Professional Documents
Culture Documents
prepared by
Muhammad Tahir
Adeem Aslam
S. Irfan Shah
Sahar Idrees
Contents
1 Exponential Fourier Series
13
4 Amplitude Modulation
18
5 Envelope Detection
20
22
28
31
33
37
39
Experiment 1
Background
In mathematics, a Fourier series decomposes periodic functions or periodic signals into the sum
of a (possibly infinite) set of simple oscillating functions, namely sines and cosines (or complex
exponentials). The study of Fourier series is a branch of Fourier analysis.
Using Eulers Equation, and a little trickery, we can convert the standard Rectangular Fourier
Series into an exponential form. Even though complex numbers are a little more complicated
to comprehend, we use this form for a number of reasons:
1. Only need to perform one integration
2. A single exponential can be manipulated more easily than a sum of sinusoids
3. It provides a logical transition into a further discussion of the Fourier Transform
Description
In this lab, you will perform the following tasks:
1. Evaluate the fourier series coefficients using
Dn = ejn 4 sinc(n )
4
4
(1.1)
Plot the magnitude | Dn | (in volts) and phase 6 Dn (in degrees) of the first twenty-one
coefficients {n = 10, 9, ..., 10} versus frequency (in rad/sec).
2. Plot two periods of g(t) , directly i.e., by creating a vector of samples of g(t) and plotting
that vector.
3. Plot an approximation to g(t) using these first twenty-one terms of the exponential Fourier
series.
Make a function file sinc1.m using the following code listing.
Listing 1.1: Sinc function
1 function y = sinc1 (x)
5
2
3
4
5
6
7
8
9
10
k = length (x) ;
for i = 1:k
i f x ( i ) == 0
y( i ) = 1;
else
y ( i ) = s i n ( x ( i ) ) /x ( i ) ;
end
end
end
n = [ 10:10];
% s e t s up t h e v e c t o r o f i n t e g e r i n d i c e s .
z = n( pi /4) ;
Dn = 0 . 2 5 exp( i z ) . s i n c 1 ( z ) ; % s i n c 1 ( ) i s d e f i n e d above .
magDn = abs (Dn) ;
% magnitude o f F o u r i e r s e r i e s c o e f f i c i e n t s .
argDn = a n g l e (Dn) ( 1 8 0 / p i ) ;
% phase o f F o u r i e r s e r i e s c o e f f i c i e n t s .
w = 0.5n ;
stem (w, magDn) , x l a b e l ( Frequency i n rad / s e c ( u n i t s o f p i ) )
grid
t i t l e ( Magnitude o f t h e E x p o n e n t i a l F o u r i e r S e r i e s C o e f f i c i e n t s )
stem (w, argDn ) , x l a b e l ( Frequency i n rad / s e c ( u n i t s o f p i ) )
ylabel ( degrees )
grid
t i t l e ( Phase o f t h e E x p o n e n t i a l F o u r i e r S e r i e s C o e f f i c i e n t s )
(a)
(b)
n=+10
X
Dn ej nwt
(1.2)
n=10
The following code in Listing 1.3 is used to find the sum in (1.2). The output is shown in
Figure 1.2.
Listing 1.3: Approximation of g(t)
1 n = [ 10:10];
2 z = n( pi /4) ;
3 Dn = 0 . 2 5 exp( i z ) . s i n c 1 ( z ) ; % symbol . means elementbye l e m e n t
4
5
6
7
8
9
multiplication
nwo = n ( p i / 2 ) ; % d e f i n e e l e v e n f r e q u e n c i e s f o r sum
t = [ 0 : 0 . 0 1 : 8 ] ; % d e f i n e time s a m p l i n g p o i n t s
BIG = nwo t ; % c r e a t e b i g matrix t o do matrix m u l t i p l i c a t i o n f o r sum
g = Dn exp ( i BIG) ; % here s where t h e sum i s done
p l o t ( t , r e a l ( g ) ) , g r i d , x l a b e l ( Seconds )
t i t l e ( Approximation t o g ( t ) u s i n g t h e f i r s t t e n components o f t h e F o u r i e r
series )
Figure 1.2: Approximation to g(t) using the first ten components of the Fourier Series
We can approximate g(t) using the first ten components of the Fourier series. The following
code is used to generate two periods of function g(t). It uses a custom made unit step function,
u(t), a copy of which is also provided below. The ability to use the unit step function to write
piecewise functions proves extremely effective. Below is code lisitng. You will get the output in
Figure 1.3.
