Point)
(a4) Aside from the trivial case when both signals do not overlap, there are two cases to
distinguish to calculate the convolution. They are depicted in the sketch below
-T
t
Case 1
t-T
0
Case 2
253
Calculation for Case 1:

y(t) =
T
A 2
A
A
+ A d =
t + At T
T
2T
2
Calculation for Case 2:

y(t) =
tT
A
A
A
+ A d = t2 + T
T
2T
2
Thus, it follows:
A
A 2
2T t + At 2 T
y(t) =
A t2
2T
A
T
2
T t < 0
0t<T
otherwise
Cases correctly distinguished:

Correct result for Case 1:
Correct result for Case 2:
(1 Point)
(1 Point)
(1 Point)
(b) (b1) PS = 40W

Correct result and unit:
(1 Point)
(b2) LA = 20dB + 10 log10 (2504 )dB = 20dB + 4 24dB = 116dB

Channel attenuation correct:
LE = LS LA + Lg = 43dBm 116dB + 20dB = 53dBm
(1 Point)
(1 Point)
(c) (c1) Lant = 24 cos 3 dB = 12dB

Directional antenna amplication toward user correct:
(1 Point)
SN R = LS + Lant LA Ln = 20dBm + 12dB 122dB (104dBm) = 14dB
(1 Point)
(c2) Ln = LE SN R = 98dBm 3dB = 101dBm
Correct noise level:
It follows: B = 101dBm (174dBm/Hz) = 73dB Hz = 20MHz
(1 Point)
(1 Point)
Aufgabe 2: Analog-digital Conversion

(a) Data rate with analog-digital conversion: 144 Kbps
(1 Point)
(b) (b1) Nyquist criterium: fs,min > 2fg = 6kHz
(1 Point)
254
EXAMS
(b2) Aliasing
(1 Point)
(b3) Drawing as follows

Diagram qualitatively correct - S(f ) periodically continued
(1 Point)
(1 Point)
S(f )
2
1,5
1
-7
-5
-3
-1
f / kHz

(c) (c1) Mid-rise quantizer sq (t) s
(1, 3, 5, ...q 1) because the quantization charac2
teristic does not include the quantization step sq (t) = 0
(1 Point)
(c2) | e(t) |=| sq (t) s(t) | as function of s(t) corresponds to following gure
|e(t)|
yz{
|
~ |
} |

(c3) SN R = 6.02b dB
SN R = 6.02dB/bit 3bit = 18.06 dB
} |
~ |
|
s(t )
(1 Point)
(1 Point)
(1 Point)
(1 Point)

(a) (a1) AM
(1 point)
Reasons:
(1 point)
The bandwidth of the AM signals is only 48 kHz, i.e. less than 50 kHz. It
does not exceed the bandwidth of the radio channel. The bandwidth of the FM
signal derives from carson rule as B + 2 F = 58 kHz. Hence, FM would exceed the
bandwidth of the radio channel.
(a2) FM
(1 point)
Reasons:
(1 point)
AM modulates the amplitude of the carrier signal. The pathloss of the radio
channel causes an attenuation that depends on the distance between transmitter and
the receiver and hence eects the quality of an AM signal. FM modulates frequency
of the carrier signal, which is not aected by the pathloss.
255
(b) (b1) =
max |s(t)|
A
(1 point)
15
= 1.875.
(1 point)
8
(2 points)
(b3) SBP (f ) = (0.5 A + 0.25) ((f + fc ) + (f fc ))
+ 0.25 j ((f + f1 fc ) (f f1 + fc ) + (f + f1 + fc ) (f f1 fc ))
(b2) For 0.8 follows with max |s(t)| = 1.5: A
(b4) fc 5 MHz
(1 point)
(c) (c1) Magnitude spectrum |Y (f )| - see gure below.

(1 point)
(1 point)
|Y (f )|
0.50
0.25
-200
-150
100
-100
150
200
f / MHz

(a) (a1) 4-ASK
(1 Point)
(a2)
11
01
Q
00
10
3
The gure shows an example assignment that follows the Gray mapping rule. In total
there exist the following correct assignments:
dk = 3 dk = 1 dk = 1 dk = 3
00
01
10
11
10
01
11
00
00
10
11
01
00
01
11
10
11
01
00
10
10
00
01
11
01
11
10
00
11
01
00
10
From the depicted signal segment follows: d0 = 1, d1 = 3, d2 = 1 and d3 = 3. For
the example assignment the bit sequence is equal to 0, 0, 1, 0, 0, 1, 1, 1
Constellation diagram correct:
(1 Point)
256
Bit assignment correct (Gray mapping):

Bit sequence correct:
EXAMS
(1 Point)
(1 Point)
(a3) hT (t) = 1/Ts rect(t/Ts 1/2)
(1 Point)
fc = 3/Ts
(1 Point)
(b) (b1) The receiver is only optimal for (C). In case (A), I and Q cannot be decided independently. In case (B), a threshold equal to A2 /2 is required in the detectors.
Correct choice:
Explanation correct:
(1 Point)
(1 Point)
(b2) For equal bit error rate

constellation points must be
the possible distances between

equal. Thus, we have 2A1 = A2 and hence A2 = 2.
(1 Point)
Average energy:
(A) ES =
4A21
4
=1
(B) ES =
0+A22 +A22 +2A22

4
= A22 = 2
for equal bit error rate a larger (the double) average energy is required in case (B).
(1 Point)
257
(c) (c1) Decision boundary:
(1 Point)
Q
(c2) The receiver uses the

part of the signal, which reduces the distance to d cos
(inphase
2
(d cos )
.
(2 Points)
BER = 12 erfc
8 2
(c3) We have: (d cos
)2 = d2 /2

0.01 = erfc
d2 /(2 2) = erfc (d/2)
From the curve we have: d/2 1.8 d 3.6
(1 Point)
258
D.17
EXAMS
Exam SS 2010

(a) (3 points)
A discrete time system is dened by
y[n] = 0.5x[n] - 0.25 n x[n 1] ,
where n is the discrete time index.
Answer the following questions with reason!
(a2) Is the system time-invariant?
(a3) Is the system causal?
(b) (6 points)
An LTI system, as described below in the gure
x(t)
h(t)
y(t)
has an output of y(t) = rect( 2t ) when an impulse, x(t) = (t) is applied at the input.
(b1) Determine the frequency response H(f ) of the system! Also sketch the waveform of
H(f )!
(b2) Find an analytical expression and sketch the output signal y(t), when the input signal
is x(t)=rect(t)!
(b3) Determine the frequency spectrum of the output signal y(t), when the input signal is
x(t)=rect(t)!
259
D.17 Exam SS 2010
(c) (6 points)
The gure shows the block diagram of a radio transmission link between a mobile and a
base station.
5DGLR &KDQQHO
0RELOH
$PSOLILHU
*DLQ G%
/5
/7
/$
$
37
'HPRGXODWRU
35
31 :
31
:

LA = 100dB + 10 log10 (d/km)4 dB
(c1) The transmit power of the mobile is 1W. Calculate the received power, PR in Watt at
the base station located at a distance of d = 1000m!
(c2) Calculate the Signal to Noise Ratio (SNR) in dB at point A, the input of the amplier!
(c3) The minimum SNR required at point B, the input of the demodulator is 10dB. Calculate
the maximum allowable distance between the mobile and the base station!
Note:
x
log10 (x)
0.1 1 1.5 2
3 10 100 1000
1 0 0.2 0.3 0.5 1
2
3
Problem 2: Analog-digital conversion

Analog-digital (AD) converters transform analog signals to digital signals. Following gure shows
the connection.
Analog
signal
s(t)
Sampling
ss(t)
Quantisation
sq(t)
Digital
signal
Analog-digital converter
(a) (2 Points)
260
EXAMS
(a1) Which sampling frequency fS is neccessary to reconstruct the original signal s(t) with
fg = 2kHz from the sampling values without errors. Motivate the answer!
(a2) We consider the AD conversion of the signal s(t). Quantisation is done with
1024 quantisation steps. Is it possible to reconstruct the original signal s(t) from
the resulting signal sq (t) without errors? Motivate the answer!
(b) (4 Points)
S(f ) indicates the spectrum of the analog signal s(t). S(f ) is dened by the following function
with fg = 1kHz.

f
.
S(f ) = rect
2fg
s(t) is sampled with the sampling frequency fS = 3kHz.
(b1) Sketch the spectrum of the sampled signal ss (t) in the interval [4kHz, 4kHz]!
(b2) Now, we consider the single carrier signal s(t) = A cos(2f t) with A = 1 and f = 2kHz.
Draw the spectrum of the sampled signal ss (t) in the range [4kHz, 4kHz]!
(c) (4 Points)
We consider s(t) = sin(2 f4S t) as input signal of the AD conversion and the
sampling frequency fS = T1S . Here, two alternative quantisers are available. ss (t) denotes
the sampled signal before the quantisation.
(c1) The quantisation of ss (t) is applied by a mid-rise quantiser at a resolution of 4 bits, a
step width of s = 50mV and a symmetric recording level [Umax , Umax ]. Calculate the
width of the recording level at the input of the quantiser and the maximum absolute
value of the quantization error emax !
(c2) The second quantiser has following quantisation characteristic q().
1 for ss (t) > 0

q(ss (t)) = 0 for ss (t) = 0
1 for ss (t) < 0 .

Draw the output signal sq (t) = q(ss (t)) for 0 t 6TS ! Make sure you label the axes
of the diagram correctly!
261
D.17 Exam SS 2010
Task 3: Analog Modulation

(a) (3 Points)
(a1) Explain the term analog modulation!
(a2) Give two reasons why it is advantageous to transmit modulated carriers compared to
base band transmission!
(b) (8 Points)
An analog signal sQ (t) = cos(2f1 t) + cos(8f1 t) shall be transmitted through a carrier
sc (t) = A cos(2f
c t) by amplitude modulation. The amplitude modulated signal is given

as sAM (t) = A + sQ (t) cos(2fc t) .
(b1) Determine the modulation index of the amplitude modulated signal for A = 2.5 and
A = 0.4! Reason whether or not the signal can be demodulated correctly by an envelope
demodulator!
(b2) Compute the square of the amplitude modulated signal s(t) = s2AM (t) for A = 1! Compute the spectrum of s(t). Sketch the spectrum for 8f1 f 8f1 ! Assume fc f1 .

Hint: cos2 (x) = 21 cos(2x) + 1
In the following, the analog signal sQ (t) shall modulate the phase of the carrier sc (t) instead
of its amplitude. The phase modulated carrier shall be denoted by sPM (t) .
(b3) Give an equation for the phase modulated carrier sPM (t)!
(b4) Determine an equation for the frequency of the phase modulated carrier sPM (t)!
A model pilot wants to develop a remote control system for his newest plane. He decides to use a
digital data transmission method. The remote control transmits data packets to the model.
(a) (4 points)
A channel with a bandwidth B = 10kHz is available for the transmission. One packet has a
size of 25 bytes.
(a1) State the general relationship between symbol duration T and bandwidth B for linear
modulation!
The bandwidth of a transmit pulse shall now be given as B =
2
.
T
(a2) How many packets can be transmitted per second when on-o-keying (OOK) is used?
262
EXAMS
(a3) Tests have shown that at least 80 packets per second are necessary for accurate control.
However, the channel bandwidth cannot be increased. Is there a modulation scheme
that fullls these requirements? If yes, give the name of the method and sketch its
constellation diagram!
(b) (3 points)
After implementing a better compression method, its now sucient to use a two-valued
modulation scheme. The following gure shows a possible transmitter for a BPSK-signal.

