Documente Academic
Documente Profesional
Documente Cultură
Communication
Systems
Submitted to:
Sir Noman Aftab
Submitted by:
Awais Khalid
Registration Number:
2012-EE-418
LAB MANUAL
UNIVERSITY OF ENGINEERING AND TECHNOLOGY
LAHORE, FAISALABAD CAMPUS
Experiment
#
Experiment Title:
Page
No
4
13
22
30
41
56
List of Experiment
Instructor: Noman Aftab
Name
Reg. No
Marks / Grade
M.Awais Khalid
2012-EE-418
EXPERIMENT # 1
Low Pass Filters
Objective:
To understand the basic principles of low pass filters and make Bode plot of low pass filters
Electronics
In an electronic low-pass RC filter for voltage signals, high frequencies contained in the
input signal are attenuated but the filter has little attenuation below its cutoff
frequency which is determined by its RC time constant.
For current signals, a similar circuit using a resistor and capacitor in parallel works in a
little bit into the future. This delay is manifested as phase shift. Greater accuracy in
approximation requires a longer delay
T: 2 ms/DIV
CHN A [2 V/DIV] DC
2) Frequency (f = 4kHz)
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
CHN A [2 V/DIV] DC
3) Frequency (f = 6kHz)
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
CHN A [2 V/DIV] DC
6)
CHN B [2 V/DIV] DC
XT
T: 1 ms/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT
2. Bode Diagram:
A Bode plot is a graph of the transfer function of a linear, time-invariant system
versus frequency, plotted with a log-frequency axis, to show the system's frequency response. It
is usually a combination of a Bode magnitude plot, expressing the magnitude of the frequency
response gain, and a Bode phase plot, expressing the frequency response phase shift.
In Bode' plots, commonly encountered frequency responses have a shape that is simple. That
simple shape means that laboratory measurements can easily be discerned to have the common
factors that lead to those shapes. For example, first order systems have two straight line
asymptotes and if you take data and plot a Bode' plot from the data, you can pick out first order
factors in a transfer function from the straight line asymptotes.
2.1 Bode Plot Determination (Connection Diagram):
(Draw yourself)
150
120
|F | [d B ]
p h i [D e g ]
180
10
90
60
30
0
-5
-30
-60
-10
-90
-120
-15
-150
-180
-20
1E1
1E2
1E3
1E4
1E5
1E6
f [Hz]
Bode diagram
1.940
60
2.4 Quantitative measurement of the limit frequency of the (Low pass Filter)
First, calculate the limit frequency, f3dB.
Typical result:
Instructor: Noman Aftab
Hz
14.142
VPP ;
1.940
kHz
Lab. Exercise:
Q.1: Can a low pass filter be used as an integrator. Draw circuit diagram?
Answer:
When a low pass filter is used with a sine wave input, the output is also a sine wave. The output
will be reduced in amplitude and phase shifted when the frequency is high, but it is still a sine
wave. This is not the case for square or triangular wave inputs. For non-sinusoidal inputs the
circuit is called an integrator.
Circuit Diagram:
When I measured the voltage across the capacitor as output voltage while changing the input
frequencies, I was able to simulate a low-pass filter which only outputs the lower frequency
signals. So basic diagram of low pass filter and its response is given as:
(circuit diagram)
(Response)
Name
Reg. No
Marks / Grade
Awais Khalid
2012-EE-418
EXPERIMENT # 2
High Pass Filters
Objectives:
To study the basic principles of high pass filters and implementation of
(i)
frequency response of high pass filter
(ii)
Bandwidth, Centre frequency,
(iii)
Critical frequency
(iv)
Bode Diagram
The simple first-order electronic high-pass filter shown in Figure 1 is implemented by placing an
input voltage across the series combination of a capacitor and a resistor and using the voltage
across the resistor as an output. The product of the resistance and capacitance (RC) is the time
constant (); it is inversely proportional to the cutoff frequency fc, at which the output power is
half the input power. That is,
High-pass filters have many applications. They are used as part of an audio crossover to
direct high frequencies to a tweeter while attenuating bass signals which could interfere
with, or damage, the speaker.
Rumble filters are high-pass filters applied to the removal of unwanted sounds near to the
lower end of the audible range or below. For example, noises (e.g. footsteps, motor noises
from record players and tape decks) may be removed because they are undesired or may
overload the RIAA equalization circuit of the preamp.