7
Listing 1.4: g(t) using unit step
1 g t = ( u ( t ) u ( t 1) ) + ( u ( t 4) u ( t 5) ) + u ( t 8) ;
2 pl o t ( t , gt )
3 grid , xla bel ( seconds )
4 t i t l e ( The r e a l g ( t ) )
5 a x i s ( [ 0 8 0.2 1 . 2 ] ) ;
function [ y ] = u(x)
y =0.5+0.5 s i g n ( x ) ;
end
Experiment 2
Dn =
j(1)n
n 6= 0
n=0
(n)
Background
In mathematics, the discrete Fourier transform (DFT) is a specific kind of discrete transform,
used in Fourier analysis. It transforms one function into another, which is called the frequency
domain representation of the original function (which is often a function in the time domain).
The DFT requires an input function that is discrete. Such inputs are often created by
sampling a continuous function, such as a persons voice. The discrete input function must
also have a limited (finite) duration, such as one period of a periodic sequence or a windowed
segment of a longer sequence.
Discrete Fourier Transform(DFT) decomposes a sequence of values into components of different frequencies. This operation is useful in many fields but computing it directly from the
definition is often too slow to be practical. An FFT is a way to compute the same result
more quickly, computing a DFT of N points using the definition, takes O(N 2 ) arithmetical
operations, while an FFT can compute the same result in only O(N log N ) operations.
The difference in speed can be substantial, especially for long data sets where N may be
in the thousands, the computation time can be reduced by several orders of magnitude in such
cases. This huge improvement made many DFT-based algorithms practical. FFTs are of great
importance to a wide variety of applications, from digital signal processing and solving partial
differential equations to algorithms for quick multiplication of large integers.
In particular, the DFT is widely employed in signal processing and related fields to analyze
the frequencies contained in a sampled signal, to solve partial differential equations, and to
perform other operations such as convolutions or multiplying large integers. A key enabling
factor for these applications is the fact that the DFT can be computed efficiently in practice
using a fast Fourier transform (FFT) algorithm.
8
Description
Plotting the Signal
This is the code segment for ploting the signal g(t). Vector g contains samples of function
g(t), which is formed by concatenating three individual vectors g1, g2 and g3. The result in
Figure 2.1 shows the plot of g(t) signal.
Listing 2.1: Plotting signal g(t)
1 g1 = [ 0 : 1 / 3 2 : 1 ( 1 / 3 2 ) ] ;
2 g2 = [ 1 : 1 / 3 2 : 1 ( 1 / 3 2 ) ] ;
3 g3 = [ 1 : 1 / 3 2 : 0 ( 1 / 3 2 ) ] ;
4 g = [ g1 , g2 , g3 ] ;
5 t = [ 1:1/64:1 (1/64) ] ;
6 plot ( t , g) ;
7 a x i s ( [ 1 . 5 1 . 5 1.5 1 . 5 ] )
8 grid
10
by one i.e. they are not arranged as a normal Fourier series spectrum. FFTSHIFT command
helps us to reach there. We will get a plot using FFTSHIFT command such that DC component
is at the centre, and plot gets the shape of a normal Fourier series plot.
Listing 2.3: Shifted version of FFT of g(t)
1 z1 = f f t s h i f t ( z ) ;
2 n = [1:1:128];
3 a = n65;
4 f = 0.5 a ;
5 stem ( f , abs ( z1 ) )
11
9 a = a +1;
10 end
11 n = m/ 2 :m/2 1;
12 stem ( n , abs (Dn) )
Comments
1. FFT function plot the fourier series but with the DC component.
2. FFTSHIFT shifts the DC component to the center of spectrum.
3. Magnitude of the fourier series is plotted against frequency to remove the complex part
from the fourier series.