D
(b1) Draw a brief sketch of a suitable receiver for this modulation scheme from the antenna
to the estimated bitstream!
(b2) Is your receiver able to compensate a random phase rotation that is caused by the
channel? State a reason and explain the required enhancements if not!
(c) (7 points)
The data is now modulated using BPSK with the symbols s0 and s1 . Both symbols are
sent with equal probability. The signal is distorted by additive white Gaussian noise with
variance 2 . The decision boundary of the receiver is D. It is then possible to calculate the
bit error rate using the following equation:
1
BER = erfc
4
s0 D
2 2

s1 D
1
+ erfc
.
4
2 2
The complementary error function erfc() is shown on the next page. Furthermore, s0 =
s1 = 4mV, 2 = 2(mV)2 and D = 0mV are given.
(c1) Give one advantage of BPSK over OOK!
(c2) Calculate the bit error rate of the receiver!
(c3) State a reason whether or not the decision boundary D = 0mV minimizes the bit error
rate!
Due to tolerances of the electrical components the values of all received symbols are shifted
by a constant value c given in [mV] with 0mV c < s0 . Assume that the receiver algorithms
are not aware of this shift and therefore they dont compensate it.
Hint: c < s0 implies that the symbols are not shifted over the desicion boundary.
(c4) Extend the equation for the bit error rate by the inuence of c!
263
D.17 Exam SS 2010
(c5) The error correction capabilities of the system only work well if the bit error rate does
not exceed BER = 4 102 . What is the maximum value of c such that this limit is not
exceeded?
Hint: The smaller of the two erfc() terms can be omitted for c > 1mV.
Complementry Error Function
10
10
y=erfc(x)
10
10
10
10
0.5
264
D.18
EXAMS

(a) (a1) For two input sequences x1 [n] and x2 [n], the outputs are
y1 [n] = 0.5x1 [n] - 0.25 n x1 [n 1]
y2 [n] = 0.5x2 [n] - 0.25 n x2 [n 1] .
A linear combination of the two input sequences results in the output
y3 [n] =T [a1 x1 [n] + a2 x2 [n]]
y3 [n] =0.5a1 x1 [n] - 0.25a1 nx1 [n 1] + 0.5a2 x2 [n] - 0.25a2 nx2 [n 1] .
Again, the linear combination of the two outputs results in
a1 y1 [n] + a2 y2 [n] = 0.5a1 x1 [n] - 0.25a1 nx1 [n 1] + 0.5a2 x2 [n] - 0.25a2 nx2 [n 1] .
From the above two equations, we can say that the system is linear.
(1 Point)
(a2) The response of the system to x[n k] is

y[n, k] = 0.5x[n] - 0.25nx[n k 1] .
Now, if we delay y[n] by k units in time, we obtain
y[n k] = 0.5x[n k] - 0.25(n k)x[n k 1] .
Hence from the above we see that the system is NOT time invariant.
(1 Point)
(a3) The system is causal, as the present output depends only on the present input and
past inputs and not on any future inputs.
(1 Point)
(b) (b1) The impulse response is

h(t) = rect( 2t ) .
The Fourier transform yields
1

H(f ) = rect( 2t )ej2f t dt = 1 1 ej2f t dt
H(f ) =
sin(2f )
f
= 2 si(2f )
(1 Point)
265
+I

PDJQLWXGH

SL I
(1 Point)
(b2) The output signal is given by
y(t) = h(t) x(t) =
rect( 2 )rect(t )d
0
t 32
t+1/2
3
3
1
1 d = 2 + t 2 t 2
1
y(t) =
d = 1
12 t 12 .
0
d = 32 t 21 t 33
t1/2
0
3
t
2
(2 Points)
\W

W
(1 Point)
(b3) The spectrum of the output is given as

266
D
sin(2f )
f
Y (f ) = H(f ) X(f ) =
sin(f )
f
sin(f ) sin(2f )
2 f 2
EXAMS
.
(1 Point)
(c) (c1) The transmit power is 1 W, which corresponds to 30 dBm

LA = 100 + 10 log10 (1)4 = 100 + 40(0) = 100 dB.
The received power level at the receiver is
LR = LT LA = 30 dBm100 dB= 70 dBm.
The received power in Watt is
PR = 1010 W .
(2 Points)
(c2) The SNR at the input of the amplier is given as
SN R =
PR
PN 1
1010 W
1014 W
= 104 = 40 dB.
(1 Point)
(c3) The SNR at the output of the demodulator is 10 dB.

Therefore the signal power is
10 PB = 10(gPN 1 + PN 2 ) = 1011 W.
The signal power before the amplier is
1011
10
= 1012 W = 109 mW.
The signal power level before the amplier is 90dBm.

If the minimum distance is x, then the signal power at the receiver before the amplier
is
LR = 30 (100 + 40 log10 (x)) = 70 40 log10 (x) .
Therefore we have
70 40 log10 (x) = 90
20
log10 (x) = 40
= 0.5
x = 3.
(2 Points) for correct derivation
The maximum allowable distance between the mobile and the base station is 3 km.
(1 Point) for the correct value
267
Problem 2: Analog-digital conversion

(a) (a1) The original signal s(t) can be reconstructed from the sampling values without errors
if the sampling follows fS > 2fg (sampling theorem).
(1 Point)
(a2) Quantisation is always lossy. Therefore it is not possible to reconstruct s(t) from sq (t)
without errors.
(1 Point)
(b) (b1) Spectrum | Ss (f ) | is the periodic continuation of the original spectrum S(f).
Diagram qualitatively correct - (| Ss (f ) |) periodically continued
(1 Point)
(1 Point)
|S s(f )|
1
-3 -2 -1
-4
4 f / kHz
(b2) Spectrum | Ss (f ) | according to the gure below.

Diagram qualitatively correct - (aliasing)
(1 Point)
(1 Point)
|S s(f )|
1
-3 -2 -1
-4
4 f / kHz
(c) (c1) Assuming the maximum absolute quantisation error emax , the width of the recording
level 2U , the step width s and a resolution in bit b, calculation results in
q = 2b = 16
2U
q = s
2U = q s 2U = 16 50mV = 800mV
emax = s
emax = 25mV .
2
(1 Point)
(1 Point)
(c2) Signal sq (t) according to the gure below.

(1 Point)
(1 Point)
s q(t )
1
0
t / TS
-1
268
EXAMS

(a) (a1) Analog modulation is understood as imposing an analog source signal on a carrier
signal.
(1 Point)
(a2) Reasons for transmitting modulated carriers are
(2 Points)
Adjustment of the source signal into the transmitting medium

Possibility of transmission of multiple signals over the same medium (frequency
multiplexing).
(b) (b1) The modulation index is dened as =

(1 Point) for two correct values
max |sQ |
.
A
For A = 2.5 and A = 0.4 we have
2
4
=
5
5
5
2 = 2 = 5 .
2
1 = 2
Since the information is contained not only in the amplitude but also in the phase of
the carrier if > 1, an envelope detector cannot be used in the second case where
= 5.
(1 Point)
(b2) The squared signal s(t) = s2AM (t) for A = 1 is given as

s(t) = a(t) 1 + cos(4fc t)
where
)
*
1
1
a(t) = 1+cos(2f1 t)+cos(8f1 t)+cos(2f1 t) cos(8f1 t)+ cos(4f1 t)+ cos(16f1 t) .
4
4
(1 Point)
) we obtain
For the spectrum S(f
) = A(f ) + A(f 2fc ) + A(f + 2fc )
S(f
where
1
(f f1 ) + (f + f1 ) + (f 4f1 ) + (f + 4f1 ) +
2
1
+ (f 3f1 ) + (f + 3f1 ) + (f 5f1 ) + (f + 5f1 ) +
4
1
+ (f 2f1 ) + (f + 2f1 ) + (f 8f1 ) + (f + 8f1 ) .
8
A(f ) = (f ) +
269

S(f
)
1
2
1
4
1
8
8f1
5f14f13f12f1 f1
f1 2f1 3f1 4f1 5f1
8f1
Figure D.8: Amplitude spectrum of the signal s2AM (t) in the range 8f1 f 8f1
(2 Points)
Since fc f1 we do not need to consider the terms A(f 2fc ) and A(f + 2fc ) in the
sketch. The spectrum of the signal in the range 8f1 f 8f1 is depicted in Figure
D.8.
(1 Point) if drawing corresponds to the spectrum equation obtained above (even if
equation is not correct), correct axes, correct range.
(b3) In case of a phase modulation the modulated signal is given as
(1 Point)

sPM (t) = A cos 2fc t + kPM sQ (t) + 0

= A cos 2fc t + kPM cos(2f1 t) + cos(8f1 t) + 0 .

(b4) With the phase of the modulated signal (t) = 2fc t+kP M cos(2f1 t)+cos(8f1 t) +0
the frequency writes as
(1 Point)
f (t) =
1 d(t)
= fc + kP M (f1 sin(2f1 t) + 4f1 sin(8f1 t)).
2 dt

(a) (a1) B
1
T
(1 Point)
270
EXAMS
(a2) With M = 1 bit per symbol, the bit rate R is given as:
(1 Point)
M
M B
R=
=
= 5kbit s1
T
2
With 25 8 = 200 bits per packet it is possible to transfer 5000/200 = 25 packets per
second!
(a3) The data rate must be increased by factor 80/25 = 3.2. Quadrupling the rate yields the
closest result. Possible modulation schemes are therefore: 16-PSK, 16-QAM or schemes
with even more bits per symbol.
(1 Point)
Y

/

Diagram correct:
(1 Point)
(b) (b1) Valid receivers: I/Q-receiver, receiver with only I-branch

Block diagram correct
Labels correct

^

D
(1 Point)
(1 Point)
D
&

d

(b2) In case of an I/Q-receiver:

The phase rotation can be estimated and mitigated using a complex multiplication in
the digital baseband
For receiver with only an I-branch:
No, for growing phase rotation, the resulting signal power will decrease. If the rotation
reaches 90 the signal power becomes zero and no detection is possible. The problem
can be solved by adding a Q-branch which transforms the receiver to an I/Q-receiver.
Explanation correct
(1 Point)
(c) (c1) Possible reasons:
Smaller BER with the same transmit power
Smaller transmit power while retaining the BER
(1 Point)
271

1
1
4mV 0mV
4mV 0mV
1
+ erfc
= erfc(2)
(c2) BER = erfc
2
2
4
4(mV)
4
4(mV)
2
erfc(2) = 4, 7 103 (Correct range: [4 . . . 5] 103 )
BER = 2, 3 103 (Correct range: [2 . . . 2, 5] 103 )
(1 Point)
(1 Point)
(c3) Yes, the distance to s0 and s1 is maximized and equal. Therefore, the BER is minimized!
(1 Point)

1
s0 + c D
s1 + c D
1
+ erfc
(1 Point)
(c4) BER = erfc
4
4
2 2
2 2

s1 + c
4mV c
1
1
=
(c5) Using the approximation we get: BER
erfc
erfc
4
4
2mV
2 2
(1 Point)
erfc(x) = 4 4 102 = 1, 6 101 at roughly x = 1
4mV c
= 1 c = 2mV
2mV
The maximum value for c is 2mV!
(1 Point)
272
D.19
EXAMS
Exam WS 2010/2011

(a) (5 points)
Given the following signal
x(t) =
5

(1)n (n 1) (t n)
n=5

(a1) Draw x(t)! Pay attention to correct axis labeling!
(a2) Can the signal x(t) be a valid impulse response of a causal system?
(a3) Is the signal x(t) odd or even?
(a4) Is the spectrum X(f ) of x(t) real valued?
(b) (5 points)
Two LTI systems with impulse responses h1 (t) =rect(t + 21 ) and h2 (t) = 2rect(t) are serially
connected, as shown in the gure below, with overall impulse response h(t).
h(t) = h1 (t) h2 (t)

(b1) Draw the impulse response of the overall system! Pay attention to correct axis labeling!
(b2) Determine the transfer function H(f ) of the system!
(b3) Draw the output signal y(t) when the input signal x(t) is
x(t) = (t) + (t 1) 0.5(t 2) + 1.5(t 3)
(c) (5 points)
base station.
273
D.19 Exam WS 2010/2011
5DGLR &KDQQHO
/7 G%P
$PSOLILHU *DLQ
J G%
$PSOLILHU *DLQ
J G%
/5
/6
$
0RELOH
'HPRGXODWRU
G%P
/1

LS = 20 log10
100d
m

dB,
where d is the distance between the transmitter and receiver in meter.

(c1) Calculate the maximum distance, d that ensures a minimum signal to noise ratio (SNR)
of 5dB at point B, the input of the demodulator!
(c2) Calculate the minimum power level LT (in dBm) necessary, in order to reach a SNR of
20dB at point B, at a distance of d = 50m!
Note:
x
log10 (x)
0.1 1 1.5 2
3 10 100 1000
1 0 0.2 0.3 0.5 1
2
3

In general, digital signals are stored after the analog-digital (A/D) conversion in a digital memory,
e.g. the main memory. In order to convert digital signals back to analog signals, digital-analog
(D/A) converters are needed. This relationship is illustrated in the gure below.