High-pass filters are also used for AC coupling at the inputs of many audio amplifiers, for
preventing the amplification of DC currents which may harm the amplifier, rob the
amplifier of headroom, and generate waste heat at the loudspeakers voice coil.
Image Processing
High-pass and low-pass filters are also used in digital image processing to perform image
modifications, enhancements, noise reduction, etc., using designs done in either
the spatial domain or the frequency domain.[6] The unsharp masking, or sharpening,
operation used in image editing software is a high-boost filter, a generalization of highpass.
T: 2 ms/DIV
CHN A [2 V/DIV] DC
7) Frequency (f = 4kHz)
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
CHN A [2 V/DIV] DC
8) Frequency (f = 6kHz)
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
CHN A [2 V/DIV] DC
9) Frequency (f = 6kHz)
CHN B [2 V/DIV] DC
XT
T: 1 ms/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT
2. Bode Diagram
A Bode plot is a graph of the transfer function of a linear, time-invariant system
versus frequency, plotted with a log-frequency axis, to show the system's frequency response. It
is usually a combination of a Bode magnitude plot, expressing the magnitude of the frequency
response gain, and a Bode phase plot, expressing the frequency response phase shift.
In Bode' plots, commonly encountered frequency responses have a shape that is simple. That
simple shape means that laboratory measurements can easily be discerned to have the common
factors that lead to those shapes. For example, first order systems have two straight line
asymptotes and if you take data and plot a Bode' plot from the data, you can pick out first order
factors in a transfer function from the straight line asymptotes.
2.1 Bode Plot Determination (Connection Diagram):
180
150
120
|F | [ d B ]
p h i [D e g ]
10
90
60
30
0
-5
-30
-60
-10
-90
-120
-15
-150
-180
-20
1E1
1E2
1E3
1E4
1E5
1E6
f [Hz]
Bode diagram
1.940
30
1.940
kHz
1.414
VPP ;
1.940
kHz
Lab. Exercise:
Q.1: Can a high pass filter be used as a differentiator. Draw circuit diagram?
Answer:
The High-pass RC circuit is also known as a differentiator. Because the output voltage is directly
proportional to the derivative of the input voltage. This can be seen from following diagram:
They can be used to form a band pass filter with the conjunction of Low pass filter
Also used for audio crossover ( transfer of audio signals)
Use in the digital processing of the photos such as modification of image, noise reduction, and
enhancement and usually done either in the frequency domain or spatial domain.
Its algorithmic implementation can be used for the determination of the output samples on the
basis of input samples
The high pass filters are also used in audio amplifiers to evade lower frequency signals.
Hence delay time can be reduced by increasing the value of either capacitor or resistor or by reducing the
cut-off frequency.
When I measured the voltage across the capacitor as output voltage while changing the input
frequencies, this time the circuit acted as a high-pass filter which allowed signals with higher
frequencies to pass. This shows that we can actually use the same circuit as both a LPF and a
HPF and we just need to measure the output voltages across different circuit elements. We also
found that we could not get a filter that perfectly filtered all the frequencies we did not want. So
basic diagram of low pass filter and its response is given as:
(circuit diagram)
(Response)
Name
Reg. No
Marks / Grade
Awais Khalid
2012-EE-418
EXPERIMENT # 3
Band Pass Filters
Objectives:
To study the basic principles of band pass filters and implementation of
(v)
frequency response of band pass filter
(vi)
Bandwidth, Centre frequency,
(vii) Critical frequency
(viii) Bode Diagram
of signals that can be transferred in a system, while minimizing the interference or competition
among signals.
Acoustic
A stiff physical barrier tends to reflect a specific range of frequencies, and so acts as a band-pass
filter for transmitting sound. When music is playing in another room, the notes are easily heard,
and the notes outside this range are attenuated.
T: 2 ms/DIV
CHN B [2 V/DIV] DC
2) Frequency (f = 3.1kHz)
XT
T: 100 s/DIV
CHN B [2 V/DIV] DC
XT
p h i [D e g ]
180
150
120
|F | [ d B ]
10
90
60
30
0
-5
-30
-60
-10
-90
-120
-15
-150
-180
-20
1E1
1E2
1E3
1E4
1E5
1E6
f [Hz]
__148.85____kHz
3.1 Quantitative measurement of the Mid frequency of the (Band pass Filter)
What is the value of the frequency fm, and what is the amplitude of the output voltage U2
im in comparison to the input voltage U1?
