FM radio receiver

P5 projekt, AAU,
Elektronik og elektroteknik

Gruppe 415
Carsten
Jes Toft Kristensen
Gustav
Kingo
Onkel Boye
NC
 Elektronik og Elektroteknik
Fredrik Bajers Vej 7B
Telefon 96 35 98 36
Fax 98 15 36 62
http://www.esn.aau.dk

Title:
FM radio receiver
Theme:
Realtime systems

Projectperiod:
P5, fall semester 2005

Project group:
506

Synopsis:
Members:
Carsten
Jes Toft Kristensen Insert synopsis here. . . found in mainRe-
Gustav port/frontMatter/synopsis.tex
Kingo
Onkel Boye
NC

Supervisor:
Persefonis

Copies: 9
Pages: 7
Appendices: 19
Finished October 12, 2005
Contents

A Radiotechnolgy 7
A.1 Purpose of the projekt . . . . . . . . . . . . . . . . . . . . . . . . . . 8

B Sound card 9

C Modulation 11
C.1 Amplitude modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 11
C.2 Frequency modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 14

D Downconverter 19
D.1 Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
D.2 Building blocks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
D.2.1 Mixer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
D.2.1.1 Filter . . . . . . . . . . . . . . . . . . . . . . . . . . 21
D.3 Block diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
D.4 Mathematical analysis . . . . . . . . . . . . . . . . . . . . . . . . . . 22
D.5 Dimensioning . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
D.6 Simulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
D.7 Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

E Sampling 26
E.1 From continuous to discrete . . . . . . . . . . . . . . . . . . . . . . . 26
E.2 Frequency attributes . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

F Measurements of downconverter 31
F.1 Method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
F.2 Results . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
F.3 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

G Filter 34

4
Insert text here. . .

5
BIBLIOGRAPHY

Oppenheim, Alan V. & Schafer,

Ronald W. (1998).
Discrete time signal processing,
2nd edition.
ISBN 0-13-754920-2
(Prentice Hall, 1998).

Bibliography

Christensen, Anders (1999).

HF-teknik,
1st edition.
ISBN 87-600-0129-1
(Industriens Forlag, 1999),
URL www.if.dk.

Haykin, Simon (2001).

Communication Systems,
4th edition.
ISBN 0-471-17869-1
(John Wiley & Sons, Inc., 2001).

Intel Corporation (2002).

Audio Codec ‘97
(Intel Corporation, 2002),
URL http://www.intel.com/
design/chipsets/audio/ac97_r23.
pdf. Revision 2.3 Revision 1.0.

Johnson, David E., Johnson, Johnny R.,

Hilburn, John L. & Scott, Peter D.
(1999).
Electric circuit analysis,
3rd edition.
ISBN 0-471-36571-8
(John Wiley & Sons, Inc., 1999).

Laskar, Joy, Matinpour, Babak &

Chakraborty, Sudipto (2004).
Modern Receiver Front-Ends - Sys-
tems, Circuits, and Integration,
1st edition.
ISBN 0-471-22591-6
(John Wiley & Sons, Inc., 2004).

6
 APPENDIX A. RADIOTECHNOLGY

Appendix A
Radiotechnolgy

The purpose of a radio transmitter (fig. A.1)is to transform a signal to radiowaves,

thus enabling a wireless signal transfer. This is done by utilizing the fact that an
alternating current creates a electromagnetic field around the antenna. This field
emanates from the antenna, and can be received by another antenna.

Figure A.1: Function diagram of radio transmitter

To enable the transmission of multiple signals, the signals are modulated to

another frequency by either Amplitude Modulation (AM), Phase Modulation (PM)
or Frequency Modulation (PM). Modulation tecniques are discussed in appendix
C. This enables transmission of more than one signal at a time, as each signal has a
different carrier frequency (fig. A.2). Due to the requirements of sound quality for
commercial radiostations, they require a bandwidth of 150kHz. The radio amateurs
only require 16kHz, due to the fact that they do not need a high sound quality,
as they only transmit speech. [Christensen, 1999] FiXme: find sidetal i HF-teknik,
eller en anden kilde

Figure A.2: Frequency overview of a FM-modulated signal.

The purpose of a radio receiver (fig. A.3) is to receive the transmitted signal,
and transform it to its original form.