Assignment
Given the signal in Figure 2.3:
1. Plot the signal in MATLAB for two time periods.
2. Find its Fourier series and plot it. The plot should have DC component at the centre.
12
Experiment 3
Background
The correlation of a signal with itself is called the Autocorrelation. The Autocorrelation ( )
of a real signal g(t) is defined as
Z
g ( ) =
(g(t)g(t + )) d )
(3.1)
It measures the similarity of a signal with itself. The Autocorrealtion provides valuable spectral
information which is helpful in analyzing the spectral energy density. The Energy Spectral
Density (ESD) is result of energies contributed by all the spectral components of the signal g(t)
i.e.
g (f ) = |G(f )2 |
(3.2)
An important relationship between the Autocorrelation of a signal g(t) and Energy Spectral
Density g ( ) exist that is the Energy Spectral Density(ESD) of a periodic signal is equal to
the Fourier Transform of Autocorrelation of the signal i.e.
g (f ) = g ( )
(3.3)
Description
Consider the following Figure 3.1 of signal x(n) which shows a discrete time rectangular signal
with length N=5.
The magnitude of the Fourier Transform of this signal is given below
sin wN
2
X(w) =
sin w2
Plot the discrete time domain signal x(n) in Matlab using the following piece of code.
13
(3.4)
14
1 n=[ 2:2];
2 x=[1 1 1 1 1 ] ;
3 stem ( n , x ) ;
4 a x i s ([ 5 5 0 2 ] )
5 t i t l e ( D i s c r e t e Time Domain S i g n a l )
The output graph produced should be similar to one in Figure 3.1. Using the signal x[n],
Discrete Time Domain Signal
2
1.8
1.6
1.4
1.2
1
0.8
0.6
0.4
0.2
0
5
The Autocorrelation function y[n] should be similar to the Figure 3.2. In order to plot the
Fourier Transform F [n] of the Autocorrelation function y[n] the following piece of code is used.
15
Autocorrelation Function y[n]
6
0
5
After plotting the Fourier Transform F [n] of the Autocorrelation Function y[n], find out
the Energy Spectral Density E[n] of the signal x[n] using its Fourier Transform. The Energy
Spectral Density E[n] of a signal is the magnitude square of its Fourier Transform. The following
piece of code finds out the Energy Spectral Density E[n].
Listing 3.4: Plot for Energy Spectral Density E(n) of the signal x(n)
1 N=5;
2 M=20;
3 i =1;
4 f o r k=M/ 2 :M/2
5
6
7
8
9
10
11
w=(2 p i /M) k ;
i f k==0
$ X( i )=N; $}
else
$X( i )=s i n (wN/ 2 ) / s i n (w/ 2 ) ; $}
end
i=i +1;
12 end
13 ESD=(abs (X) ) . 2 ;
14 w=(2 p i /M) [M/ 2 :M/ 2 ] ;
15 stem (w, ESD)
16
30
25
20
15
10
0
4
Figure 3.3 representing the Fourier Transform F [n] of the Autocorrelation Function y[n]
and Energy Spectral Density E[n] of the signal x[n] represented by Figure 3.4 are similar to
each other thus showing that the Energy Spectral Density(ESD) of a periodic signal is equal
to the Fourier Transform of Autocorrelation of the signal. Plot all the figures on the same
window using the subplot command for the ease of comparison. Vary the period and range of
the sampling points separately to observe their effects in Matlab.
Listing 3.5: Plot for both the figures on the same window
1 n=[ 2:1:2];
2 x=[1 ,1 ,1 ,1 ,1];
3 figure (1) ;
4 stem ( n , x )
5 a x i s ([ 3 3 0 1 . 5 ] ) ;
6 t i t l e ( Gate Function ) ;
7 y=x c o r r ( x , x ) ;
8 ny = [ 4 : 1 : 4 ] ;
9 figure (2) ;
10 stem ( ny , y )
11 a x i s ([ 5 5 0 6 ] ) ;
12 t i t l e ( Auto C o r r e l a t i o n ) ;
13 M=20;
14 k=M/ 2 :M/ 2 ;
15 w=(2 p i /M) k ;
16 Y=y ( exp (2 j p i /M) ) . ( ny k ) ;
17 f i g u r e ( 3 ) ;
17
ESD of Actual Signal
30
25
20
15
10
0
4
Assignment
Experiment 4
Amplitude Modulation
Objective
The objective of this experiment is to build a simple unbalanced Amplitude Modulator.