Compare the measured results with the theoretical values.
Lab. Exercise:
Q.1: Can a band pass filter be used as a frequency controller? Draw circuit diagram?
Answer:
A particular band, or spread, or frequencies can be filtered from a wider range of mixed signals. Filter
circuits can be designed to accomplish this task by combining the properties of low-pass and high-pass
into a single filter. The result is called a band-pass filter. Creating a bandpass filter from a low-pass and
high-pass filter can be illustrated using block diagrams:
What emerges from the series combination of these two filter circuits is a circuit that will only allow
passage of those frequencies that are neither too high nor too low. Using real components, here is what a
typical schematic might look like Figure below.
Name
Reg. No
Marks / Grade
Awais Khalid
2012-EE-418
EXPERIMENT # 4
Band stop Filters
Objectives:
To study the basic principles of band stop filters and implementation of
(ix)
frequency response of band stops filter
(x)
Bandwidth, Centre frequency,
(xi)
Critical frequency
(xii) Bode Diagram
T: 2 ms/DIV
CHN B [2 V/DIV] DC
2) Frequency (f = 4kHz)
XT
T: 500 s/DIV
3) Frequency (f = 6kHz)
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
4) Frequency (f = 6kHz)
Instructor: Noman Aftab
CHN B [2 V/DIV] DC
XT
T: 1 ms/DIV
CHN B [2 V/DIV] DC
XT
150
120
|F | [ d B ]
p h i [D e g ]
180
10
90
60
30
0
-5
-30
-60
-10
-90
-120
-15
-150
-180
-20
1E1
1E2
1E3
1E4
1E5
1E6
f [Hz]
Bode diagram
The centre (mid-) frequency, fm is approx.
The bandwidth If is defined as
49.95
kHz.
99.55
kHz.
Lab. Exercise:
Q.1: Band stop filter can be used to eliminate unwanted noises. How?
Answer:
Band stop filters are commonly used to eliminate particular frequencies of noise. If you've ever dealt with
the phone company in preparing to install DSL service, they may have provided you with a filter for you
household phone.
When designing a band stop filter like this, you can choose your center frequency by picking appropriate
values for your resistor, capacitor, and inductor. It's that easy. When designing a band stop filter for any
given application, you can determine where you want your cutoff frequencies, also called roll off, and
center frequency to be located.
Q.2: Why it is called as a notch filter. Support your answer with labeled figure?
Answer:
Band stop filter used as Twin-T configuration is called notch filter. Twin-T configuration is band
stop filter constructed using two capacitive filter sections.
Circuit Diagram:
Given these component ratios, the frequency of maximum rejection (the notch frequency) can be
calculated as follows:
(response)
Bandstop filter used as notch filter with Twin-T configuration. Twin-T configuration is bandstop
filter Constructed using two capacitive filter sections.
Circuit Diagram:
Given these component ratios, the frequency of maximum rejection (the notch frequency) can be
calculated as follows:
The response of notch filter is given as under:
(Response)
Name
Reg. No
Marks / Grade
Awais Khalid
2012-EE-418
EXPERIMENT # 5
Amplitude Modulation Analysis
Objectives:
To study Amplitude modulation, implementation and applications of
(i)
Response of output frequency at various input frequencies
(ii)
Response of AF generator
(iii)
Reversing the connections
(iv)
Amplitude demodulation
(v)
Half wave rectifier
When the amplitude for the carrier oscillation is varied, then this is known as amplitude
modulation.
This form of signal modulation is the subject of this exercise.
The more important terms used in AM will be outlined in short explanations and practical
exercises.
AM can be represented mathematically as a multiplication of a carrier oscillation with the
frequency and a modulating signal with the frequency .
In the usual form of AM, which in practice is used, for instance, in long, medium and short wave
transmissions, the amplitude of the carrier is greater than that of the useful signal. Also, only
50% of the useful signal is in the two sidebands (see previous formula). This means that the main
part of the transmitter power is in the carrier. To achieve a higher power component of the useful
signal in the transmitted signal, use is made of the fact that the carrier is not really needed for the
Instructor: Noman Aftab
transmission of information. Therefore, with suitable circuits (e.g. filters) the carrier is
suppressed and only the upper (USB) and lower (LSB) sidebands remain.
f = 455kHz
2)
f= 25kHz
CHN B [1 V/DIV] AC
XT
Lab. Exercise:
Q.1: What are main effective variables in AM?