7
A.1. PURPOSE OF THE PROJEKT

Figure A.3: Function diagram of radio receiver

A.1 Purpose of the projekt

The purpose of this projekt is to convert a FM-modulated signal to sound. This
signal has been chosen to have a carrier wave of 145MHz and a bandwidth of 16kHz.
This is due to:
• The limited bandwidth of the soundcard (see appendix B for capabilities of
the sound card, and appendix D.4 for the consequenses of downconversion)
• The 145MHz is useable by all for transmission, which will enable us to test
the system with a transmitted signal

8
 APPENDIX B. SOUND CARD

Appendix B
Sound card

Sampling a signal using using a PC sound card is only possible if the sound cards
meets certain specification set by the properties of the input signal. The input
signal may also have to be adapted to meet the input specification of the sound
card. The purpose of this appendix is to examine the specification of a general PC
sound card. Audio Codec ’97 is a royalty-free sound card standard developed by
Intel Corporation. The specification defines the architecture and digital interface
of a sound card including analog performance characteristics of the input signal.
In order to sample the input signals there are a few key parameters the have to
be met by the AC’97 standard. These parameters include the bandwidth and
the sampling resolution. In the AC ’97 v2.3 Component Specification the key
parameters concerning the frequency response and sampling frequency is listed as
in table B.1 Intel Corporation

Parameter Min typ Max Units

Sampling Frequency - - 48000 Hz
Analog frequency responce ± 1 dB 20 - 20000 Hz
Transition band 19100 - 28800 Hz
Stop band 28800 - - Hz
Stop band rejection -74 - - dB
Group delay - - 1 ms

Table B.1: Frequency response of an AC’97 compliant sound card

Other parameters that have to be taken into consideration when sampling using
an AC’97 sound card is listed in table B.2 AC’97 defines the sampling resolution

Parameter Min typ Max Units

Mic full scale input voltage (20 dB boost) - 0.1 - Vrms
Mic full scale input voltage (0 dB boost) - 1 - Vrms
Input imdance 10 - - kΩ
Input Capasitance - 7.5 - pF

Table B.2: Input parameters of an AC’97 compliant sound card

9
as full-duplex 16 bit.

10
 APPENDIX C. MODULATION

Appendix C
Modulation

The purpose of a communication system is to transmit an information-bearing

signal from a transmitter to a receiver. The information-bearing signal is referred
to as baseband signal, where the term baseband is used to designate the band of
frequencies representing the original signal.
Often it is necessary to shift the baseband frequencies to a frequency range more
suitable for transmission, which is done by modulation. Modulation is a method
where a carrier frequency is changed in accordance to a modulation signal. As
carrier frequency a sinusoidal wave is often used, in which case the modulation
becomes a continuous-wave modulation process. The modulation frequency is the
baseband signal, and the result of the modulation is referred to as the modulated
signal.
In this report two types of modulation technics are described amplitude mod-
ulation (AM) and frequency modulation (FM). In AM the modulation signal fm
modulates the amplitude of the carrier signal fc , and in FM the modulation sig-
nal fm modulates the frequency of the carrier signal fc . In figure C.1 a AM- and
FM-signal is shown. For the AM signal the frequency is determined by fc and
the amplitude is determined by fm , whereas for the FM signal the amplitude is
determined by fc and the frequency is determined by fm . The appendix is based
on [Haykin, 2001, chapter ?] and [Johnson et al., 1999, Page 325 and chapter 17].

C.1 Amplitude modulation

AM is defined as a process in which the amplitude of the carrier wave fc is varied
about a mean value, linearly with the baseband signal fm . If the carrier- and
baseband-signal are given as

fc = Ac cos(ωc t) (C.1)
fm = m(t) (C.2)

then the modulated signal can be described in its general form as

s(t) = Ac [1 + ka m(t)] cos(ωc t) (C.3)

11
C.1. AMPLITUDE MODULATION

Carrier wave

Amplitude
1
0
−1
0 2 4 6 8 10 12
t [s]
Modulation signal
Amplitude

1
0
−1
0 2 4 6 8 10 12
t[s]
Amplitude modulated signal
Amplitude

2
0
−2
0 2 4 6 8 10 12
t[s]
Frequency modulated signal
Amplitude

1
0
−1
0 2 4 6 8 10 12
t[s]

Figure C.1: Illustration of the carrier wave, baseband signal, amplitude modulated
signal and frequency modulated signal

Amplitude modulated signal Ka = 0.5

2
Amplitude

−2
0 2 4 6 8 10 12
t[s]
Amplitude modulated signal Ka = 1
2
Amplitude

−2
0 2 4 6 8 10 12
t[s]
Amplitude modulated signal Ka = 2
5
Amplitude

−5
0 2 4 6 8 10 12
t[s]

Figure C.2: Amplitude modulation of a signal using three different values of ka .

Values above 1 result in over modulation and phase reversal

Where ka is a constant called the amplitude sensitivity. In equation C.3 it is clear

that the value of ka m(t) determines the amplitude of the modulated signal. The
value of ka m(t) must therefore be chosen suitable in order to avoid over modulation
and phase reversal. Figure C.2 illustrate at AM signal with three different values

12
 APPENDIX C. MODULATION

of ka and
m(t) = sin(1 · t) (C.4)
The upper graph illustrate the situation with ka = 0.5 where the original signal
m(t) is clearly represented as a envelope of the AM signal. The middle graph
illustrate the situation with ka = 1. As with ka = 0.5 the original signal is clearly
represented as a envelope, but the modulation signal now completely eliminate the
carrier signal during it’s minimum level. The bottom graph illustrate the situation
with ka = 2 and the baseband signal is no longer an envelope of the AM signal,
hence it follows that to avoid over modulation and phase reversal
|ka m(t)| < 1. (C.5)
An other criteria for the modulation to be successful is
fc >> W (C.6)
where W is the highest frequency component of the baseband signal m(t). If this
criteria is not fulfilled it will not be possible to visualize an envelope. The criteria
is illustrated in Figure C.3, where the graphs on the left illustrate the AM using
a carrier frequency 20 times as fast as the baseband signal, resulting in a clear
envelope of the baseband, as in the bottom left graph. To the right a carrier
frequency only twice as fast as the baseband signal is used, and the baseband
signal can hardly be recognized in the AM signal as in the bottom right graph.