(a) (2 points)
(a1) We consider x (t) the recovered, analog signal after the sampling and quantization.
Why is an errorless reconstruction of the analog signal x(t) not possible even assuming
ideally working components?
274
EXAMS
(a2) Name two advantages of digital signals over analog signals!
(b) (4 points)
(b1) x1 (t) denotes a bandwidth-limited, analog signal, as shown in the gure below.
In the sampling, x1 (t) is multiplied by the Dirac-comb function XT (t).
XT (t) =
(t nT ).
n=
The resulting sampled data sequence xs1 (t) results from xs1 (t) = x1 (t)XT (t).
Draw xs1 (t) in the region 4T t 4T ! Pay attention to correct axis labeling!
275
D.19 Exam WS 2010/2011

In sampling x2 (t), the sampling frequency fs2 = 5kHz is used. Draw the amplitude
spectrum of the sampled signal xs2 (t) in the interval f [0kHz, fs2 ]! Name and highlight
the eect caused by the low sampling frequency! Pay attention to correct axis labeling!
(c) (4 points) Now, we consider a quantizer with distorted (biased) quantization characteristic,
shown in the gure below. s denotes the step width of the quantizer.
q
biased
(c1) How large is the dynamic range 2U of the quantizer with s = 50mV? Determine the
width of the dynamic range at the input of the quantizer and the resolution b of the
quantizer in bit!
(c2) The error signal e(t) = x(t) xq (t) is the deviation of the quantized signal xq (t)
from the original signal x(t). Draw e(t) using the quantizer shown above in the region
4s t 4s! Pay attention to correct axis labeling!
276
EXAMS

A signal sQ (t) = AQ sinc (t) is transmitted by a carrier c(t) = Ac cos(2fc t) via amplitude modulation. The amplitude modulated signal is given as

s1 (t) = Ac + sQ (t) cos(2fc1 t).
(a) (2 points)
(a1) Calculate the modulation index of s1 (t)!
(a2) Specify the range of the amplitude AQ , for which the signal s1 (t) exhibits no phase
jumps!
(b) (4 points)
The receiver consists of the components depicted in Figure D.9, where a band pass lter
and mixer are used for channel selection. In addition to s1 (t), the signal s2 (t) = (Ac +
sQ (t)) cos(2fc2 t) from a neighboring channel is received at the antenna and the sum s(t) =
s1 (t) + s2 (t) is fed into the mixer. In the following we assume AQ = Ac = 1, fc1 + 21 Hz <
fM < fc2 12 Hz.
(b1) Explain briey the purpose of the low pass lter and the high pass lter in the demodulator in Figure D.9!
(b2) Sketch the amplitude spectrum |S(f )| of the signal s(t) at the mixer input for f > 0!
(b3) Sketch the amplitude spectrum |S (f )| of the signal s (t) at the mixer output for f > 0!
(c) (5 points)
Consider the band pass lter in Figure D.9 as being ideal with a bandwidth of 1Hz and denote
its center frequency by fbp . For a certain range of the mixer frequency fM it is possible that
the signals s1 (t) and s2 (t) spectrally overlap after the bandpass lter.
(c1) State the conditions on the mixer frequency fM such that s1 (t) and s2 (t) do NOT
spectrally overlap after the band pass lter! Distinguish the cases fbp = fM fc1 and
fbp = fM + fc1 . Also take into account the bandwidth of the signals s1 (t) and s2 (t)!
(c2) State another simple signal processing technique at the receiver to avoid spectral overlap
of s1 (t) and s2 (t) after mixing!

(a) (6 points)
The following gure shows the plots of three dierent digitally modulated transmit signals.
All possible transmit symbols are shown in each plot
277
D.19 Exam WS 2010/2011
W
>W
,W
Figure D.9: Receiver block diagram
2
t/Ts
y/A
y/A
y/A
2
t/Ts
2
t/Ts
(a1) Which digital modulation schemes are shown? Give the names and the valency!
(a2) A radio system shall transmit a video stream with a data rate of 768kbit/s. The symbol
duration of the system is xed at Ts = 5s. Is there a modulation scheme that achieves
the required data rate with the given symbol duration? Give the name of the scheme
and the resulting maximum data rate of the system!
(a3) Sketch the constellation diagrams of QPSK and 8-PSK!
The following gure shows a transmitter that is to be used for transmission of digital data.
Two incoming information bits are assigned to one baseband symbol and all possible symbols
occur with equal probability. A lter with rectangular impulse response is used for pulse
shaping such that the kth symbol is transmitted in the time interval [k Ts , (k + 1) Ts ]. Ts
denotes the symbol duration.

(b) (4 points)
(b1) Draw a brief sketch of a suitable receiver for this modulation scheme and give the names
of all components!
278
EXAMS
(b2) The data sequence {01, 10, 00, 11} shall be transmitted. fc = 1kHz and Ts = 2ms are
given. Draw the resulting signal x(t) in the interval 0 t 4Ts ! Pay attention to
279
D.19 Exam WS 2010/2011
(c) (4 points)
The signal r(k) at the input of the detector can be written as follows:
r(k) = s(k) + n(k)
n(k) is white Gaussian noise with variance 2 . The resulting bit error rate is then given by
the following formula:
3
BER = erfc
8
"%
A2
8 2
Furthermore, the average symbol energy is Es = E {|s(k)|2 } and the signal-to-noise ratio is
SN R = Es / 2 .
(c1) The system allows for dynamic adaptation of the parameter A to the current transmission conditions. An attached error protection scheme (not shown) is able to compensate
a maximum bit error rate of 0.375 102 . On the other hand, choosing a very large A
might be a waste of transmit power. Let the noise variance be 2 = 0.5. Calculate the
smallest possible A for which the required bit error rate is reached!
(c2) When using the system in a dierent scenario you calculate that A = 1 is required for
reliable operation. For this value, the average symbol energy (according to the formula
given above) is Es = 3.5. However, regulatory conditions state an upper limit of Es 2.
The system allows you to change the mapping between bits and symbols in the mapper.
However, you are neither allowed to change the data throughput nor the error tolerance
of the system. Give a new mapping that fullls all these requirements! Give the average
symbol energy for the new mapping for A = 1!
280
Complementry Error Function
10
10
y=erfc(x)
10
10
10
10
0.5
2
x
EXAMS
281
D.20

(a) (a1) The plot of the signal is as follows
Shape correct
Axis labels correct
(1 point)
(1 point)
(a2) No, the system would not be a causal system as x(t) = 0 for t < 0 is not satised.
(1 point)
(a3) The signal is neither odd nor even, as x(t) = x(t) and also x(t) = x(t). (1 point)
(a4) No, X(f ) is complex valued as x(t) is neither odd nor even.
(1 point)
(b) (b1) The overall impulse response of the system is
282
Shape correct
Axis labels correct
EXAMS
(1 point)
(1 point)
(b2) From
h(t) = rect(t + 1/2) 2 rect(t)
follows
H(f ) = 2ejf si(f ) si(f )
(1 point)
(b3) The output signal is
Shape correct
Axis labels correct
(1 point)
(1 point)
(c) (c1) The noise power level at the input of the demodulator is
LN,dem = LN + g2 = 50dBm + 60dB = 10dBm
The signal power level at the input of the demodulator is
LT,dem = LT LS + g1 + g2 = 15dBm 20 log10 (100d/m) dB + 20dB + 60dB
LT,dem = 95dBm 20 log10 (100d/m) dB
Correct equations
The SNR at the input of the demodulator is
SNR = LT,dem LN,dem 5dB
95dBm 20 log10 (100d/m) dB 10dBm 5dB
log10 (100d/m) 4
d 100m
(1 point)
283
Therefore maximum distance d=100m.

Correct equations till the penultimate step
Correct value
(1 point)
(1 point)
(c2) Solving for LT

SNR LT LS + g1 + g2 LN g2
LT SNR + LS g1 + LN
LT 20dB + 20 log10 (5000) dB 20dB 50dBm
LT 24dBm
Therefore the minimum transmitter power level LT = 24dBm.
Correct equations
Correct value
(1 point)
(1 point)

(a) (a1) Quantization noise (&saturation).
(1 point)
(a2) Digital signals are easily stored, technology-and temperature-independent.
(1 point)
(b) (b1) The sampled data sequence xs1 (t) preserves the shape of the corresponding analog
signal in principle.
Qualitatively correct: xs1 (t)
(1 point)
Quantitatively correct: (axis label)
(1 point)
(b2) Amplitude spectrum | Xs2 (f ) | according to gure.

Qualitatively correct: (aliasing visible)
Quantitatively correct: (axis label)

(1 point)
(1 point)
Aliasing
kHz
284
EXAMS
(c) (c1) With the width of the dynamic range 2U , the number of quantization steps q, the step
width s and the resolution b in bit follows
q = 7,
b = ld q = 3,
2U = q s 2U = 7 50mV = 350mV.
(1 point)
(1 point)
(c2) Error signal e(t) according to gure (shifted with s/2).

Qualitatively correct: (Curve)
Quantitatively correct: (Axis label)
(1 point)
(1 point)

(a) (a1) The modulation index is dened as =
A
= AQc . (1 point)
max(sQ (t))
.
Ac
For sQ (t) = AQ sinc (t) we get
(a2) It is required that 1, which implies AQ Ac . (1 point)
(b) (b1) Low pass lter: Reconstruction of the envelope. (1 point)

High pass lter: Removal of DC component. (1 point)
(b2) See Figure D.10 (1 point)
(b3) See Figure D.11 (1 point)
(c) (c1) To avoid spectral overlap for fbp = fM fc1 and signal bandwidth 1 Hz it is required
that
1
1
fc2 fM + Hz < fM fc1 Hz
2
2
1
fM > (fc2 + fc1 + 1Hz)
2
or
or
1
1
fc2 fM Hz > fM fc1 + Hz
2
2
1
fM < (fc2 + fc1 1Hz)
2
285
(1 point)
(1 point)
To avoid spectral overlap in case of fbp = fM + fc1 and signal bandwidth 1 Hz we need
1
1
fc2 fM + Hz < fM + fc1 Hz
2
2
1
fM > (fc2 fc1 + 1Hz)
2
(1 point)
or
or
1
1
fc2 fM Hz > fM + fc1 + Hz
2
2
1
fM < (fc2 fc1 1Hz)
2
(1 point)
Note, that fM can only fulll one of the conditions for each case, i.e., it is either greater
or smaller than the invalid frequency range.
(c2) A low pass lter or band pass lter with cuto frequency less than fc2
the mixer would remove s2 (t) and prevent spectral overlap. (1 point)
,
1
2
Hz before
,
Figure D.10: Amplitude spectrum |S(f )|

(a) (a1) QPSK, OOK, 3-FSK
Two correct: 1 point, All correct: 2 points
(2 points)
(a2) fs = 200kHz, i.e. at least 4 bits per symbol (e.g. 16-QAM, 16-PSK)
Reachable rate: 800kbit/s
(a3) Arbitrary phase rotations also correct
(1 point)
(1 point)
(2 points)
286
EXAMS
Figure D.11: Amplitude spectrum |S (f )|
QPSK
8PSK
1
0
I
0
I
(b) (b1) Drawing as follows, alternatively I/Q-receiver, also correct without demapper

!
Correct drawing
Correct labeling
(b2) Drawing as follows
"
$#

$#

"
(1 point)
(1 point)
287
x(t)
3
2
y/A
1
0
1
2
3
2
t/T
Correct plot
Correct labeling
(c) (c1)
(1 point)
(1 point)
"%
#
2
3
3
A
0.375 102 = 102 = erfc
8
8
8 2
"%
#
A2
102 = erfc
8 2
%
A2
= 1.8
8 2
Correctly read from plot (correct: 1.7 to 1.9)
(1 point)
A = 1.8 2 2
A = 1.8 2 = 3.6
Correct result (correct: 3.4 to 3.8)
(1 Punkt)
(c2) In order to preserve both data throughput and error tolerance, neither the amount nor
the distance between the symbols may change. Since the transmitter does not have
a Q-branch, the constellation cannot be changed to QPSK. However, if all symbols
are moved along the I-axis by 1.5A, all properties are preserved. This results in the
following mapping:
00 1.5A
01 0.5A
11 0.5A
10 1.5A
Correct mapping (also correct without A)
The average symbol energy for A = 1 is:
Es =
(1 point)
(1.5)2 + (0.5)2 + 0.52 + 1.52

= 1.125
4
288
Correct Es given
EXAMS
(1 point)
289
D.21 Exam SS 2011
D.21
Exam SS 2011

Note: Questions (a), (b) and (c) can be solved independently of each other.
(a) (4 points)
A discrete time system with input signal x[n] and output signal y[n] is dened as
y[n] = 0.5x2 [n] + 0.3x[n + 1],
(a4) What is y[n] if x[n] = [n] + [n 1]?
Note: The digital unit pulse [n] is given as:

1 for n = 0
[n] =
0 otherwise.
(b) (6 points)
An LTI system, as shown in the gure below, has an impulse response h(t) = rect( Tt 12 ).
An input signal x(t) = rect( 2t
12 ) is applied to the system.
T
x(t)
h(t)
y(t)
(b1) Determine the transfer function H(f )! Draw |H(f )| with proper axis labeling!
(b2) Draw y(t) with proper axis labeling!
(b3) Determine the Fourier Transform Y (f ) of the output signal y(t).
290
EXAMS
(c) (5 points)
base station.
Radio Channel
PTx = 5 W
PRx
LA
Mobile
Base Station
PN = 8 pW
The transmit power is PT x = 5 W. The noise power at the receiver is given as PN = 8 pW.

LA = 70 dB + 10 log10
d
km
3
dB,
where d is the distance between the transmitter and the receiver in km.
(c1) Calculate the maximum distance d, if the minimum received transmit signal power level
at the receiver is 78 dBm.
(c2) Calculate the Signal to Noise Ratio (SNR) at point A, if the receiver is at a distance
of d = 5 km from the transmitter.
Note:
x
log10 (x)
0.1 1 1.5 2
3 10 100 1000
1 0 0.2 0.3 0.5 1
2
3

The illustrated, analog-digital (A/D) converter is used to transform the time-continuous and valuecontinuous signal s(t) into a time-discrete and value-discrete signal zq (k). The analog input signal
is initially sampled with the sampling frequency fs and then converted into a digital signal via
quantization. Finally, the sample rate is reduced to the sampling frequency fs in order to adjust
the signal to subsequent processing units.


291
D.21 Exam SS 2011
(a) (2 points)
(a1) If the sampling frequency is too small, the sampled signal will be distorted. What is
the name of this eect?
(a2) Which sampling frequency is necessary to be able to reconstruct the sampled signal
distortion-free?
(b) (4 points)
(b1) S1 (f ) denotes the spectrum of the bandwidth-limited, analog signal s1 (t) illustrated in
the following gure.