Answer:
Modulation index.
Carrier Frequency.
Modulating Frequency.
Q.2: Can we say that AM is the addition of two frequencies?
Answer:
The two frequencies are combined in a non-linear signal processing device such as vacuum tube,
transistor or diode usually called a mixer in the most application, two signal at frequencies f1 and
f2 are mixed creating to new signal one at sum of the two frequencies f1 + f2 and other at the
difference f1-f2 .so, the AM is sum and difference of two frequencies.
This form of amplitude modulation is referred to as double sideband modulation (DSB). This
form of modulation is used, for instance, in the transmission of stereo information in VHF
broadcasting.
Because of the fact that the actual useful information is transmitted twice, i.e. in the upper
sideband and lower sideband, there is consequently another form of amplitude modulation,
namely Single Sideband modulation (SSB). Here, only one of the two sidebands is transmitted
and the frequency band can be used to an optimum. SSB is used in carrier frequency techniques
in multi-channel systems in the telecommunications or in commercial short-wave transmissions.
Vpp=7.5V
T: 200 s/DIV
dT: 221.014 s
f: 4.52459 kHz
dUB: 7.28289 V
CHN A [1 V/DIV] AC
2) f = 4.5kHz
CHN B [2 V/DIV] AC
XT
Vpp = 950mV
dT: 221.014 s
f: 4.52459 kHz
dUB: 949.942 mV
XT
Modulation in DSB-SC:
Instructor: Noman Aftab
If, the demodulator has constant phase, the original signal is reconstructed by passing v(t) through an
LPF.
Vpp = 1.9mV
dT: 22.8261 s
f: 43.8095 kHz
dUA: 1.92782 V
Name
Instructor: Noman Aftab
XT
Awais Khalid
Department of Electrical Engineering,
UET Faisalabad
Reg. No
Marks / Grade
2012-EE-418
EXPERIMENT # 6
Frequency Modulation Analysis
Objectives:
To study frequency modulation, implementations and applications of
(i)
modulation process
(ii)
Output response at changing input voltage wave shapes
(iii)
Frequency deviation and phase deviation
(iv)
Frequency demodulation
1.3 FM Performance:
Bandwidth
The bandwidth of a FM signal may be predicted using:
BW = 2 (M + 1 ) fm
where M is the modulation index and
fm is the maximum modulating frequency used.
FM radio has a significantly larger bandwidth than AM radio, but the FM radio band is also
larger. The combination keeps the number of available channels about the same.
The bandwidth of an FM signal has a more complicated dependency than in the AM case (recall,
the bandwidth of AM signals depend only on the maximum modulation frequency). In FM, both
the modulation index and the modulating frequency affect the bandwidth. As the information is
made stronger, the bandwidth also grows.
Efficiency of FM
The efficiency of a signal is the power in the side-bands as a fraction of the total. In FM signals,
because of the considerable side-bands produced, the efficiency is generally high. Recall that
conventional AM is limited to about 33 % efficiency to prevent distortion in the receiver when
the modulation index was greater than 1. FM has no analogous problem.
The side-band structure is fairly complicated, but it is safe to say that the efficiency is generally
improved by making the modulation index larger (as it should be). But if you make the
modulation index larger, so make the bandwidth larger (unlike AM) which has its disadvantages.
As is typical in engineering, a compromise between efficiency and performance is struck. The
modulation index is normally limited to a value between 1 and 5, depending on the application.
Noise
FM systems are far better at rejecting noise than AM systems. Noise generally is spread
uniformly across the spectrum (the so-called white noise, meaning wide spectrum). The
amplitude of the noise varies randomly at these frequencies. The change in amplitude can
actually modulate the signal and be picked up in the AM system. As a result, AM systems are
Instructor: Noman Aftab
very sensitive to random noise. An example might be ignition system noise in your car. Special
filters need to be installed to keep the interference out of your car radio.