Carrier wave (20 ω/s) Carrier wave (2 ω/s)

1 1
Amplitude

Amplitude

0.5 0.5
0 0
−0.5 −0.5

0 2 4 6 8 10 12 0 2 4 6 8 10 12
t [s] t [s]
Modulation signal (1 ω/s) Modulation signal (1 ω/s)
Amplitude

Amplitude

0.5 0.5
0 0
−0.5 −0.5

0 2 4 6 8 10 12 0 2 4 6 8 10 12
t[s] t[s]
Amplitude modulated signal Amplitude modulated signal
1
Amplitude

Amplitude

1
0
0
−1 −1

0 2 4 6 8 10 12 0 2 4 6 8 10 12
t[s] t[s]

Figure C.3: Amplitude modulation using two different carrier frequencies.

In order to determine the frequencies contained in the modulated signal equation

C.3 is rewritten, and m(t) is replaced by a sinusoidal wave.

s(t) = Ac cos(ωc t) + Ac ka cos(ωc t) sin(ωm t) (C.7)

13
C.2. FREQUENCY MODULATION

From equation C.7 it can be seen that the modulated signal contains the carrier
frequency fc and the frequencies of the product

Ac ka cos(ωc t) sin(ωm t). (C.8)

In order to determine the frequencies of the product, the product is rewritten by

Euler.

Ac ka  jωc t  
e + e−jωc t · ejωm t − e−jωm t (C.9)
j4
Ac ka jωc t jωm t
 
e ·e − ejωc t · e−jωm t + e−jωc t · ejωm t − e−jωc t · e−jωm t (C.10)
j4
Ac ka  j(ωc +ωm )t 
e − ej(ωc −ωm )t + ej(−ωc +ωm )t − ej(−ωc −ωm )t (C.11)
j4
Ac ka  
sin((ωc + ωm )t) + sin((−ωc + ωm )t) (C.12)
2
Equation C.12 shows that the frequencies contained in the product is (ωc + ωm )
and (−ωc + ωm ). Figure C.4 shows the frequency spectrum of a AM signal where

fc = cos(2 · π · 200 · t)
fm = sin(2 · π · 5 · t)
ka = 0.5

Figure (a) shows that the three frequencies exist as both positive and negative
frequencies. This means that the frequency (−ωc + ωm ) also exist as a positive
frequency (ωc − ωm ) and (ωc + ωm ) exist as a negative frequency (−ωc − ωm ). In
practice negative frequencies does not exist, and only the positive frequencies is left
as shown in figure C.4 (b). Figure (b) shows that the side frequencies (ωc + ωm )
and (ωc − ωm ) is placed on each side of the carrier frequency with a dictance of fm

C.2 Frequency modulation

FM is defined as a process in which the frequency of the carrier wave fc is varied
about a mean value, as a function of the amplitude for the baseband signal fm . If
the modulation signal is a sinusoidal signal defined as

m(t) = Am cos(ωm t) (C.13)

then the instantaneous frequency is defined by

fi (t) = fc + kf Am cos(ωm t) (C.14)

fi (t) = fc + ∆f cos(ωm t) (C.15)

Where

∆f = kf Am (C.16)

14
 APPENDIX C. MODULATION

4 (a) 4 (b)
x 10 x 10
5 5

4.5 4.5

4 4

3.5 3.5

3 3
S(f)

S(f)
2.5 2.5

2 2

1.5 1.5

1 1

0.5 0.5

0 0
−400 −200 0 200 400 185 190 195 200 205 210
frequency (Hz) frequency (Hz)

Figure C.4: (a) Spectrum of the AM signal (b) Detailed spectrum of the positive
frequencies

The unit ∆f is the frequency deviation, representing the frequency variation from
the carrier frequency. The characteristic for FM is that ∆f is proportional with
the amplitude of the modulation signal, and independent of the frequency of the
modulation signal. To get a expression for how much the frequency changes over a
periode from 0 to t equation C.15 is integrated.
Z t
Θi (t) = 2π fi (τ )dτ (C.17)
0
∆f
Θi (t) = 2πfc t + sin(ωm t) (C.18)
fm
The ration of the frequency deviation ∆f , to the modulation frequency is called
the modulation index which is given as
∆f
β= (C.19)
fm
If the modulation index is substituted into equation C.18 a new expression for Θi (t)
is achieved.