Now, s1 (t) is sampled using the above A/D-converter and the increased sampling frequency fs is set to fs = 9 MHz. Draw the spectrum Zs (f ) of the sampled signal zs (k)
in the interval f [18 MHz, 18 MHz]! Pay attention to correct axis labeling!
292
EXAMS
(b2) Now, the input of the A/D-converter is fed by the single-carrier signal s2 (t) = sin(2f t)
with f = 500 kHz. Furthermore, s2 (t) is sampled with fs = 80 MHz at the Sample & Hold element. The sampled signal zs (k) is discretized with the ve-stage, linear
mid-tread quantizer {s (0, 1, 2, . . . , q/2 1); s (1, 2, 3, . . . , q/2)} with s = 0.5.
Moreover, the reduction of the sampling rate is applied with the factor fs /fs = fs /2fg =
10.
How many sampling values are created in each period of s2 (t)?
Sketch the sampling values in the interval k/fs = t [0 s, 2 s)!
Note:
x
sin(x)
0 /16 /8 3/16 /4 5/16 3/8 7/16 /2

0 0.20 0.38 0.56 0.71 0.83
0.92
0.98 1.00
(c) (4 points)
Now, discretization of the sampled signal zs (k) is done with a linear, mid-rise quantizer { s
(1, 3, 5, . . . , q 1)} in the A/D converter. The resolution amounts to 12 bit and the
2
dynamic range of the quantizer is dened as Umin = 2.048 V zs (k) 2.048 V = Umax .
(c1) The Signal-to-Noise Ratio (SNR) is used as a measure for the interference suppression
in the A/D conversion. What can be changed in the circuit of the A/D converter to
increase the SNR at the quantizer output by 6 dB?
(c2) Calculate the amount of quantization steps q and the width of one quantization step s!
(c3) Now, the quantizer is planned to have a maximum, absolute quantization error emax =
0.1 V. Determine the necessary number of quantization steps q!
293
D.21 Exam SS 2011

(a) (5 points)
The source signal x(t) should be transmitted by the use of amplitude modulation (AM).
The associated magnitude spectrum of the source signal |X(f )| is depicted in the following
gure.
|X(f )|
1
f
fg
fg
(a1) Formulate the common functional relationship between the the modulated transmit
signal s(t) and the source signal x(t) in time domain!
Note: Consider x(t) as given.
(a2) Which common condition has to be fullled by the modulation index to avoid zero
crossings of the envelope of the amplitude modulated signal s(t)?
(a3) Calculate the bandwidth BAM of the amplitude modulated signal s(t)!
(a4) Let the carrier frequency fc = 3fg . Draw the magnitude spectrum |S(f )| of the depicted
baseband signal |X(f )| after modulation! Pay attention to correct axis labeling!
294
EXAMS
(b) (3 points)
Another possible modulation technique is frequency modulation (FM). One application for
FM is analog radio transmission. In this case, the available bandwidth for one radio station
is limited to Bmax = 200 kHz. Additionally, the frequency shift is upper bounded by F =
75 kHz.
(b1) Calculate the amplication factor kFM to achieve the given frequency shift F ! Assume
a maximal magnitude of Am = 3 for the source signal.
(b2) Imagine, you want to transmit your own radio signal in the free frequency range depicted
in the following gure.
|S(f )|
fmin
fmax
Calculate the carrier frequency fc in a way that maximizes the available transmit
bandwidth and minimizes the interference to neighboring channels!
Calculate the maximum possible cut-o frequency fg for the source signal! Keep
in mind the maximum bandwidth Bmax of the frequency modulated signal.
(c) (3 points)
In practice, multiple radio stations are transmitted in parallel. If the radio listener selects
his favoured station, the analog signal is rstly selected by a bandpass lter and afterwards
amplied. With a proper resonant circuit, the frequency modulated signal is then translated
to an amplitude modulated signal. This enables the recovery of the source signal by an AM
envelope demodulator. The explained FM demodulator concept is shown in the following
gure.
$0
HQYHORSH GHPRGXODWRU
EDQG SDVV
ILOWHU
UHVRQDQW
FLUFXLW
"
295
D.21 Exam SS 2011
(c1) Sketch the structure of an AM envelope demodulator by means of a block diagram!

Entitle each block of your sketch!
(c2) Describe the tasks of the individual blocks in the AM envelope demodulator!
Note: You can do this in bullet point form.

(a) (6 points)
The following gure shows the constellation diagrams of two digital modulation schemes.

(a1) Give the names of both modulation schemes!

with a symbol duration of
(a2) A transmission system reaches a throughput of 1200 bit
s
Ts = 2.5 ms. Is it possible that one of the given modulation schemes was used? If yes,
which one? Give reasons for your answer!
(a3) Assign bits to the symbols of modulation scheme (2) such that the bit error rate is
minimized! Sketch the optimal decision boundaries, too!
(a4) Another system which also has a symbol duration of Ts = 2.5 ms reaches 1600 bit
. Give
s
the name of a modulation scheme that might have been used in the system! Give reasons
for your answer! Sketch the constellation diagram of the modulation scheme, too!
Now, there is a system which uses 2-frequency shift keying (2-FSK). The system transmits a bit
sequence dk . The bits are assigned to the frequencies f0 and f1 (f1 > f0 ) as follows:

s(t) =
s0 (t) = cos(2f0 t) f
ur dk = 0 t (kTs , (k + 1)Ts )
s1 (t) = cos(2f1 t) f
ur dk = 1 t (kTs , (k + 1)Ts )
The symbol duration is Ts = 1 ms.

(b) (5 points)
(b1) Let f0 = 1 kHz and f1 = 2 kHz. The bit sequence dk which is being transmitted is
given as dk = {0, 1, 1, 0, 1, 0}. Draw s(t) in the interval t [0, 6Ts ]! Pay attention to
296
EXAMS
(b2) The frequencies f0 and f1 are orthogonal if the following condition holds:

Ts
cos(2f0 t) cos(2f1 t)dt = 0
Let f0 = 0.5 kHz and f1 = 2.5 kHz. Are those frequencies orthogonal? Give reason for
your answer with a calculation!
Note: cos(a) cos(b) = 12 [cos(a b) + cos(a + b)]
(b3) Assume that the frequency f0 of s0 (t) is xed to a specic value. Find a rule for f1 as
a function of f0 such that there are no phase discontinuities when switching between
the transmit symbols! Note: The phase is periodic with 2. There are no phase
discontinuities if the cosine terms in s0 (t) and s1 (t) have equal arguments at the times
t = kTs .
(c) (3 points)
The following gure shows the correlation receiver which is used to receive the 2-FSK modulated signal. The 2-FSK transmitter that was used in task (b) is also used here. The symbol
duration remains Ts = 1 ms, too.
(k+1)Ts
r(t)
kTs
(k+1)Ts
kTs
r(t) cos(2f0 t + 0 )dt
r0 (k)
dk {0, 1}
r(t) cos(2f1 t + 0 )dt
r1 (k)
A synchronisation error causes a random phase rotation of 0 [0, 2) in the correlation

signals (see gure above). Let f0 = 0.5 kHz and f1 = 2.5 kHz. Furthermore, the channel
between transmitter and receiver is noiseless, i.e. r(t) = s(t).
(c1) At time k = 0 the bit d0 = 0 was transmitted. Give r0 (0) and r1 (0) as a function of 0 !
(c2) Does the random phase rotation inuence the detection? If yes, how?
297
D.22

(a) (a1) The system is not linear.
For the input ax1 [n] + bx2 [n], the output is y1 [n] = 0.5(ax1 [n] + bx2 [n])2 + 0.3(ax1 [n] +
bx2 [n]).
If two inputs ax1 [n] and bx2 [n] are applied separately, then the nal combined output
signal is given as y2 [n] = 0.5(ax1 [n])2 + 0.3ax1 [n] + 0.5(bx2 [n])2 + 0.3bx2 [n].
As y1 [n] = y2 [n], the system is nonlinear.
(1 point)
(a2) Unshifted input: y1 [n] = 0.5x2 [n] + 0.3x[n + 1]
Input shifted by k: y2 [n] = 0.5x2 [n + k] + 0.3x[n + k + 1]
The system is time invariant, since y2 [n] = y1 [n + k].
(1 point)
(a3) The system is not causal since when x[n] = 0 for n < 0, it does not hold that y[n] = 0
for n < 0.
(1 point)
(a4) y[n] = 0.3[n + 1] + 0.8[n] + 0.5[n 1].
(1 point)
(b) (b1) H(f ) is given as

H(f ) = ejf T T si(f T )
(1 point)
|H(f)|/T
0.8
0.6
0.4
0.2
0
3
Shape correct
Axis correct
0
fT
(1 point)
(1 point)
298
EXAMS
(b2) The output signal is given as
T/2
1/2
3/2
Shape correct
Axis correct
(1 point)
(1 point)
(b3) The frequency response of the output signal is given as

Y (f ) = H(f ) X(f )
Y (f ) = e
jf T
T si(f T ) ejf 2 T2 si(f T2 )

(1 point)
(c) (c1) Solving for minimum distance d

37 dBm (70 + 30 log10 (d)) dBm = 78 dBm
30 log10 (d) = 45
log10 (d) = 1.5
d = 30 km
Correct equations
Correct value
(1 point)
(1 point)
(c2) Signal level at the receiver is

LRx = LT x LA = 10 log10
5103 mW
1 mW
dBm (70 + 30 log10 (5)) dB = 54 dBm
where 10 log10 5 = 10 log10 10 10 log10 2 = 7

SNR is
(1 point)
LRx LN = 54 dBm 10 log10 (8 109 ) dBm = 27 dB

Equations correct
Values correct
(1 point)
(1 point)
299

(a) (a1) Aliasing
(1 point)
(a2) fs > 2fg (sampling theorem)
(1 point)
(b) (b1) Spectrum Zs (f ) of the sampled signal zs (k) with aliasing and overlaps in the spectrum
respectively.
Qualitatively correct: (Aliasing shown)
(1 point)
Quantitatively correct: (Axis label)
(1 point)
:jSF
F
-(Z
(b2)
FS
FS
-(Z
16 sampling values
(1 point)
quantization values and quantized signal zq (k) respectively according to the gure
below.
(1 point)
S
ST
ZQK
S

S
S

NIV

W LQ V
(c) (c1) Increase number of necessary bits to represent the signal alphabet by one or sample
with fourfold sampling frequency fs respectively
(1 point)
(c2) With the width of the dynamic range U = Umax Umin , the number of quantization
steps q, the step width s and the resolution in bit b follows
300
EXAMS
q = 212 = 4096.
U
s =
= 1 mV.
q
(2 points)
(c3) With the maximum, absolute quantization error emax , the number of quantization
steps q, the step width s and the width of the dynamic range U = Umax Umin
follows
U
.
q
s
e
emax .
2
U
.
q
2emax
q 21.
s =
(1 point)
(a) (a1) s(t) = (1 + x(t)) cos(2fc t)
(1 point)
(a2) |x(t)| 1 or max [x(t)] 1 0
(1 point)
(a3) BAM = 2fg
(1 point)
(a4) see gure magnitude spectrum |S(f )|.
|S(f )|
1
2
Axis labeling correct.

Shape (position, amplitude, bandwidth) correct.
(b) (b1) kFM =
(b2)
F
Am
= 25 kHz
min
i. fc = fmax +f
or fc =
2
ii. from B 2(fg + F )
f
fg
(1 point)
(1 point)
(1 point)
Bmax
2
+ fmin
fg,max = 25 kHz
(1 point)
(1 point)
301
$0
HQYHORSH GHPRGXODWRU
UHVRQDQW
FLUFXLW
EDQG SDVV
ILOWHU
UHFWLILHU
/3
+3
(c) (c1) Realization of the FM receiver

Rectier, LP (low pass) and HP (high pass) named.
Block diagram in the correct order.
(c2)
(1 point)
(1 point)
(1 point, if all three are correct)
component
rectier
low pass lter
high pass lter
task
negative parts of the signal were removed (half-wave rectication)
removes parts with high frequencies; envelope of the desired signal is reconstructed
removes the direct current (dc) component of the signal

( 12 point each)
(a) (a1) (1): QPSK, (2): 8-PSK

1
= 400 symbols
Ts
s
1200 bits/symbol
bit
=
3
400 symbols/s
symbol
(a2) fs =
(1 point)
Three bits per symbol correspond to eight constellation symbols. 8-PSK might have
been used.
(a3) Bit assignment with gray mapping
Boundaries sketched correctly
(1 point)
(1 point)
001
000
101
100
110
111
011
(a4)
010
1600 bits/symbol
400 symbols/s
bit
= 4 symbol
A modulation scheme with 16 symbols is required, e.g. 16-QAM.
Corresponding constellation diagram:
(1 point)
(1 point)
302
EXAMS

Q
I
(b) (b1) Qualitative correct

Quantitative correct
(1 point)
(1 point)
1
0.8
0.6
0.4
s(t)
0.2
0
-0.2
-0.4
-0.6
-0.8
-1
0
3
t/T
(b2)
Ts
?
cos(2f0 t) cos(2f1 t)dt = 0
0