Background
Read Sections 4.2 and 4.3 of the text [Lathi(1998)] for theory background of the experiment.
Description
Before you build the circuit you need to make an inductor of approximately 22uH. The Appendix
at the end of this handout can be used as a guide to design your inductor. Build the circuit
in Figure 4.1 to implement an unbalanced Amplitude Modulator. In the circuit shown, V1 and
V2 are two sinusoidal sources for generating the message and carrier signals respectively. The
R1-V1-V2 network adds the carrier and message signal. The diode D1 is the nonlinear device
used to achieve modulation, while the network C1-L1-R4 is the band-pass (BP) filter. The
inductor developed will not have exactly the same inductance for which it is designed and hence
the resonant frequency of the BP filter will be different. Change the frequency of V2 source
and find out at what frequency the signal is maximum across the BP filter. This will be the
resonant frequency of the BP filter.
Note that the resistor R4 is not required in the actual hardware implementation. It was
used in the simulation to control the resistance of the inductor. You should be able to realize
that changing R4 affects the Q-factor of the band-pass filter.
r 2 n2
9r + 10l
In (4.1),
L is inductance in micro-Henry,
r is coil radius in inches (center of coil to center of conductor),
n is number of turns,
l is coil length in inches (center of starting turn to center of ending turn).
18
(4.1)
19
R2
10k
V1
D1
SINE(0 .7 1K)
R3
R1
20k
1N4148
C1
.03
10k
L1
22
V2
R4
.001k
SINE(0 3 200K)
.tran 0 1s 0
This equation is generally accurate to around one percent for inductors of common dimensions. It is more convenient to work with coil diameter and (4.1) can be written as:
L=
d2 n2
18r + 40l
(4.2)
Experiment 5
Envelope Detection
Objective
To demonstrate envelope detection of AM signals.
Background
In an envelope detector, the output of the detector follows the envelope of the modulated signal.
The simple circuit shown in figure functions as an envelope detector. On the positive cycle of
the input signal, the input grows and may exceed the charged voltage on the capacity, turning
on the diode and the capacitor C to charge up the peak voltage of the input signal cycle. As
the input falls below this peak value, it falls quickly below the capacitor voltage(which is nearly
the peak voltage),thus causing the diode to open. The capacitor now discharges through the
resistor R at a slow rate(with a time constant RC ). During the next positive cycle,the same
drama repeats. As the input signal rises above the capacitor voltage,the diode conducts again.
The capacitor again charges to the peak value of this (new)cycle. The capacitor discharges
slowly during the cutoff period.
During each positive cycle,the capacitor charges up to the peak voltage of the input signal and
then decays slowly until the next positive cycle. Thus,the output voltage closely follows the
(rising)envelope of the input AM signal. Further knowledge about envelope detectors can be
found from topic 4.3 of the text book.
20
21
Description
1. Generate AM signal (D1,R1,R2,C1,L1) with modulation index less than 1 i.e. < 1.
2. Build the envelope detector (D2,C2,R3) as shown in figure 1 below.
3. Display the demodulated output of the envelope detector on the oscilloscope.
4. Compare the message signal to the demodulated signal on the oscilloscope.
5. Draw the message, modulated and the demodulated signal on the next page.
6. Investigate the effect of variation of varying the message frequency and modulation index.
Assignment
Suppose that we have a Single Side Band Signal with a Carrier (SSB+C) of the form
= Acoswt + [m(t)coswt + mh(t)sinwt]
Show that the envelope of such a signal is given by e(t)=A+m(t).