Θi (t) = ωc t + β sin(ωm t) (C.20)

From equation C.20 it can be seen that the β represents the phase deviation of the
FM signal. This means that β represents the maximum variation from the angle
ωc t. The FM signal can now be expressed as
h i
s(t) = Ac cos ωc t + β sin(ωm t) (C.21)

Determining the spectrum of a FM signal is not as easy as it was for a AM signal.

This is because an FM signal modulated by a sinusoidal wave shown in equation
C.21 is not a periodic function unless the carrier signal frequency fc is a integral

15
C.2. FREQUENCY MODULATION

multiple of the modulation signal fm . A different method to determine the spectrum

is therefore necessary. By rewriting equation C.21 the following expression for s(t)
can be achieve
h i
s(t) = Re Ac ejωc t+jβ sin(ωm t) (C.22)
h i
s(t) = Re s̃(t)ejωc t (C.23)

Where s̃(t) is a complex envelope of the FM signal s(t), given by

s̃(t) = Ac ejβ sin(ωm t) (C.24)

Unlike the FM signal, the complex envelope s̃(t) is periodic over time. This means
that s̃(t) can be expanded by the complex fourier series, which is defined as

∞
X
s̃(t) = cn ejωm nt (C.25)
n=−∞

where the complex fourier coefficient cn is defined by

T
1
Z 2
cn = s̃(t)e−jωm nt dt (C.26)
T − T2

Inserting the value of s̃(t) from equation C.24 into equation C.26
Ac π jβ sin(ωn t) −jωn t
Z
cn = e ·e dt (C.27)
2π −π
Ac π jβ sin(ωn t)−jωn t
Z
cn = e dt (C.28)
2π −π
Defining a new variable x = ωm t and inserting it in equation C.28 gives

Z π
Ac
cn = ej(β sin(x)−nx) dx (C.29)
2π −π

Except for Ac equation C.29 can be recognized as the n‘th order Bessel function of
the first kind and argument β, which is commonly denoted as

π
1
Z
Jn (β) = ej(β sin(x)−nx) dx (C.30)
2π −π

This gives a new expression for cn

cn = Ac · Jn (β) (C.31)

Substituting equation C.31 in equation C.25

∞
X
s̃(t) = Ac Jn (β)ejωm nt (C.32)
n=−∞

16
 APPENDIX C. MODULATION

Now substituting the new expression for s̃(t) back into equation C.23 gives

∞
h X i
s(t) = Ac · Re Jn (β)ej(ωc +nωm )t (C.33)
n=−∞

By remembering that
h i i
Re ej(ωc +nωm )t = cos (ωc + nωm )t

(C.34)

equation C.33 can be rewritten to

∞
X h i
s(t) = Ac Jn (β) · cos (ωc + nωm )t (C.35)
n=−∞

Fourier transform of equation C.35 is done by using the cosine fouries transform
pair, thus the frequency spectrum of a FM signal equals
∞
Ac X h i
S(t) = Jn (β) · δ(f − nfm − fc ) + δ(f + nfm + fc ) (C.36)
2 n=−∞

A plot of the Jn (β) is shown in Figure C.5. The first 4 orders are plotted with β

1.2
0 order
1st order
2nd order
1 3rd order

0.8

0.6

0.4
J β
n

0.2

−0.2

−0.4

−0.6
0 5 10 15
β

Figure C.5: Besselfunktion of 0 to 3 orders as a funtion of β

values from 0 to 10, however when modulation an base band signal a fixed β is used.
Figure C.6 show the first 25 orders of the besselfunction with β = {1 5 15}. From
equation C.19 it is clear then when frequency modulation a signal tone base band
signal the frequency deviation ∆f is increased when β is increased, as illustrated
in the figure. As a result of equation C.16 the amplitude of the FM modulated

17
C.2. FREQUENCY MODULATION

0.5

J (1)
n
0

−0.5
−25 −20 −15 −10 −5 0 5 10 15 20 25
n
0.4

0.2
J (5)

0
n

−0.2

−0.4
−25 −20 −15 −10 −5 0 5 10 15 20 25
n
0.4

0.2
J (15)

0
n

−0.2

−0.4
−25 −20 −15 −10 −5 0 5 10 15 20 25
n

Figure C.6: First 25 orders of the besselfunction with 3 fixed β values

signal variate. The opposite is often the case; when frequency modulating, the base
band signal often contain multiple frequency components and the amplitude of
the modulation signal is kept constant, thereby keeping ∆f constant. When ∆f is
constant and β is increased there is no longer fm between each frequency component
in the fm signal. This result in a reduced bandwidth of the fm signal needed to
modulate a given base band signal. As β approaches infinity the bandwidth of the
fm signal approaches 2 · ∆f . This rule is knowned as Carson’s rule
 
1
BT = 2 · ∆f 1 + (C.37)
β

18
 APPENDIX D. DOWNCONVERTER

Appendix D
Downconverter

A modulated radio signal that is received through an antenna, sr (t), has two im-
portant caracteristics; a carrier frequency fc and a signal bandwidth fBW . The
bandwidth is centered around the carrier frequency as illustrated in figure D.1. In
order to restore the message signal from the modulated signal, it is conveinient
to translate the signal down in frequency by some fdis as also shown in figure
D.1. Displacement of a signal spectrum is referred to as frequency translation and
translating a signal down in frequency is referred to as frequency-down conversion.
[Haykin, 2001, page 103]

Figure D.1: Displacement of frequency spectrum.