Ts
Ts
?
1
1
cos(2(f
f
)t)dt
+
cos(2(f0 + f1 )t)dt = 0
0
1
2
2
0
0

=0
=0
f0 f1 = 2 kHz, f0 + f1 = 3 kHz and fs = T1s = 1 kHz

(Approach: 1 point)
Integration is done over whole periods in both summands and is therefore 0 which is
then also true for the sum. The orthogonality condition is satised!
(Conclusion:
1 point)
(b3) To avoid phase discontinuities, the following must hold:
2kTs f0 = 2kTs f1 nk 2 k
The term nk 2 is allowed because the phase is periodic. This gives:
nk
f1 = f0 +
kTs
Setting nk = nk gives:
n
Ts
In words: f1 must be larger than f0 by integer multiples of the symbol frequency
fs = Ts1
(1 point)
f1 = f0 +
(c) (c1) r0 (0) and r1 (0) are calculated as:

Ts
r0 (0) =
cos(2f0 t) cos(2f0 t + 0 )dt
0
303
1
=
2
Ts
0
1
cos(0 )dt +
2

Ts
0
cos(22f0 t + 0 )dt

=0, since 2f0 =fs =1 kHz
Ts
=
cos(0 )
2
(1 point)

Ts
r1 (0) =
cos(2f1 t) cos(2f0 t + 0 )dt = 0
A phase oset does not inuence orthogonality.
(1 point)
(c2) The distance between r0 (0) and r1 (0) is reduced. In a noisy channel this results in
reduced error tolerance and increased error rate. In the limiting case 0 = 2 it holds
that r0 (0) = 0 and a desicion is not possible even in the noise-free case.
(1 point)
304
D.23
EXAMS
Exam WS 2011/2012

Hint: The following questions (a), (b), and (c) can be answered independently.
(a) (4 Points)
y[n] = x[n 1] + x[n 2] x[n 3],
(a4) Calculate y[n] for x[n] = [n] + [n 1]!
Hint: Here [n] is the unit impulse, which is dened as

1 if n = 0
[n] =
0 else.
(b) (5 Points)
An LTI system with impulse response h(t) has an input signal x(t) and an output signal
y(t), where

1
if 0 t 2T
T
h(t) =
0 otherwise
and

x(t) =
t
T
if 0 t 2T
otherwise.
(b1) Is the system causal? Give reason for your answer!

(b2) Derive the output signal y(t) between 0 t 2T !
(b3) Draw the output signal y(t) between 0 t 2T ! Pay attention to correct axis labeling!
(c) (6 Points)
A radio station in Dresden transmits with power PT = 500 kW. A radio receiver receives
this signal in Calcutta, at a distance d = 7500 km away. The channel attenuation is given as

d
dB,
LA = 30 log10
km
305
D.23 Exam WS 2011/2012
(c1) Calculate the transmit power level, LPT , in dBm!

(c2) Calculate the channel attenuation, LA , between Dresden and Calcutta!
(c3) Calculate the received power level, LPR , at the radio receiver in Calcutta, in dBm!
(c4) Calculate the distance, d, between Dresden and Tokyo, if the received power in Tokyo
is 12 W!
Hint:
x
log10 (x)
0.1 1 1.5 2
3 10 100 1000
1 0 0.2 0.3 0.5 1
2
3

An analog signal s(t) with cut-o frequency fg = 4 MHz is digitized. The following gure illustrates
the spectrum S(f ) of the signal s(t).
3F
F0+]
(a) (3 Points)
(a1) Sampling is applied with a sampling frequency fs = 3/2 fg . Which disadvantageous
eect occurs in doing so? How must the sampling be changed in order to prevent this
eect?
(a2) The signal s(t) is sampled with a sampling frequency fs = 8 MHz. The sampled signal
is dened as x(t). Draw the spectrum X(f ) in the interval [12 MHz; 12 MHz]! Pay
attention to correct axis labeling!
(b) (4 Points)
In the following, we still consider a sampling of the signal s(t). In doing so, the sampling
frequency is set to fs = 6 MHz.
(b1) Draw the spectrum X(f ) of the sampled signal x(t) in the interval [10 MHz; 10 MHz]!
(b2) After the sampling, ltering is applied and an output signal y(t) results. In the process,
y(t) is considered as result of the convolution of the sampled signal x(t) with the
function h(t), with h(t) = si( t/t) and t = 1/2fs . Draw the amplitude spectrum
|Y (f )| for [10 MHz; 10 MHz]! Pay attention to correct axis labeling!
306
EXAMS
(c) (3 Points)
The time discrete signal x[k], k Z is given after the sampling. Subsequently, x[k] is quantized. A linear mid-rise quantizer is used for the quantization. The dynamic range of the
quantizer is given by 4s xin 4s. Here s denotes the step width of the quantizer.
(c1) Draw the characteristic of the quantizer xout as a function of xin ! Pay attention to
(c2) Determine the maximum, absolute quantization error for the dynamic range
U = 1.6 V !

(a) (4 Points)
An arbitrary source signal s(t) is modulated on the frequency of the carrier signal sc (t). This
is called frequency modulation (FM). The technique is widely applied to radio, television, and
communications systems. One requirement for parallel transmission of multiple modulated
signals is the band-limitation of each signal. The necessary bandwidth of a modulated signal
can be derived from its cut-o frequency fg .
(a1) Give one advantage and one disadvantage of FM!
(a2) The source signal s(t) with its maximum amplitude of |smax | = 2 is transmitted by
using FM. The maximal frequency deviation is limited to F = 4 fg . Calculate the
amplication factor kFM as a function of fg !
(a3) Give an estimate about the absolute bandwidth BFM as a function of fg by applying
the Carson rule!
(a4) The source signal is now transmitted by using AM instead of FM. How much will the
bandwidth BAM of the AM signal be smaller or greater than the FM signal?
(b) (3 Points)
The following gure shows the source signal s(t) in time domain. The modulated signal is
given by

F
sFM (t) = cos 2fc t + 2
s(t)t .
(D.1)
fg
In addition, the values for the carrier frequency fc =
F = T1 fg max (|s(t)|) are given.
2
,
T
and the frequency deviation
(b1) Give the occurring frequencies f1 and f2 of the modulated signal sFM (t) as a function
of T!
(b2) Draw the shape of the modulated time domain signal sFM (t) in the interval t = [0; 4T ]!
307
D.23 Exam WS 2011/2012
(c) (4 Points)
A speech of a famous book author is transmitted to the loudspeaker system with the help of
wireless microphones. For this purpose two microphones and a central receiver station are
installed. For parallel usage of both microphones, separate frequency ranges have to be used.
The cut-o frequency of the acoustic signal is given by fg = 40 kHz. The resulting bandwidth
of the AM bandpass signal for each microphone is given by BAM = 2 fg = 80 kHz.
(c1) Calculate the minimal bandwidth Bmin of the central receiver station for parallel reception of two microphones using AM!
(c2) The following gure shows the magnitude spectrum |SAM (f )| of the signal at the input
of the central receiver station for one active microphone. The depicted receiver structure for microphone no.1 shows, that the bandpass signal is down-converted into the
baseband (fc = 800 MHz) rst and low-pass ltered afterwards.
Draw the magnitude spectrum |SBB (f )| at the output of the receiver for microphone
no.1 in the interval f [0.12; 0.12] MHz! Pay attention to correct axis labeling!
^D
D,
(c3) Assume that microphone no.2 is using frequencies in the range from 800.08 MHz to
800.16 MHz.
Give the maximal possible cut-o frequency fg,max of the low-pass lter at the central
receiver station for receiving microphone no.1 without cross-talk from microphone no.2!
(a) (6 Points)
308
EXAMS
The following gure shows the constellation diagrams and waveform plots of three digital
modulation schemes.
Constellation A
Constellation C
Constellation B
1.5
jQ
jQ
jQ
0.5
0
0.5
1.5
1
0
I
Signal 1
1.5
0.5
0
0.5
I
Signal 2
7 6 5 4 3 2 1 0 1 2 3 4 5 6 7
I
Signal 3
1.5
1.5
8
1
0.5
0.5
s(t)
s(t)
s(t)
2
0
2
0
0.5
0.5
4
1
2
4
kT
8
0
4
kT
1.5
0
4
kT
(a1) Assign to each constellation diagram a tting waveform! Give the names and orders of
each modulation scheme!
(a2) A picture in a videostreaming system has an average size of 10 kByte. To avoid stuttering it is required to transfer at least 30 pictures per second. The symbol duration is
xed at T = 1 s. What is the minimum number of bits per transmit symbol in order
to fulll the requirements? Which of the given modulation schemes reaches or surpasses
the requirements regarding data throughput?
(a3) Which of the given modulation schemes can be received using an envelope detector.
(b) (5 Points)
The following gure shows the receiver of a digital transmission system.
309
D.23 Exam WS 2011/2012
kT
hr (t)
1 - 0
-1 - 1
b2k
1 - 0
-1 - 1
b2k+1
cos(2fc t)
sin(2fc t)
kT
hr (t)
(b1) Sketch the constellation diagram of the systems modulation scheme! For each constellation point, give the corresponding bits in the form b2k b2k+1 !
(b2) Sketch the block diagram of an appropriate transmitter! Give the name of each component!
(b3) How should hr (t) be chosen in order to minimize the bit error rate?
Hint: To answer this question, use your components from (b2).
(c) (3 Points)
The symbol error probability can be approximated by the pairwise error probability between
the symbols with minimal distance dmin . For the AWGN channel, the error probability Ps
for two symbols with distance d is given as:
1
Ps = erfc
2
"%
(d/2)2
2 2
#
,
where 2 is the variance of the noise.
(c1) Let 2 = 0.25. For the constellations A and C (see Task 4a), give the resulting symbol
error probabilities Ps !
(c2) Give the bit error rate Pb for constellation C using the given approximation under the
assumption of an optimal bit assignment to the constellation symbols!
(
Hints:
1
2
0.7
310
10
10
10
10
10
erfc(x)
10
EXAMS
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
311
D.24

(a) (a1) The system is not linear.
For the input ax1 [n] + bx2 [n], the output is
y1 [n] = (ax1 [n 1] + bx2 [n 1]) + (ax1 [n 2] + bx2 [n 2]) (ax1 [n 3] + bx2 [n 3]).
signal is given as y2 [n] = ax1 [n1]+a2 x1 [n2]x1 [n3]+bx2 [n1]+b2 x2 [n2]x2 [n3].
(1 point)
(a2) The system is time invariant.
If the input time index is changed to k + n, then the new output is given as
y1 [k + n] = x[k + n] + x[k + n 2] x[k + n 3].
If the output time index is changed, then we have
y2 [k + n] = x[k + n] + x[k + n 2] x[k + n 3].
As both the responses are same, the system is time invariant.
(1 point)
(a3) The system is causal as y[n] = 0 for n < 0 is satised.
(1 point)
(a4) y[n] = [n 1] + [n 2] + [n 3].
(1 point)
(b) (b1) The system is causal as h(t) = 0 for all t < 0.
(1 point)
(b2)
y(t) = h(t) x(t)

=
h( ) x(t ) d
For t < 0, y(t) = 0.

For 0 t < 2T , y(t) =
t
For 2T t < 4T , y(t) =
1 t
0 T T
d =
2T
1 t
t2T T T
t2
.
2T 2
d = 2
(t2T )2
.
2T 2
For t 4T , y(t) = 0.
For the interval of interest, the output signal is given as
y(t) =
Approach correct
Answer correct
t2
2T 2
for t [0; 2T ]
(1 point)
(1 point)
312
EXAMS
(b3) The output signal is given as

y(t)
2.0
0.5
2T
Shape correct
Axis correct
(1 point)
(1 point)
(c) (c1) The transmit power level is given as

9
500 103
10
= 87 dBm
= 10 log10
LPT = 10 log10
3
1 10
2
(1 point)
(c2) The channel attenuation is

LA = 30 log10 (7500) = 30 log10 (325100) = 30 log10
3
104
4

= 30(0.50.6+4) = 117 dB
(1 point)
(c3) The received power level in Calcutta is

LPR = LPT LA = 87 dBm 117 dB = 30 dBm
(1 point)
(c4) The received power level in Tokyo is

3
0.5 106
10
4
LPR = 10 log10
dBm = 33 dBm
=
10
log
(510
)
dBm
=
10
log
10
10
1 103
2
LPR correct
The channel attenuation suered to transmit to Tokyo is
LA = LPT LPR = 87 dBm (33 dBm) = 120 dB
(1 point)
313
LA correct
If d is the distance between Dresden and Tokyo, then
(1 point)
30 log10 (d) = 120

d = 104 km
Answer correct
(1 point)

(a) (a1) Aliasing, fs 2fg (sampling theorem)
(1 Point)
(a2) Spectrum X(f ) of the sampled signal x(t) without aliasing and without overlaps in the
spectrum, respectively.
Qualitatively correct: (no aliasing)
(1 Point)
Quantitatively correct: (axis labels)
(1 Point)
8F

F0+]
(b) (b1) Spectrum X(f ) of the sampled signal x(t) with aliasing and overlaps in the spectrum,
respectively.
Qualitatively correct: (aliasing)
(1 Point)
(1 Point)
8F

F0+]
(b2) Amplitude spectrum |Y (f )| results from the multiplication of |X(f )| and |H(f )|, with
H(f ) = 2f1s rect(f /2fs ) and fs = 1/t.
Qualitatively correct: (X(f ) ltered)
(1 Point)
(1 Point)
314
EXAMS


315
(c) (c1) Quantization characteristic according to the gure below.