Experiment 6
Description
1. Set the values of C1 =0.03uF,R1=R2=18K .
2. Find out the fmin and fmax of the VCO. To find fmin simply connect the VCO input (pin
9) to ground and to find fmax connect pin 9 to VDD .
3. Find out the free-running frequency fo of the VCO.This is the frequency of the output
signal when input is not applied to phase detector.
4. Apply an incoming signal Vi from the signal generator. Adjust its frequency to approximately match the free-running frequency f0 of the VCO. When Vi is applied, the PLL
should operate in the locked condition,with fo exactly equal to fi .The locked condition
can be easily verified by observing Vi and Vosc simultaneously on a dual-trace oscilloscope.If fi =fosc ,stable waveforms of both Vi and Vosc can be observed.Otherwise, one of
the waveforms on the scope screen is blurred or is moving with respect to the other.
5. By changing the frequency of the incoming signal,determine the actual lock range of the
PLL,i,e., determine the maximum and the minimum frequency fi such that starting from
the locked condition the PLL remains in the locked condition.The lock range should be
equal to fmax -fmin .
6. Record the readings for the lock range and the capture range below. fmin = ............
fmax = ............
fcap1 =fo -fc = ............
fcap2 =f0 +fc = ............
Lock range=fmax -fmin = ..............
Capture range=fcap2 -fcap1 = ............
1
Refer to Appendix to understand the theory and the working principle of PLL
22
23
Voltage-Controlled oscillator(VCO)
Phase detector(PD)
Low-pass loop filter(LPF)
VCO is an oscillator of the frequency of which fosc is proportional to input voltage Vo .The
input voltage to VCO determines the frequency fosc of the periodic signal Vosc at the output of
the VCO. //Phase comparator is device that compares the phase of the output signal of VCO
and the incoming signal and produces a signal proportional to the phase diference between
the incoming signal and the VCO output signal.The output of the phase detector is filtered
by a low-pass loop filter.The loop is closed by connecting the filter output to the input of the
VCO.When the loop is locked on the incoming signal Vi ,the frequency of the VCO output fosc is
exactly equal to the frequency fi of the periodic signal Vi
fosc = fi
The basic function of PLL is to maintain the frequency lock(fosc =fi ) between the input and the
output signals even if the frequency fi of the incoming signal varies with time.Assuming that
the PLL is in the locked condition and then if the frequency fi of the incoming signal increases
slightly , the phase difference between the VCO signal and the incoming signal will begin to
increase in time.As a result,the filter output voltage Vo increases, and the VCO output frequency
fosc increases until it matches fi ,thus keeping the PLL in the locked condition. //The range
of frequencies from fi =fmin to fi =fmax where the locked PLL remains in the locked condition
24
is called the lock range the PLL.If the PLL is intially locked,and fi becomes smaller than
the fmin ,or if fi exceeds fmax ,the PLL fails to keep fosc equal to fi ,and the PLL becomes
unlocked ,i.e. fosc !=fi .When the PLL is unlocked ,the VCO oscillates at the frequency fo
called the subtitle center frequency ,or the free-running frequency of the VCO .The lock can
be established again if the incoming singal frequency fi gets closed enough to fo .The range of
frequencies fi =fo -fc to fi =fo +fc such that the initially unlocked PLL becomes locked is called
the capture range of the PLL. The lock range is wider than the capture range.So,if the VCO
output frequency fosc is plotted against the incoming frequency fi ,we obtain the PLL steadystate characteristics shown in Fig 2. The characteristics simply shows that fosc =fi in the locked
condition,and that fosc =fo =constant when the PLL is unlocked. A hysteresis can be observed
in the fosc (fi ) characteristic because the capture range is smaller than the lock range.
Phase Detector
The phase detector on the 4046 is simply an XOR logic gate,with logic low output (V =0V)
when the two inputs are both high or low,and the logic high output V =VDD )otherwise.Following
figure shows the operation of the XOR phase detector when the PLL is in the locked condition.