D.1 Requirements
In this particular project the requirement to downconversion is to displace a small
spectrum FM signal so that the signal can be sampled using an AC97 compliant
sound card, described in appendix B, which yields the specifications mentioned in
table D.1. The received signal will be the one described in appendix A.1, hence the
carry frequency is approximately known and so is the bandwidth of the signal. A
quick calculation yields that it should be possible to fit the received signal into the
valid band of input frequencies of the sound card. In the design it is assumed that
the received radio channel does not have any neighbourgh channels.

19
D.2. BUILDING BLOCKS

Parameter Value Units

Received signal fc 1̃44 MHz
Received signal fBW 16 kHz
Downconverted signal fmin 20 Hz
Downconverted signal fmax 19.2 kHz

Table D.1: Specifications of input and output of the downconverter.

D.2 Building blocks

Downconverters are very common and despite the fact that they can be constructed
in many ways, they all consist of a few simple building blocks, which will be de-
scribed in the following.

D.2.1 Mixer
The mixer is essentially a product modulator, that multiplies the received signal
sr (t) with a local oscillator (LO) signal Ac cos(ωLO t). As a result of this, the
spectrum of sr (t)is moved along the frequency axis with fLO . The amplitude
of the translated signal will be Ac sr (t). Because the phase of the mixed signal
reverses whenever the received signal sr (t) crosses zero, the new spectrum has a
mirror image around the frequency fLO . In practice this means that whenever
a signal is mixed in order to translate it in frequency, two spectras are created
- each with the same bandwith but with different carrier frequencies; the one is
the received signal shifted downwards and the other is the received signal shifted
upwards in frequency. This is explained in appendix FiXme: Ref til Jes’s afsnit
om sampling and also sketched in figure D.2 and shown mathemathically in section
D.4. [Haykin, 2001, pages 94-95,103-104]

f1 = fc − fLO (D.1)
f2 = fc + fLO (D.2)

Figure D.2: Mirrors as a result of product modulation.

Refering to figure D.2, it is possible to chose fLO so that the spectrum near −f1
overlaps with the spectrum near f1 . This is called sideband overlap and basically

20
 APPENDIX D. DOWNCONVERTER

turns the signal into a mess. This is also why it is not trivial to build zero IF
receivers, i.e. converting directly to baseband. [Laskar et al., 2004, pages 32-44]

Mixers are available as premanufactured building blocks for various frequency

ranges. A mixer has 3 terminals; RF (Radio Frequency), LO (Local Oscillator) and
IF (Intermediate Frequency). The block diagram symbol of a mixer is shown in
figure D.3.

Figure D.3: Symbol of a 3-terminal mixer.

D.2.1.1 Filter
To perform image rejection on either the down converted or up converted signal,
a bandpass filter can be applied. The filter should have a midband frequency near
either f1 or f2 depending on which one is wanted, and a bandwidth equal to the
bandwidth of the signal fBW . Most modern wireless standards require 60 - 90 dB
of image rejection. [Laskar et al., 2004, page 30]

Filters are available as premanufactured building blocks, with various standard

intermediate frequencies. There are other building blocks that can be utilized in
downconverters, e.g. phase-locked loops (PLLs) and phase shifters. Since these are
not nescessary for this application, they will not be discussed further.

D.3 Block diagram

Often the design of the downconverter is closely related to the demodulator that
typically follows it, and they are often very integrated, as the downconverter does
a part of the demodulation or prepares the signal for a particular demodulation.

In this project the aim is to keep as much of the processing as possible in

the digital domain, and therefore it is desireable to capture the received signal
as unprocessed as possible, hence simplifying the downconverter. One solution is
shown in the block diagram of figure D.4.
Between the antenna and the first mixer some amplification will be nescessary
(RF AMP), however the strength of the received signal is not known FiXme: at
this point .

The downconversion is done in two steps using two mixers. At the output of
each mixer a filter is applied for mirror selectivity. To interface correctly with the
sound card described in chapter B, the filter should contain an adjustable gain

21
D.4. MATHEMATICAL ANALYSIS

Figure D.4: Block diagram of a possible downconverter solution.