(1 Point)
(1 Point)
XOUT
S
S
S
S
S S S
S
S
S
XIN
S
S
S
(c2)
(1 Point)
U
1.6 V
=
.
q
8
s
=
= 0.1 V.
2
s =
eabs,max
(a) (4 points)
(a1) Advantages:
(if 1 element each is right, than 1 point)
constant envelope of modulated signal

transmit amplier without back-o
insensitive against interference
Disadvantages:
high amount of bandwidth compared to AM
(a2) kFM =
F
smax
4fg
2
= 2fg
(1 point)
(a3) BFM 2 (F + fg ) = 2 (4fg + fg ) = 10fg
(1 point)
(a4) smaller, because BAM 2 fg
(1 point)
B = BFM BAM = 2 F = 8 fg
(b) (3 points)
(b1) f1 =
1
T
and f2 = T3 , because
(1 point)

1
2
1
2
s(t) t for t [0; 4T ]
sFM (t) = cos 2 t 2 s(t)t = cos 2
T
T
T
T
316
(b2) Axes labeling correct.

Shape correct.
EXAMS
(1 point)
(1 point)
(c) (4 points)
(c1) Bmin = 2 (2 fg ) = 160kHz
(1 point)
(c2) Axes labeling correct.

Spectrum shape correct.
(1 point)
(1 point)
^
(c3) |fg,max | = 80kHz
D,
(1 point)

(a) (6 Points)
( 12 Points per assignment / name)
(Total: 3 Points)
(a1) A - 1 - QPSK
B - 3 - 4-ASK
C - 2 - 8-ASK
(a2) Calculating the required data rate gives
R = 10 8 30 = 2400 kBit/s.
317
For a sampling frequency of fs =
1
T
= 1 MHz
nbit =
R
= 2.4
fs
bits per transmitted symbol are required.

8-ASK can be used since it has 3 bits per symbol.
(1 Point)
(1 Point)
(a3) Only modulation scheme C has all information in the amplitude and not in the phase.
For this reason, only scheme C can be received using an envelope detector. (1 Point)
(b) (5 Points)
(b1) Diagram and bit assignment as follows:
jQ
10
00
11
01
0
I
Diagram qualitatively correct

Correct bit assignment
(1 Point)
(1 Point)
(b2) Block diagram as follows or similar

b2k
0 - 1
1 - -1
ht (t)
cos(2fc t)
sin(2fc t)
b2k+1
0 - 1
1 - -1
ht (t)
Symbolmapper Pulseformer
Block diagram correct
Correct names
Mixer
(1 Point)
(1 Point)
(b3) Using the matched lter principle yields hr (t) = ht (t), where ht (t) is the impulse
response of the transmit lter (pulse former).
(1 Point)
318
EXAMS
(c) (3 Points)
(c1) Reading from the diagrams yields dmin,A = 2 and dmin,C = 1. Therefore,
"% #
1
1
0.16
2
Ps,A = erfc
= erfc (1)
0.08
2
2
2
2
and
Ps,C
1
= erfc
2
"% #
1
0.32
1
erfc (0.7)
0.16.
2
2
2
For each correct result
(1 Point)
(c2) Constellation C exhibits 3 bits per symbol. Since the symbol error rate is dominated
from pairwise errors of neighboring symbols. Due to Gray Coding, all pairs of neighboring symbols dier in exactly one bit. Therefore,
Pb =
Correct result
Ps,C
0.053.
3
(1 Point)
319
D.25 Exam SS 2012
D.25
Exam SS 2012

(a) (6 points)
WIFI services are provided inside a building. The access point (AP) can transmit at a power
of PT = 0.1 W, the receive signal power level at the distance of d = 10 m is LPR = 57 dBm.
The channel attenuation can be expressed as a function of the distance

LA = 40 dB + 10 log10
d
d0

dB,
where d is the distance between transmitter and receiver and d0 = 1 m is a reference distance.
(a1) Calculate the path-loss exponent !
Assume = 4 in the following.
(a2) Find the receive power level at distance d1 = 50 m!
(a3) Assume the noise power level is 140 dBm, and minimum required SNR = 40 dB at
the receiver. Calculate the maximum distance dmax the AP can cover!
(a4) Assume the same SNR requirement as in (a3). If we want to double the maximum
achievable distance, by how many dB does the transmit power have to be increased?
Hint:
x
0.1 1 1.5 2
3 10 100 1000
log10 (x) 1 0 0.2 0.3 0.5 1
2
3
(b) (3 points)
y[n] = x2 [n] + nx[n 2],
where n is the discrete time index. Answer the following questions with reason!
(b1) Is the system linear?
(b2) Is the system time invariant?
(b3) Is the system causal?
(c) (6 points)
A system with input signal x(t) and output signal y(t) is dened as
T
dy(t)
+ y(t) = x(t),
dt
where T is a constant.
320
EXAMS
(c1) Calculate the transfer function H(f ) of the system by the Fourier Transform!
n
Hint: The Fourier Transform theorem for dierentiation d dtx(t)
is (j2f )n X(f ).
n
(c2) Calculate the impulse response h(t) of the system accordingly!
Hint: A Fourier Transform pair could be used: x(t)
X(f ) =

=
e T , t 0
and
0, t < 0
T
.
1+j2f T
(c3) If the input signal x(t) is a rectangle pulse as

#
"
t T2
1, 0 t T
x(t) = rect
=
,
T
0,
else
calculate the output signal y(t) and sketch it!
(a) (2 points)
Analog signals can be converted into digital signals by means of analog-to-digital converters.
For this, the analog signal s(t) with its cuto frequency fg is sampled rst. The result is a
discrete time signal.
(a1) Which relation between sampling frequency fs and cuto frequency fg has to be fullled
in order to avoid aliasing errors?
(a2) The sampling procedure can be realized by the circuit depicted in the following gure.
What is such a circuit called?
(b) (4 points)
In the following, we are considering an analog speech signal s(t) with the magnitude
spectrum |S(f )|, which has to be digitally transmitted. The sampling frequency is set to
fs = 7 kHz. In order to avoid aliasing, the given signal s(t) needs to be initially limited in
bandwidth by an anti-aliasing lter with its transfer function H(f ).
321
D.25 Exam SS 2012
)| after low pass ltering!

(b1) First sketch the magnitude spectrum of |S(f
(b2) Draw the magnitude spectrum of the sampled signal
10.5 kHz f 10.5 kHz! Pay attention to correct axis labeling!
|Sd (f )|
between
(b3) For reconstructing the signal s(t), an ideal low pass lter should be used. Give the
cuto frequency fg,D/A of this lter!
322
EXAMS
(c) (4 points)
The following gure shows a mid-rise-quantizer. s corresponds to the distance between the
quantization stages. The input signal is given by s(t), the quantized output signal by sq (t).
(c1) The mid-rise quantizer has no quantization stage at sq (t) = 0. Justify briey, why this
could be a drawback for a speech signal with a lot of pauses in speech!
Hint: The speech signal consists of the useful signal and a basic noise oor.
(c2) Draw the quantization error e(t) = sq (t) s(t) as a function of the input signal s(t) in
the dynamic range 7s
s(t) 7s
! Pay attention to correct axis labeling!
2
2
(c3) An input signal s(t), which has an uniformly distributed amplitude in the interval
[2s; 2s], is applied at the given quantizer. Give the signal-to-noise ratio SN R at
the output of the quantizer!
323
D.25 Exam SS 2012

(a)
(b)
1.5
1
0.5
s(t)
s(t)
0.5
0
0.5
0.5
1
1.5
1
0
0.5
1.5
0.5
1.5

(a) (2 points)
A basic principle in communications is called modulation.
(a1) What does the term modulation describe in communications?
(a2) Why is modulation necessary in communications?
(b) (4 points)
The following two gures are showing two typical modulation schemes.
(b1) Give the name of the applied modulation scheme for (a) and (b) !
(b2) Which of the two depicted modulation schemes is more power ecient for radio transmission?
(b3) Why is the chosen modulation scheme more power ecient?
(b4) Which scheme requires less bandwidth for transmission when the same source signal is
used?
(c) (5 points)
The following picture shows the magnitude spectrum |R(f )| of the receive signal r(t) at the
receiver input.
|R(f )|
A
fc
B
fc
f
324
EXAMS
The receive signal r(t) should be translated from the original carrier frequency fc to an
intermediate frequency fZF by means of the depicted receiver. The given parameters are as
follows: carrier frequency fc = 100 MHz and signal bandwidth B = 20 MHz.
r(t)
s(t)
H(f )
z(t)
cos (2fm t)
(c1) Give the necessary input frequency fm of the mixer so that the intermediate frequency
of the output signal is fZF = 50 MHz!
(c2) Draw the magnitude spectrum |S(f )| of the signal s(t) at the output of the mixer in
the interval [300, 300] MHz! Pay attention to the correct axis labeling!
(c3) Design the succeeding lter H(f ) in the way that only the signal with carrier frequency
fZF is visible at the output of the receiver. Name the type and the cuto frequency fg
of this lter!
(c4) Sketch the magnitude spectrum |Z(f )| of the signal z(t) at the output of the receiver
in the interval [100, 100] MHz!
(a) (8 points)
The following picture shows a plot of a digitally modulated passband signal sBP (t) and the
corresponding transmitted bits.
10
00
11
01
BP
(t)
1
2
3
0
2
t / ms
(a1) Give the name of the modulation scheme! Sketch the constellation diagram! Add the
given bit assignment to the constellation diagram!
(a2) The bit assignment is not optimal with regard to minimal bit error rate. Give reason
why this is the case! Give the optimal bit assignment for this constellation!
325
D.25 Exam SS 2012
(a3) Find the data rate of this modulation scheme!

(a4) Generally, the signal sBP (t) can be expressed mathematically as
sBP (t) = Ts
dk hT (t kTs ) cos(2fc t)
k=
where Ts denotes the symbol duration and dk the k-th transmitted symbol. Give the
carrier frequency fc and the impulse response hT (t) of the transmit lter!
(b) (4 points)
A new transmission system using BPSK is to be developed. The bit value 0 shall be assigned
to the amplitude 1 and the bit value 1 to the amplitude -1.
(b1) Sketch the transmitter of the system from the incoming bitstream to the transmit
antenna! Give the names of the components!
(b2) The bit error rate of a BPSK transmission is given by
"%
#
1
SN R
BER = erfc
.
2
2
Furthermore
SN R[dB] = 10 log10 (SN R) .
Calculate the bit error rate for an SNR of 9dB!
Hints: Use the logarithm table of problem 1. The erfc-function is given by the diagram
below.
0
10
10
10
10
10
erfc(x)
10
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
x
326
EXAMS
(c) (2 points)
Again consider a BPSK transmission. There are conditional probability density functions
(PDF) of the amplitudes of the received signal rk at the input of the detector at the receiver. They are denoted by p(rk |dk ) where dk denotes the k-th transmitted BPSK-symbol.
Therefore dk = {1, 1}.
(c1) Sketch the conditional PDFs p (rk |dk ) of both BPSK-symbols as functions of rk in one
diagram for a channel without noise!
(c2) The transmit signal is now superimposed by additive white Gaussian noise (AWGN).
Sketch the resulting conditional PDFs p (rk |dk ) of both BPSK-symbols as functions of
rk in one diagram!
327
D.26

(a) (a1) The channel attenuation at d = 10 meters is

PT
dBm = 20 dBm
LPT = 10 log10
1mW
(1 point)
LA = LPT LPR = 20 dBm (57 dBm) = 77 dB

10
LA = 40 dB + 10 log10
dB = 40 dB + 10 log10 10 dB
1
So the path loss exponent is
=
77 40
= 3.7
10
(1 point)
(a2) The receive signal power level at d = 50 m is

LPR = LPT LA = 20 dBm (40 + 10 log10 504 ) dB = 88 dBm
(1 point)
(a3) The signal-to-noise-ratio is given as
SNR = LPR LPN = LPT LA LPN
(1 point)
To satisfy the SNR requirement
20 dBm (40 + 10 log10 d4 ) dB (140 dBm) 40 dB
20 + 140 40 40
=2
40
dmax = 100 m
log10 dmax =
(1 point)
(a4) To satisfy the SNR requirement at the distance d1 = 2dmax = 200 m
LPT1 (40 + 10 log10 (d1 )4 ) dB (140 dBm) = 40 dB
LPT1 = (40 + 10 log10 (d1 )4 ) dB + (140 dBm) + 40 dB = 32 dBm
So at least we need to increase the tranmit power by LPT1 LPT = 12 dB.
(1 point)
328
EXAMS
(b) (b1) The system is not linear.