Vi2 and Vosc are two phase-shifted periodic square-wave signals at the same frequency
fosc = fi and with 50 percent duty cycle .The output of the phase detector is a periodic square-
25
VDD =
(6.1)
The periodic signal V (t) at the output of the XOR phase detector can be written as the Fourier
series:
V (t) = Vo +
Vk sin((4kfi )t k )
(6.2)
k=1
where Vo is the dc component of V (t),and Vk is the amplitude of the kth harmonic at the
frequency 2kfi .The dc component of the phase detector output can be found easily as the
average of V (t)over a period TI =2
Vo =
VDD
(6.3)
26
Loop filter
The output V (t) of the phase detector is filtered by an external low-pass filter.In Fig3,the
loop filter is a simple passive RC filter.The purpose of the low-pass filter is to pass the dc and
low-frequency portions of V (t) and to attenuate high-frequency ac component at frequencies
2kfi .The simple RC filter has the cut-off frequency:
fp =
1
2RC
(6.4)
The cut-off frequency should be smaller than the input frequency for the output of the filter to
be approximately equal to Vo .Vo is proportional to the phase difference between the incoming
signal Vi and the signal Vosc from the VCO and the constant of proportionality ,
KD =
VDD
pi
(6.5)
is called the gain or the sensivity of the phase detector .This expression is valid for 0
.The filter output VO as a function of the phase difference is shown in Fig 5.Note that Vo if
Vi and Vosc are in phase (=0),and that it reaches the maximum value Vo =VDD when the two
signals are exactly out of phase(=).From fig4 it is easy to see that for <0,V0 increases and
for > ,V0 decreases.Of course ,the characteristic is periodic in with period 2.The range
0 is the range where the PLL can operate in the locked condition.
27
VCO operates at the maximum frequency fmax .
The actual operating frequencies can differ significantly from the values predicted by the above
equations.So,one may need to tune the component values by experiment.
For fosc between the minimum fmin and the maximum fmax ,the VCO output frequency fosc is
ideally a linear function of the control voltage Vo .The slope
Ko =
4fosc
4VO
(6.6)
HZ
V .
fc
VDD
Ko
q
2
1 + ( fc )2
(6.7)
fp
where fp is the cut-off frequency of the filter,VDD is the supply voltage and KO is the VCO
gain.Given ko and fp this relation can be solved for fc numerically which yields an approximate
theoretical prediction for the capture range 2fc .
If the capture range is much larger than the cut-off frequency of the filter, ffpc 1,then the
expression for the capture range is simplified.
2fc
2KO fp VDD
(6.8)
Note that the capture range 2fc is smaller if the cut-off frequency fp of the filter is lower.It is
usually desirable to have a wider capture range,which can be accomplished by increasing the
cut-off frequency of the filter.On the other hand a lower cut-off fp is desirable in order to better
attenuate high frequency components of v at the phase detector output and improve noise
rejection in general.
Experiment 7
Task
Build the circuit shown next. This uses the VCO portion of the 4046 PLL.
Figure 7.1:
First, investigate it using the test input circuit that is shown in Figure 2. Find the
frequency and sketch the waveform for the three VCO input voltages shown in the table below.
From that information, deterine the FM constant , Kf, for your modulator. See the data analysis
section below for guidence in this calculation.
Second, instead of the test input circuit, use, as the input, the function generator with
the sinusoidal output listed as follows:
Frequency = 5 kHz (fm = modulating frequency)
Amplitude = 2 volts (p-p)
D.C.Offset = 5V
28
29
Figure 7.2:
Examine the time-domain signal at the VCO output. It should look similar to the plot of Figure
3. Essentially, this is a rectangular waveform with a varying frequency , i.e., a frequency that is
modulated. The maximum and minimum frequencies, fmax and fmin, can be determined using
the following formulas:
f min =
f max =
1
T1
1
T2
Write an expression for the time domain output, assuming that the output waveform is sinusoidal
like. What is the for your signal? Examine the spectra using the spectrum analyzer. (Make
the Connection to the spectrum analyzer using a high impedence scope probe). Sketch the
spectra and measure the power in signicant sidebands (powers greater than one percent of the
total transmitted power). Record this data in the table shown in the report section.