D.4 Mathematical analysis

The purpose of the downconverter is to translate the frequencies of the RF signal,
in into baseband signals. Which can be sampled by an soundcard. The input signal
to the downconverter will be in the form as shown in appendix C:

h i
Ac · cos ωc · t + β · sin(ωc · t) (D.3)

At any give time the signal can be described as:

A1 · cos(ω1 · t) (D.4)

This signal is then multiplied with a cosine in a circuit know as a product mixer
this yields:

A1 · cos(ω1 · t) · A2 · cos(ω2 · t) (D.5)

By means of Euler this is:

1 1
A1 · A2 · · (ej·t·ω1 + e−j·t·ω1 ) · · (ej·t·ω2 + e−j·t·ω2 ) ⇔(D.6)
2 2
A1 · A2 h j·t·ω1 j·t·ω2 i
· e ·e + e−j·t·ω1 · ej·t·ω2 + ej·t·ω1 · e−j·t·ω2 + e−j·t·ω1 · e−j·t·ω2 ⇔(D.7)
4
A1 · A2 h 1  i
· cos[(ω1 + ω2 ) · t] + · e−j·t·ω1 · ej·t·ω2 + ej·t·ω1 · e−j·t·ω2 ⇔(D.8)
2 2
A1 · A2 h i
· cos[(ω1 + ω2 ) · t] + cos[(ω1 − ω2 ) · t] (D.9)
2
The result of the mixing is two frequency components, the sum and difference
frequencies. By bandpass filtering at (ω1 − ω2 ) the incoming signal is translated to
a lower frequency. After the filter only the difference signal will be left, as the filter
will attenuate the sum signal. Next the new signal is again mixed with a cosine at
frequency ω3 :

A1 · A2
· cos[(ω1 − ω2 ) · t] · A3 · cos(ω3 · t) (D.10)
2
The result of this mixing is:
A1 · A2 · A3 h i
· cos[(ω1 − ω2 − ω3 ) · t] + cos[(ω1 − ω2 + ω3 ) · t] (D.11)
4

22
 APPENDIX D. DOWNCONVERTER

After a bandpass or lowpass filter only the low frequency components are passed
yielding:

A1 · A2 · A3 h i
· cos (ω1 − ω2 − ω3 ) · t (D.12)
4
Remembering that the input signal was substituted with the instant case this
yields:

Ac · A2 · A3 h i
· cos (ωc − ω2 − ω3 ) · t + β · sin(ωm · t) (D.13)
4

D.5 Dimensioning
As the intermediate frequency between the mixers the frequency 70 MHz, is chosen.
The LO frequencies is determined, selecting the center frequency of the downcon-
verted signal to approximately the center frequency of the sound cards band of
valid input frequencies, 9 kHz:

fLO,1 = 75 MHz (D.14)

fLO,2 = 145 · 106 − 75 · 106 − 9 · 93 = 69.991.000 Hz (D.15)

As generators for fLO,1 and fLO,2 , laboratory RF signal generators are used.
The model names of the ones used can be found in appendix F. The mixers are
chosen regarding to the needed frequency ranges. The types used are mentioned in
table D.2.

Manufacturer Model No. LO/RF IF Units

Mini-Circuits ZFM-3 0.04 - 400 DC - 400 MHz
Mini-Circuits ZFM-15 10 - 3000 10 - 800 MHz

Table D.2: Specifications of mixers.

As the filter following the first mixer the component mentioned in table D.3 is
used. Since the attenuation in the stop band is not specified by the manufacturer
it might not be sufficient.

Manufacturer Model No. Type fc fBW Sections

Texscan (Trilithic) 3BC 70/5-3-KK BP 70 MHz 5 MHz 3

Table D.3: Specifications of bandpass filter.

To filter the output from the second mixer a bandpassfilter is needed. As center
frequency 9 kHz is chosen because it is the carrier frequency of the downconverted
signal. The bandwidth should be 16 kHz and at the nyquist frequency, 24 kHz,
the amplitude should be damped 4̃0 dB. The design of the filter is described in
appendix G.

23
D.6. SIMULATION

D.6 Simulation
The downconverter is simulated using Simulink. The reason for this is to examine
the waveforms at different points in the downconverter to improve the understand-
ing. The model will not be accurate according to the downconverter that is built,
because several parameters are left out and because the responses of the filters used
in the physical implementation are not known in detail. In the simulation only the
mixing process is taken into account, no gain considerations.

The downconverter can be modelled as shown in figure D.5. The mixers are
modelled using a product block. A Chebychew 1 bandpassfilter is selected, having
an order of 3, a passband between 424.115 · 106 rad/sec and 455.531 · 106 rad/sec
and a passband ripple of max. 0.5 dB. For the second filter a Bessel lowpass filter
is selcted, having an order of 9 and a critical frequenzy of 376.99 · 106 rad/sec. The
signal generators SG1, LO1 and LO2 all produce sine waves with amplitudes of 1
and frequencies of 145 MHz, 75 MHz and 69.991 MHz respectively. The simulation
parameters are shown in table D.4.

Tab Parameter Value

Solver Stop time 2e-4
Solver Type Fixed-step Dormand-Prince
Solver Fixed step size 0.2e-9
Workspace I/O Limit data points... Uncheck
Workspace I/O Decimation 1
Solver Stop time 0.0002

Table D.4: Simulation parameters that vary from default.

Figure D.5: Simulink model block diagram of downconverter.