For the input ax1 [n] + bx2 [n], the output is
y1 [n] = (ax1 [n] + bx2 [n])2 + n(ax1 [n 2] + bx2 [n 2]).
signal is given as y2 [n] = (a2 x21 [n] anx1 [n 2]) + (b2 x22 [n] bnx2 [n 2]).
(1 point)
(b2) The system is not time invariant.
If the input time index is delayed by k, then the new output is given as
y1 [n k] = x2 [n k] nx[n k 2].
If the output time index is delayed by k, then we have
y2 [n k] = x2 [n k] (n k)x[n k 2].
As y1 [n + k] = y2 [n + k], the system is not time invariant.
(1 point)
(b3) The system is causal as the output y[n] depends only on the current and past input.
(1 point)
(c) (c1) According to the Fourier transform, the system can be described as
(j2T f )Y (f ) + Y (f ) = X(f )
(1 point)
So the system function is then
H(f ) =
Y (f )
1
=
X(f )
1 + j2T f
(1 point)
(c2)

h(t) =
1 Tt
e
T
, t0
0,
t<0
(1 point)
(c3) The output signal can be computed by the convolution of input signal and the impulse
response
y(t) = x(t) h(t)

x( ) h(t ) d
=
For t < 0, y(t) = 0.

For 0 t < T , y(t) =
t
0
1 T1 e
t
T
d = 1 e T .
329
For t T , y(t) =
T
0
1 T1 e
t
T
d = (e 1)e T .
The output signal is given as
0,
t<0
t
y(t) =
1e T, 0t<T
(e 1)e Tt ,
tT
Approach correct
Answer correct
(1 point)
(1 point)
The output signal is given as

output signal y(t)
0.7
0.63
0.6
0.5
y(t)
0.4
0.3
0.2
0.1
0
3T
2T
1T
1T
t
2T
3T
4T
5T
(1 point)

(a) (a1) Aliasing is avoided, when fs 2fg (Nyquist sampling theorem)
(a2) sample and hold circuit
(1 Point)
(1 Point)
)| after lowpass ltering (refer to gure)

(b) (b1) Spectrum |S(f
Qualitatively correct (ltered and symmetrical at 2 kHz):
(1 Point)
330
EXAMS
)| by 7 kHz. No aliasing occurs.

(b2) Spectrum |Sd (f )| results from shifting |S(f
Qualitatively correct (no aliasing):
(1 Point)
(1 Point)
(b3) D/A conversion by means of low pass with cuto frequency fg,D/A = 3.5 kHz (1 Point)
(c) (c1) If (due to pauses in speech) a very low input signal |s(t)| << s/2 or no signal
(s(t)) = 0) is applied, there is however a noise signal generated at the output of the
quantizer. (Since the approximately maximum quantization error occurs always, the
(1 Point)
noise power is maximized to s2 /4.)
(c2) characteristic curve of the quantization error (refer to gure).
(1 Point)
(1 Point)
(c3) The dynamic range 2s leads to an eective quantization with b = 2 bit.

SN R (6 b) dB
12 dB
(1 Point)
331

(a) (2 points)
(a1) Modulation describes the adaption of the source signal to the physical transmission
medium, transmission channel or receiver. The baseband signal (to be transmitted) is
often modulated to carrier signal with a high frequency.
(1 point)
(a2) Modulation is essential in communications because
(min. 1 element, total 1 point)
multiple source signals are transmitted without interferening each other,

transmission is adapted to the characteristic of the transmission channel.
(b) (4 points)
(b1) Following modulation schemes:
(1/2 point each, total 1 point)
(a) AM - Amplitude modulation and
(b) FM/PM - Frequency or Phase modulation (more general: Angle modulation).
(b2) Angle modulation (Frequency or Phase modulation) is more power ecient. (1 point)
(b3) Reason:
(1 point)
By using amplitude modulation more than 50% of the transmission power is used for
the carrier signal. This separat carrier is not necessary when using angle modulation.
Additionally, the angle modulated signal shows a constant envelope. This fact can be
used to increase the ecieny of the ampler at the transmitter.
(b4) BAM = 2fg < BWM 2 (F + fg )
(1 point)
Amplitude modulation requires less bandwidth during transmission of the same source
signal compare to angle modulation.
(c) (5 points)
(c1) fm = 50 MHz (or fm = 150 MHz)
(1 point)
(c2) The magnitude spectrum of the signal s(t) as follows (for fm = 50 MHz):
|S(f )|
A
2
300
200
100
100
200
300
f
MHz
The magnitude spectrum of the signal s(t) as follows (for fm = 150 MHz):
|S(f )|
A
2
300
200
100
100
200
300
f
MHz
332
Axis labeling correct.

Spectrum correct.
EXAMS
(1 point)
(1 point)
(c3) The following lter:

(min. 1 lter named, total 1 point)
low pass lter with cut-o frequency 60 MHz< fg < 140 MHz or
bandpass lter with 0 MHz< fg,u < 40 MHz and 60 MHz< fg,o < 140 MHz.
(c4) The magnitude spectrum of the signal z(t) as follows:
|Z(f )|
A
2
100
100
f
MHz
Spectrum correct.
(1 point)

(a) (8 points)
(a1) It is 4-ASK
Constellation diagram:
(1 point)
(1 point each for diagram and bit assignment)
1
0
00
1
4
11
10
0
I
01
(a2) For AWGN channels, the most common error is a detection of a neighbouring symbol.
To minimize the bit error rate Gray mapping should be used since neighbouring symbols
only dier in one bit.
Reason correct
(1 point)
Diagram correct
(1 point)
1
0
00
1
4
01
11
0
I
10
(a3) fs = 1/Ts = 1 kHz. For 2 bits per symbol the resulting rate is R = 2 kBit/s. (1 point)
(a4) fc = 5 kHz

hT (t) = rect Tts 12
(1 point)
(1 point)
333
(b) (4 points)
(b1) Transmitter as below or otherwise correct

Diagram correct
Names correct
(1 point)
(1 point)
(b2) SN R[dB] = 10 log10 (SN R)

0.9 = log10 (SN R)
0.3 + 0.3 + 0.3 = log10 (SN R)
SN R = 8
BER = 12 erfc(2)
BER 12 5 103
BER 2.5 103
(1 point)
(1 point)
(c) (2 points)
(c1) Diagram correct
(1 point)
1.5
p(r |1)
p(r |1)
p(rk|dk)
0.5
0
2
0
r
(c2) Diagram correct
(1 point)
334
1.5
p(rk|1)
p(r |d )
p(rk|1)
0.5
0
2
0
r
EXAMS
335
D.27 Exam WS 2012/2013
D.27
Exam WS 2012/2013

Remark on the awarding of points of a drawing: A second point for correct axis labeling is
given only if the drawing is qualitatively correct.
(a) (5 Points)
The transmit signal power level, LT is 20 dBm.
The mobile device is situated at the distance of d meters (m) from the transmitter. The
signal attenuation is given by the following formula,

d
LA = 40 dB + 20 log10
dB.
m
Consider the received power level, LR , at the mobile station to be 60dBm.
(a1) Give the transmit power PT , in mW!
(a2) Give the channel attenuation, LA , between transmitter and receiver!
(a3) Give the distance, d, between transmitter and receiver!
(a4) A thermal noise PN = 1010 W is added at the receiver. Calculate the signal to noise
ratio SNR in dB!
Hint:
x
log10 (x)
0.1 1 1.5 2
3 10 100 1000
1 0 0.2 0.3 0.5 1
2
3
(b) (5 Points)
Consider a signal h(t) as follows,
h(t)
2
1
2T 3T
(b1) Give the analytical expression of h(t) using the rectangle (rect (t/T )) functions only,
where

1 if T /2 t T /2
rect (t/T ) =
0 otherwise.
336
EXAMS
(b2) Calculate the frequency response H(f ) of h(t) using the properties of Fourier transform!
(b3) Consider a time signal z(t), the real part is known to be odd while the imaginary part
is even with respect to time.
What can be concluded about the real and imaginary part of its Fourier transform
Z(f )?
(Please choose only one option from (i) to (iii))
i. Z(f ) is purely real.
ii. Z(f ) is purely imaginary.
iii. Z(f ) consists of both real and imaginary parts.
What can be concluded about the symmetry of its Fourier transform Z(f )?
(Please choose only one option from (iv) to (vi))
i. Z(f ) is an even function with respect to f .
ii. Z(f ) is an odd function with respect to f .
iii. Z(f ) consists of an even and an odd part with respect to f .
(c) (5 Points)
Now consider h(t) given in part (b) as an impulse response of the LTI system with x(t)
as the input to the system and the output y(t). The input x(t) is applied and is given as
follows,

x(t) =
x(t)
1
0
if 0 t T
otherwise.
h(t)
y(t)
(c1) Is the system causal? Please give reason for your answer!
(c2) Give the relation between x(t), y(t) and h(t) in time domain!
(c3) Give the expression for y(t)!
(c4) Sketch y(t)!

(a) (7 Points)
An analog signal s(t) has to be digitized by means of an analog-digital converter. At its input
the wanted signal s(t) is disturbed by an interference signal i(t). The described scenario as
well as the spectra of the signals are depicted in the following gures.
D.27 Exam WS 2012/2013
337
(a1) Which relation between sampling frequency fs and cuto frequency fg of a signal has
to be fullled in order to avoid aliasing errors?
(a2) Give the cuto frequency fg of the input signal z(t)!
(a3) Draw the magnitude spectrum |Zs (f )| at the sampler output for the case that
fs = 40 MHz! The considered range is 50 MHz f 50 MHz! Pay attention to correct
axis labeling!
(a4) Is it possible to reconstruct the wanted signal s(t) out of |Zs (f )| without any error?
Justify your answer!
(a5) Give the minimum sampling frequency fs,min for which the wanted signal s(t) can be
reconstructed just error-free!
(a6) The input signal z(t) is now sampled keeping the sampling condition in (a1). For reconstructing the wanted signal s(t) an ideal lowpass lter should be used. Additionaly,
by adjusting its cuto frequency we want to remove the interference signal i(t), so that
the following applies: z(t) = s(t).
Give a valid lter function R(f )!
Hint: The impact of the quantizer can be neglected!
338
EXAMS
(b) (3 Points)
An analog mono audio signal has to be digitized. The sampling rate of the analog-digital
converter is xed to 48 kHz, the number of quantization stages can be adjusted arbitrarily.
(b1) For the given audio signal we assume a uniformly distributed amplitude over
[2 V, +2 V]. In addition, the mean power (refered to 1 ) of the quantization
error is not allowed to exceed Pe = 1/48 V2 .
How many bits are needed to quantize the signal?
What is the signal-to-noise ration (SNR) of the quantized signal?
(b2) In the following, the audio signal is quantized with 16 bit. For each 24 byte audio data,
an additional byte is needed to record subchannel data.
How many megabyte (MB) free disk space has to be available in order to store a mono
audio signal of 3 minutes length?
Hint: 1 megabyte (MB) = 1000 1000 byte

(a) (7 points)
The signal x(t) = sin (2fs t), where fs = f8c , is amplitude modulated (AM) for transmission.
The modulated band-pass signal is given as,
sAM (t) = (1 + x(t)) cos (2fc t)
sAM (t)
, where is the modulation index and fc the carrier frequency.

The following gure shows one possible realization of the transmitted signal sAM (t).
2.0
1.5
1.0
0.5
0
0.5
1.0
1.5
2
0
10 12 14 16 18 20
t in s
(a1) Give the modulation index and the carrier frequency fc of the shown signal!
(a2) Draw the transmit signal sAM (t) in the interval t [0, 5] s given that fc = 2 MHz and
= 1/2! Pay attention to correct axis labeling!
(a3) Draw the transmit signal |SAM (f )| in the interval f [3/2fc , 3/2fc ] given that fc =
2 MHz and = 1/2! Pay attention to correct axis labeling!
339
D.27 Exam WS 2012/2013
(a4) Give the signal power Ps and the power of the carrier Pc of the transmit signal sAM (t)
given that fc = 2 MHz and = 1/2!
Note: Make use of the following equation to calculate the power of the signal and
carrier:

Px =
|X(f )|2 df.
(b) (4 points)
The spectrum R(f ) at the input of the receiver is shown in the following gure. The received
signal is known to be amplitude modulated with a modulation index and a carrier frequency
fc . The modulation index is = 1.
R(f )
fc fs,1
fc fs,2
fc + fs,1
fc + fs,2
fc fs,1
fc fs,2
0.5
fc
fc + fs,1
fc + fs,2
f
fc
0.5
The useful part x(t) of the received signal r(t) is ltered by an ideal synchronous demodulator
as shown in the following gure. After the synchronous down-conversion, the signal is ltered
by an ideal low-pass (LP) lter with cut-o frequency fs,2 < fg,TP < 2fc fs,2 , rst, and than
ltered by an ideal high-pass (HP) lter with cut-o frequency 0 < fg,HP < fs,1 .
Synchronous down-conversion mixer
r(t)
LP
HP
u(t) x(t)
carrier recovery
ideal BP at fc
cos 2fc t
(b1) Draw the spectrum of the demodulated signal X(f ) in the interval f [2fs,2 , 2fs,2 ]!
(b2) Give the useful part of the signal x(t) at the output of the demodulator!
(b3) The cut-o frequency of the low-pass lter in the demodulator is to low due to process
tolerances. In this case, the cut-o frequency is fg = (fs,1 + fs,2 ) /2.
Give the useful part of the signal x(t) at the output of the demodulator!
340
EXAMS

(a) (5 Points) The following gure shows the constellation diagrams of three dierent modulation
schemes.
&
(a1) Name each modulation scheme! Draw the decision thresholds into the gure!
(a2) A video streaming system uses 8-PSK as modulation scheme.
How many symbols have to be at least transmitted per second, if the data rate for
unimpeded transmission is 360 bit/s?
(a3) Name a modulation scheme that achieves the rate of 360 bit/s with a symbol rate of
60 symbols/s!
(b) (5 Points)
The following gure shows the receiver of a digital 4-ASK system.
FRVIF W
N7V

KUW
PL[HU
UHFHLYH ILOWHU
VDPSOHU
GHFLVLRQ PDNHU
GHPDSSHU
(b1) Sketch and label a corresponding transmitter!