Figure 7.3:
30
Data Analysis
The FM constant, Kf , can be determined by plotting the VCOs output frequency v/s the
VCOs input voltage. This should give (approximately) a straight line, its slope is Kf in hertper-volt. You will want to conver it to radians/second-per-volt in order to write the expression
for the FM signal you generate. To find , use
= [peakmodulatingtoneamplitude/modulatingtonef requency(inHz)]kf .
An alternative method is given by
=
fmax fmin
fm
Use the following circuit for FM demodulation. It is not necessary to use the buffer and
Use the following circuit for FM demodulation. It is not necessary to use the buffer and output low pass filter.
. output low pass filter.
+10v
0.1uF
16
digital link
16
14
10k
3
CD4046
10k
10uF
Vin
V1
11
10k
C=1000pF
CD4046
4
5
8
VCO
Cx=100pF
Rx=8k
5k
VCO
5
8
11
Cx=100pF
Rx=8k
Figure 7.4:
100pF
Vout
Experiment 8
Background
Read Sections 5.3 and 5.4 of the text and working of Colpitts oscillator for theory background
of the experiment.
Description
1. Implement the circuit as shown in Figure 8.1.
2. Between Port A and Port B Mic will be placed.
3. Leave wire hanging at the output.This will act as an antenna.
4. Place a FM receiver at some distance from the antenna and hear the signal sent from the
Mic.
5. Capacitor C1 is for providing AC ground for carrier frequencies and providing high
impedance at audio frequencies so that whole signal is delivered at the base of BJT .
6. C2, C3, base-emitter capacitance BEC and collector-base capacitance CBC provide the
Tank circuit capacitances.
7. C2,C3 and BEC are fix capacitances while CBC is variable.This is because base-collector
junction is reverse biased and any change in base voltage at low frequency will change
junction capacitance and thus the oscillation frequency of the tank circuit.
8. The tank circuit oscillating frequency is found by performing AC analysis. C3 is in series
with BEC giving us an equivalent capacitance CE of their series combination. This
CE,CBC and C2 are in parallel and their sum is the equivalent capacitance CEQ of the
tank circuit.
9. From the data sheet of 2N 2222, BEC and CBC are 25pF and 8pF (mean value) respectively.Thus CEQ comes out to be equal to 25pF .
31
32
10. Substituting CEQ in the formula of the oscillatiob frequency of the tank circuit we can
find the required inductance L1 for any oscillation frequency.
11. For oscillation frequency of F M 100 channel i.e. 100M Hz the inductance is 0.1H. Note
that the capacitor C1 is not required in the actual hardware implementation.
Results
Using a receiving antenna connected directly to spectrum analyzer try to locate the FM transmitter frequency of oscillation. If that frequency is in the commercial FM band of 88 MHz to 108
MHz, then the transmission of message signal from the Mic can be listened on the commercial
FM receiver.
Experiment 9
Background
Go through section 6.1 of your textbook for complete theoretical background of Sampling. A
brief overview is given below.
In signal processing, sampling is the reduction of a continuous signal to a discrete signal.
A common example is the conversion of a sound wave (a continuous signal) to a sequence of
samples (a discrete-time signal). A sample refers to a value or set of values at a point in time
and/or space. A sampler is a subsystem or operation that extracts samples from a continuous
signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of
the continuous signal at the desired points. Nyquist Sampling theorem is of great importance
in sampling the signals.
It states that:A signal can be completely determined from its samples if they are taken at uniform intervals
each of length (less than or equal to) 1/2B where B is the bandwidth of the signal.
Description
The circuit used for this lab session uses 555 timer. 555 timer is an IC used in variety of
applications like pulse generators and oscillators. It is called 555 because it has three resistances
of 5k. It has 3 modes of operation:1. Bistable Multivibator
2. Monostable Multivibator
3. Astable Multivibator
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1
3V
triggers again. The active low output of SR Latch is disabled and transistor switches off again
thus providing path for the capacitor to charge again through R1 and R2 . In this way, it creates
a pulse waveform. The duty cycle of this waveform can be controlled by properly selecting the
values of R1 , R2 and C1 .