The required sample frequency is calculated as

fs > 2 · (145 · 106 + 75 · 106 ) (D.16)

> 440 MHz (D.17)

24
 APPENDIX D. DOWNCONVERTER

due to the fact that the highest frequency is the input frequency of 145 MHz
shifted 75 MHz upwards. Converted to a maximum sampling time of 2.2727 · 10−9
s, the requirement is shown to be met by a factor of 1̃0, comparing with table D.4.

The waveforms the simulation is shown in figure D.6. Figure (a) shows the
waveforms of the signal on both sides of the bandpass filter. FiXme: fejl i legend
+ forkert sprog The dotted line shows the mixed but unfiltered signal. The curves
are not smooth due to the sampling frequency, however this does not effect the
calculation. The solid line shows the filtered signal. If the filtered signal was shown
in a longer time period it would represent a sine wave, which is the input signal
shifted downwards. This signal is mixed again, and the results of this is shown in
figure (b). Again the waveforms show that the filtering eliminate the high frequency
components and leave the downconverted signal, the solid waveform. In plot (c)
the same waveform is shown in a longer time period, revealing a sine wave with a
frequency of 9 kHz as expected.

(a)
1
Før BP
Efter BP
0.5
S(t)

−0.5
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
(b)(s)
time −8
x 10
0.4
Før LP
0.2 Efter LP
S(t)

−0.2

−0.4
0 0.5 1 1.5 2 2.5 3 3.5 4
(c)(s)
time −8
x 10
0.4
Efter LP
0.2
S(t)

−0.2

−0.4
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
time (s) x 10
−6

Figure D.6: Waveforms of simulation (a) after the first mixing, (b) after the
second mixing and (c) the resulting output.

D.7 Test
The test described in appendix F showed that the downconverter is able to translate
RF signals near 145 MHz to a low IF of 9 kHz. Furthermore the insertion loss is
meassured to 6.46 dB with an input level of -30 dBm. The insertion loss may vary
for other input levels.

25
Appendix E
Sampling

The goal of signal sampling, from the continious domain to the discrete domain, is
to produce a data representation suitable for use in computers.
This chapter will introduce the concept of sampling, and its unwanted property
of frequency aliasing. The chapter uses theory from [Oppenheim & Schafer, 1998,
Section 4].

E.1 From continuous to discrete

The discrete data representation of the continuous signal is attained, by multiplying
the input signal (xc (t)) with Diracs deltafunction (δ(t − nT )) at specific intervals.
With n as all natural numbers and T = f1s with fs as the sampling frequency. The
process is shown in figure E.1.

Figure E.1: Principal sampling block. Showing input xc (t) multiplied with δ(t −
nT ), producing the output x[n].

As δ(t − nT ) is 1 at t − nT = 0 and otherwise 0, the attained signal is:

x[n] = xc (t − nT ) (E.1)
Figure E.2 on page 29 shows the process in more steps.

E.2 Frequency attributes

Examining the preceeding process with the fourier transform, shows that “signal
mirrors” are produced in the frequency domain. This occurs because a multipli-

26
 APPENDIX E. SAMPLING

cation in time equals a convolution in frequency. This will be shown in the following.

The “sampling signal” (δ(t − nT )) can be represented as

∞
X
s(t) = δ(t − nT ) (E.2)
n=−∞

with attributes
 
1 2·π rad
fs = [Hz] Ωs =
T T s
Using the multiplication as shown in figure E.1 on the preceding page produces the
following

xs (t) = xc (t) · s(t) (E.3)

X∞
xs (t) = xc (t) · δ(t − nT ) (E.4)
n=−∞
X∞
xs (t) = xc (nT ) · δ(t − nT ) (E.5)
n=−∞

Using the fourier transfor on (E.3) produces

Xs (jΩ) = Xc (jΩ) ∗ S(jΩ) (E.6)

Using the convolution theorem

Z t
f (t) ∗ g(t) = f (τ )g(t − τ )dt (E.7)
0

and the fourier transform

Z ∞
F [f (t)] = F (jω) = f (t) · e−jωt dt (E.8)
−∞

produces (◦)
∞
1 X
Xs (jΩ) = Xc (jΩ − k · Ωs ) (E.9)
T
k=−∞

It is seen that Ω is a continuous variable, while Ωs determines the offset from the
original function because of the multitude of k’s. This is shown in figure E.3 on
page 30
If we examine figure E.4 on page 30 it is easily seen that if Ωs > 2 · Ωbw the
signals will overlap. This is called aliasing. To avoid aliasing, one must sample at
at speed faster than twice the bandwidth of the input signal. This is called the
Nyquist frequency

Ωnyquist > 2 · Ωbw (E.10)

27
E.2. FREQUENCY ATTRIBUTES

Still missing from this chapter:

• Show nyquist, explain
• Add explanation of the step at (◦) (and remove the cicle)
• Redraw graphs to proper scale
• first figure needs overhaul.. delta etc. . .