(b2) Is Gray mapping used in the symbol mapper? Please give reason for your answer!
(b3) Name one reason, why in practice rectangular lters are not used as impulse shapers!
(b4) 4-PSK is also a four-state modulation scheme. Why can the depicted receiver still not
be used for a 4-PSK system?
(c) (4 Points)
The eye diagram is a graphical depiction of signals that is used to estimate the quality of a
received signal. It is constructed by overlaying multiple periods of the signal. The following
gure shows the eye diagrams of the signals of two dierent systems after passing through
341
D.27 Exam WS 2012/2013
the receive lter.
1
Amplitude
Amplitude
0,5
0.5
0
0.5
0
t/Ts
0.5
0.5
0
t/Ts
0.5
(c1) In which system can you expect less bit errors? Please give reason for your answer!
The following gure shows another eye diagram of a purely real signal.
2.5
2
Amplitude
1.5
1
0.5
0
0.5
1
1.5
0.5
0
t/Ts
0.5
(c1) Draw the constellation diagram of the modulation scheme that was used! Pay attention
to correct axis labeling!
342
D.28
EXAMS

(a) (5 Points)
(a1) The transmit power PT is given as,

PT
LT = 10 log10
dBm
1mW

PT
dBm
20 dBm = 10 log10
1mW
PT
= 102
1 mW
= 100 mW
(1 point)
(a2) The received power level is given as,
LR = L T L A
L A = LT L R
= 80 dB
(1 point)
(a3) The path loss between transmitter and receiver is given as,

LA = 40dB + 20 log10
d
m

dB.
As LA = 80 dB,

d
40dB + 20 log10
dB
m

d
40dB + 20 log10
dB
m

d
dB
20 log10
m

d
log10
dB
m
100 m
LA dB =
80dB =
80 40 =
2 =
d =
(1 point)
343
(a4) The noise level PN = 1010 W and the received signal level is LR = 60 dBm. First
we calculate the noise power level in dBm.

PN
10 log10
1mW
7
10 log10 10
10 (7) log10 (10)
70 dBm
LN =
=
=
=
Noise level in dBm
(1 point)
SNR = LR LN
= 60 dBm (70 dBm)
= 10 dB
SNR calculation
(1 point)
(b) (5 Points)
(b1) The signal h(t) can be written as follows,

h(t) = rect
t T /2
T

+ 2 rect
t 2T
2T
(1 point)
(b2) As we know,
T si(f T )
rect(t/T )
also we know from above part,

h(t) = rect
t T /2
T

+ 2 rect
First we nd the Fourier transform of rect

rect
t T /2
T
tT /2
T
t 2T
2T

T ej2f T /2 si(f T )
Now similarly we nd the Fourier transform of second part of h(t)

t 2T
4T ej4f T si(2f T )
2 rect
2T
344
EXAMS
And nally using the linearity property of the Frourier transform

H(f ) = T si(f T )ejf T + 4T si(2f T )ej4f T
(1 + 1 points)
(b3) (ii) and (iii) are correct. As the signal z(t) has odd real and even imaginary part.
This translates into odd imaginary and even imaginary parts in Fourier transform
respectively.
(1 + 1 points)
(c) (5 Points)
(c1) Yes the system is causal as h(t) = 0 for t < 0 is satised.
(1 point)
(c2) The output of the system is given by the convolution of input x(t) and impulse response
h(t). The relation between x(t), y(t) and h(t) is gives as,
y(t) = x(t) h(t)

=
h(t )x( ) d

x(t )h( ) d
=
(1 point)
(c3) The output y(t) is given as,
For t 0 y(t) = 0.
For 0 t T ,
t
y(t) = 0 (1) (1) d = t.
For T
t
Tt 2T ,
y(t) = T +t (1) (1) d + T (1) (2) d = 3t 4T .
For 2T t 3T ,
t
y(t) = T +t (1) (2) d = 2T .
For 3T t 4T ,
3T
y(t) = T +t (1) (2) d = 8T 2t.
For t 4T y(t) = 0.
Answer correct
(2 points)
345

y(t)
2T
T
T
2T
3T
4T
Sketch correct
(1 point)

(a) (a1) Aliasing is avoided, if fs 2fg (Nyquist sampling theorem).
(1 Point)
(a2) The (upper) cuto frequency of |Z(f )| is fg = 25 MHz.
(1 Point)
(a3) Spectrum |Zs (f )| result from periodic shift of |Z(f )| by 40 MHz. No aliasing occurs.
Qualitatively correct: (no aliasing)
(1 Point)
Quantitativly correct: (axis labeling)
(1 Point)
(a4) Yes, despite subsampling of z(t), s(t) can be reconstructed error-free since there is no
spectral overlap of the wanted and interference signal (no aliasing).
(1 Point)
(a5) fs,min = 35 MHz
(a6) A rectangular lter: R(f ) = rect
f
fB
(1 Point)
with 20 MHz fB 40 MHz.
(1 Point)
(b) (b1)
Pe =
1 2 (s)2
V =
48
12
From this the distance between the quanization stages follows: s = 12 V

The number of bits b results in:

Umax Umin
bit
b = log2
s

2V + 2V
= log2
bit = log2 23 bit
0.5 V
= 3 bit
346
EXAMS
(1 Point)
Signal-to-noise ration (SNR) is given by:
SN R (6 b) dB
18 dB
(1 Point)
(b2)
data rate r = sampling rate (audio data + subchannel data)
= 48 kHz (2 B + 1/12 B)
= (96 + 4) kB/s
data D = 100 kB/s 180 s
= 18 MB
(1 Point)
(a) (7 points)
(a1)
(1 point each, total 2 points)

Modulation index is determined by the envelope (1 + x(t)) with max (x(t)) = 1
for t = 2s: (1 + max (x(t))) = (1 + 1) = 74 = 34 .
Carrier frequency is determined by the length of a period T of one oscillation. Here
T = 1s fc = T1 = 1 MHz.
(a2) Signal sAM (t):

1.5
sAM (t)
1.0
0.5
0
0.5
1.0
1.5
0
0.5 1.0 1.5 2.0 2.5 3.0 3.5 4.0 4.5 5.0
t in s
Qualitatively correct: (signal shape)
Quantitatively correct: (axis labeling)
(1 point)
(1 point)
347
(a3) Spectrum |SAM (f )|:

|SAM (f )|
0.5
0.25
fc fc /8 fc + fc /8
fc
fc fc /8 fc + fc /8
fc
0.25
(a4)
(1 point)
(1 point)
(1/2 point each, total 1 point)

The power of the signal and the carrier can be determined by using the transfer function
SAM (f ):
2
2
1
Ps = 2 2 4 = 4 = 16
2
Pc = 2 12 = 12
(b) (4 points)
(b1) Spectrum X(f ):
X(f )
fs,1
fs,2
0.5
0
fs,1
fs,2
f
0.5
(1 point)
(1 point)
(b2) Calculation of x(t):
(1 point)
1
3
((f + fs,2 ) + (f fs,2 )) + ((f + fs,1 ) + (f fs,1 ))
2
4
3
x(t) = cos (2fs,2 t) + cos (2fs,1 t)
2
3
x(t) = cos (2fs,2 t) + cos (2fs,1 t)
2
X(f ) =
348
(b3) Calculation of x(t):
EXAMS
(1 point)
x(t) =
3
cos (2fs,1 t)
2

(a) (5 Points)
(a1) A - 16-QAM
B - 4-PSK
C - 4-ASK
&
( 12 point for each correct name/ correct decision threshold)
(Total: 3 Points)
(a2) The symbolrate can be calculated as

R=
360 bit/s
= 120 symbols/s
3 bit/symbol
(1 Point)
(a3) At 60 Symbols/s, a scheme with

nbit =
360 bit/s
= 6 bit
60 symbole/s
per constellation symbol is required. For example, 64-QAM can be used.
(b) (5 Points)
(b1) Diagram like depicted or similar
(1 Point)
349
FRVIF W

PDSSHU
'
KWW
$
GLJLWDO WR DQDORJ
FRQYHUWHU
LPSXOV VKDSHU
PL[HU
Diagram correct
Labeling correct
(1 Point)
(1 Point)
(b2) No, because the symbols for 01 and 10 dier by 2 bit.

(b3)
(1 Point)
Innite bandwidth
Dicult to implement
One correct reason
(1 Point)
(b4) The quadrature phase must be demodulated separately, a second path is required.
(1 Point)
(c) (4 Points)
(c1) For the right signal less errors can be expected, since the vertical and horizontal eye
opening is wider.
Correct choice
(1 Point)
Correct reason
(1 Point)
(c2) Diagram like depicted or similar:
Qualitatively correct: (I/Q points)

(1 Point)
(1 Point)
350
Notes

Communications Script 2013 en

Uploaded by

Document Information

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Communications Script 2013 en

Uploaded by

Copyright:

Available Formats

Lectures script

Prof. Dr.-Ing. Dr. h.c. Gerhard P. Fettweis

February 12, 2013

2 Goal of the Lecture

Systems for Information Transmission . . . . . . . . . . . . . . . . . . . . . . . . .

Dirac Delta Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Integration of Generalized Functions . . . . . . . . . . . . . . . . . . . . .

Dierentiation of Generalized Functions . . . . . . . . . . . . . . . . . . .

Calculation with Dimensional Parameters . . . . . . . . . . . . . . . . . .

4 Linear Time Invariant System

Convolution as the Basic Operation in Communications . . . . . . . . . . . . . . .

Consideration in Time and Frequency . . . . . . . . . . . . . . . . . . . . . . . . .

Fourier Transformation of Rect Function . . . . . . . . . . . . . . . . . . .

Zeit- und Frequenzbegrenzung . . . . . . . . . . . . . . . . . . . . . . . . .

Central Limit Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Summer semester 2013

Up and Down Conversion of Real Baseband Signals . . . . . . . . . . . . . . . . .

Up and Down Conversion of Complex Baseband Signal . . . . . . . . . . . . . . .

General Bandpass Signal Equivalent Low-Pass Signal . . . . . . . . . . . . . . .

Description in Time Domain . . . . . . . . . . . . . . . . . . . . . . . . . .

Description in Frequency Domain . . . . . . . . . . . . . . . . . . . . . . .

Dierent Types of Amplitude Modulation . . . . . . . . . . . . . . . . . .

Phase and Frequency Modulation (Angle Modulation) . . . . . . . . . . . . . . . .

Description in Time Domain . . . . . . . . . . . . . . . . . . . . . . . . . .

Description in Frequency Domain . . . . . . . . . . . . . . . . . . . . . . .

Comparison between Amplitude and Angle Modulation . . . . . . . . . . . . . . .

Signal Reconstruction Digital-Analog-Conversion . . . . . . . . . . . . .

Quantization Characteristic Curve

Quantization Error Signal . . . . . . . . . . . . . . . . . . . . . . . . . . .

Error Modelling of AD Converter . . . . . . . . . . . . . . . . . . . . . . .

Script Introduction to Communications

BPSK Binary Phase Shift Keying . . . . . . . . . . . . . . . . . . . . . . . . . .

Description in Time Domain . . . . . . . . . . . . . . . . . . . . . . . . . .

Description in Frequency Domain . . . . . . . . . . . . . . . . . . . . . . .

Data Transmission over Noisy Channel . . . . . . . . . . . . . . . . . . . .

Calculation of Bit Error Ratio . . . . . . . . . . . . . . . . . . . . . . . . .

QPSK Quaternary Phase Shift Keying . . . . . . . . . . . . . . . . . . . . . . .

Further ASK/PSK Methods . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Frequency Shift Keying . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

A.2 Fourier Transformation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

A.3 Notes on Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

B.1 Analytical Signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

C.2 Solutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

D.1 Exam SS 2006 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147

D.8 Solutions: Exam WS 2007/2008 . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191

Script Introduction to Communications

Summer semester 2013

2 GOAL OF THE LECTURE

Goal of the Lecture

Systems for Information Transmission

Figure 2.1: General transmission system in communications

Script Introduction to Communications

2 GOAL OF THE LECTURE

In the following, we name few characteristics of the communication systems.

abs. Power in [mW] 0.1 0.5 1 2 4 8 10 100 1000

2.5 Signal Level

In passive systems attenuation is often given instead of amplication.

gives the signal-to-noise-ratio.

Summer semester 2013

frictionless vibration equation

damped oscillation equation

current ow in the resonant circuit

results in the expression