Experimental Tasks
1. Connect the circuit as shown in Figure 1.2.
2. 555 timer is used to generate the switching sequence for the transistor which is operated in
its saturation region. 555 timer is used in its astable mode as described above. The duty
cycle of the output square wave is set to be much higher than 50% to generate the sampled
impulses at the output. This is in accordance with the Nyquist Sampling Theorem.
3. The input to the transistor is shown in Figure 1.3.
4. The time period of such an astable output is given by the sum of high time and low time
TH =0.7(R1 + R2 ) C1
TL =0.7*R2 *C1
where TH and TL are high and low times respectively.
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10. Finally, you can design a filter to obtain the signal m(t). The filter will be a simple first
order RC low pass as shown in Figure 1.4. Choose the cut off frequency of your filter less
than the sampling frequency to obtain m(t) at the filter output.
RC <
where f s =
1
2 f s
1
Ts
Experiment 10
EE 354 Communication Systems
Lab #11
Objective
Objective
Theaobjective
this experiment
is to build a simple Pulse Width Modulator (PWM).
To build
simple of
pulse
width modulator.
Background
Background
Pulse width modulation has numerous applications. It is widely used in motor speed control,
light dimming control applications to name a few.
Pulse width modulation has numerous applications. It is widely used in motor speed conDescription
trol,light
dimming control applications to name a few.
Using the circuit diagram below, construct an astable multi-vibrator. To check the circuit
functionality you can connect a capacitor at pin 5 (labelled CV) of the chip. In this mode a
Description
constant width pulse train will be produced.
the above
mentioned
step construct
is validatedan
experimentally,
connect a sinusoidal
source
1V funcUsing Once
the circuit
diagram
below
astable multivibrator.
To check
thewith
circuit
(rms),
and 3.0V a
DC
offset added
tionality
you200Hz
can connect
capacitor
at into
pin it.
5 of the chip. In this mode a constant width pulse
Figure 10.1:
Once the above mentioned step is validated experimentally, connect a sinusoidal source with
1V RMS, 200Hz and 3V dc offset added into it. The graph of Figure 10.2 shows the modulating
and the PWM signal simultaneously.
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38
Results
The graph on the next page shows the modulating and the PWM signal simultaneously.
V(n004)
5.4V
V(n006)
4.8V
4.2V
3.6V
3.0V
2.4V
1.8V
1.2V
0.6V
0.0V
0.0ms
0.5ms
1.0ms
1.5ms
2.0ms
2.5ms
3.0ms
3.5ms
4.0ms
4.5ms
5.0ms
Experiment 11
Background
Description
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40
V(n006)
5.0V
4.5V
4.0V
3.5V
3.0V
2.5V
2.0V
1.5V
1.0V
0.5V
0.0V
0.0ms
0.4ms
0.8ms
1.2ms
1.6ms
2.0ms
2.4ms
2.8ms
3.2ms
--- D:\UET\Teaching\BSc_Courses\EE354_Communication Systems\Labs_EE354\Lab12\PPM.raw ---
3.6ms
4.0ms
Figure 11.2: The modulating signal (sine wave) and the resulting PPM output.
The circuit diagram of the PPM is shown in Figure 11.1. The first 555 timer IC is used in
its astable multivibrator mode to produce a square wave of 50% duty cycle. The second 555
timer IC is used in its monostable mode. To verify the operation of the circuit, connect 0.1uF
capacitor at pin 5 of both ICs. Doing so should produce a constant width square wave with
equal high and low times. Once, the circuit has been verified, connect a sinusoidal signal of
0.5V peak and 2.5V DC offset at pin 5 of the astable multivibrator and verify that the output
is a pulse train with constant high time but varying low time (PPM) as shown in Figure 11.2.
Bibliography
[Lathi(1998)] B.P. Lathi. Modern Digital and Analog Communication Systems 3e Osece. Oxford
University Press, 1998.
[Murray(1967)] R. Murray.
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