28
 APPENDIX E. SAMPLING

a)
1.6
Input signal

1.4

1.2

1
Value

0.8

0.6

0.4

0.2

0
0 0.2 0.4 0.6 0.8 1
Time

b)
1.6
Input signal
Sampling signal
1.4

1.2

1
Value

0.8

0.6

0.4

0.2

0
0 0.2 0.4 0.6 0.8 1
Time

c)
1.6
Sampled signal

1.4

1.2

1
Value

0.8

0.6

0.4

0.2

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

Figure E.2: a) The continous input signal b) Continous input signal multiplied
with sampling function c) Discrete time result

29
E.2. FREQUENCY ATTRIBUTES

Xs (jΩ)

k=1 k=0 k = −1

Ω
Ωs Ωs

Figure E.3: Frequency spectrum of sampled signal. The displacement by Ωs is

shown for each k.

Xs (jΩ)

k=1 k=0

Ω
Ωbw

Ωs

Figure E.4: Superimposed frequency picture

30
 APPENDIX F. MEASUREMENTS OF DOWNCONVERTER

Appendix F
Measurements of downconverter

The purpose of the measurements is to document that the downconverter is capable

of translating radio frequency signals at fRF = 145 MHz down to fIF = 9 kHz. The
bandwidth of the signal is fBW = 16 kHz. The downconverter is being tested trans-
lating a number of sinusoidal signals having fixed frequencies between fRF −0.5·fBW
and fIF + 0.5 · fBW . The frequency of the output signals is measured with an os-
cilloscope, and it is verified that the waveforms have the correct frequency and form.

In the test the filter mentioned in table F.1 is used in stead of the analog filter
described in appendix G. The filters has an attenuation of less than 1 dB in the
pass band and more than 40 dB in the stop band.

Manufacturer Model No. Type Passband Stop band Units

Mini-Circuits SLP-1.9 LP DC - 1.9 4.7 - 200 MHz

Table F.1: Specifications of lowpass filter.

F.1 Method
The test setup is shown i figure F.1. Note that the generators for the LO signals
are considered as part of system. The generator and indicators used are mentioned
in table F.2.

Symbol Type Model Manufacturer AAU-nr

SG1 Signal Generator 2022 Marconi 08158
IND1 Oscilloscope 2254A Tektronix 08388
LO1 Signal Generator 2022D Marconi 33336
LO2 Signal Generator 2022D Marconi 33337

Table F.2: Equipment used in test.

The test frequencies is chosen to cover the maximum and minimum frequencies
in a 16 kHz wide signal around 145 MHz:

31
F.2. RESULTS

Figure F.1: Test setup for downconverter.

f1 = fRF − 0.5 · fBW = 145 · 106 − 0.5 · 16 · 103 = 144.992.000 Hz (F.1)

f2 = fRF = 145 MHz (F.2)
f3 = fRF + 0.5 · fBW = 145 · 106 + 0.5 · 16 · 103 = 145.008.000 Hz (F.3)

The outcome of these three inputs should be 1 kHz, 9 kHz and 17 kHz.

1. Adjust the SG1 to a sine wave with with amplitude of -30 dBm, without
modulation.
2. Adjust the frequency of SG1 to f1 .
3. On IND1, adjust the timebase and volt input attenuator to obtain the best
accuracy.

Repeat above procedure for the three test cases.

F.2 Results
The results of the test is shown in table F.3.

Carrier frequency Output frequency Output level Time base Input attenuator
144.992 MHz 1.0 kHz 3.36 mV 200 µs 2 mV
145.000 MHz 9.0 kHz 3.36 mV 20 µs 2 mV
145.008 MHz 17.0 kHz 3.32 mV 10 µs 2 mV

Table F.3: Result of test.

The total voltage loss through the converter can be calculated as

V 
in
VLoss = 20 · Log [dB] (F.4)
Vout
The input power in Watts is given by

inputlevel
Pin = 10( 10 )
· 1mW (F.5)

32
 APPENDIX F. MEASUREMENTS OF DOWNCONVERTER

With an input level of -30 dBm this yields 1 µW, which is the equivalent of 7
mV assuming 50 Ω impedance. Inserting the results of table F.3 and the 7 mV in
equation (F.4) yields a loss of:

VLoss = 6.46 [dB] (F.6)

F.3 Conclusion
The test verifies that the downconverter is able to translate the RF signals to a low
IF of 9 kHz. The conversion loss of 6.46 might vary with different input levels due
to the non linear nature of the mixers. However it has not be possible to verify
this, as the oscilloscope is not able to measure voltages lower that the one used in
the test.

33
Appendix G
Filter

bla bla bla

34
 LIST OF CORRECTIONS

List of Corrections

FiXme: find sidetal i HF-teknik, eller en anden kilde . . . . . . . . . . . . 7

FiXme: Ref til Jes’s afsnit om sampling . . . . . . . . . . . . . . . . . . . 20
FiXme: at this point . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
FiXme: fejl i legend + forkert sprog . . . . . . . . . . . . . . . . . . . . . . 25

35