FM systems are inherently immune to random noise. In order for the noise to interfere, it would
have to modulate the frequency somehow. But the noise is distributed uniformly in frequency
and varies mostly in amplitude. As a result, there is virtually no interference picked up in the FM
receiver. FM is sometimes called "static free" referring to its superior immunity to random noise.
So it is concluded that
In FM signals, the efficiency and bandwidth both depend on both the maximum
modulating frequency and the modulation index.
Compared to AM, the FM signal has a higher efficiency, a larger bandwidth and better
immunity to noise.
1.4 Applications of FM :
Broadcasting
FM is commonly used at VHF radio frequencies for broadcasting of music and speech (see FM
broadcasting). Normal (analog) TV sound is also broadcast using FM. A narrow band form is
used for voice communications in commercial and amateur radio settings. The type of FM used
in broadcast is generally called wide-FM, or W-FM. In two-way radio, narrowband narrow-fm
(N-FM) is used to conserve bandwidth. In addition, it is used to send signals into space.
Magnetic Tape Storage
FM is also used at intermediate frequencies by all analog VCR systems, including VHS, to
record both the luminance (black and white) and the chrominance portions of the video signal.
FM is the only feasible method of recording video to and retrieving video from Magnetic tape
without extreme distortion, as video signals have a very large range of frequency components.
Sound
FM is also used at audio frequencies to synthesize sound. This technique, known as FM
synthesis, was popularized by early digitalsynthesizers and became a standard feature for several
generations of personal computer sound cards.
Observations(Draw Yourself)
Use one channel of the oscilloscope to measure the signal at the output of the modulator and the
second to measure the AF signal.
Lab. Exercise 1:
Q.1: How does the signal at the output of the modulator behave, if there is no signal at the input
and if an AF signal is applied at the input of the modulator?
Answer:
At output there will be only the audio signal. With AF signal applied, the output will be modulated signal.
Q.2: The value of the frequency changes every moment. Therefore it is referred to as the so-called?
Answer:
The modulating frequency.
Q.3: With the above-mentioned frequency variations a constant change takes place between
higher frequencies (frequent polarity changes) and lower frequencies (less frequent polarity
changes). Therefore, these conditions are referred to as
______the "swing" in the frequency_________________
in the case of low frequencies?
fc + fm - f
fc + fm + f
Q.4: The change between these rarefaction and compression regions follows the rhythm of
____modulating frequency.
Now change the signal shape of the AF generator from sinusoidal to rectangular.
T: 50 s/DIV
dT: 51.1775 s
f: 19.5398 kHz
dUB: 2.37485 V
CHN A [2 V/DIV] AC
CHN B [5 V/DIV] AC
XT
On the basis of the output signal, explain the term "frequency deviation" and determine
this for the case in hand:
Answer:
Frequency deviation is used in FM radio to describe the max. Instantaneous difference between
an FM modulated frequency and nominal carrier frequency.
Result:
Answer:
Because in rectangular input change in polarity is fast from high polarity to low that is why it is
suitable than sine wave or any other.
Change the amplitude of the AF signal by reducing it slowly. How does the signal change
at the output of the modulator? If the amplitude of the modulating signal is reduced?
Answer:
In frequency modulation, , modulating signals are changed by changing the frequency of
input signals, when amplitude of modulating signal is also reduced with the same time its
carrier(AF generator) input also reduced then there is no change in output.
Now change the frequency of the AF signal by slowly increasing it. How does the signal
change at the output of the modulator? When the frequency of the AF signal is increased?
Answer:
When frequency of AF generator is changing slowly, the output at modulator is also
going to change.
The modulation index is calculated using the formula:
Observations:
Use the oscilloscope to observe the signal at the output of the demodulator (NF demod).
Lab exercise 2:
Q.1: If modulation index of a signal is increased, how does bandwidth behave?
Answer:
As we know that =f/B. in case of FM tone modulation is modulation index and its inversely
proportional to bandwidth.
Q.2: While using Frequency modulation, A rectangular signal is more suitable. Why?
Answer:
As rectangular wave changes its amplitude constantly, then it shows better frequency change and
good modulation results.
Q.3: Can a Digital data be sent using modulation. How?
Answer:
Yes, digital data can be sent using digital modulation like pulse code modulation.
Name
Reg. No
Marks / Grade
Omer Farooq
2012-EE-431
EXPERIMENT # 8
PULSE AMPLITUDE MODULATION
Objective:
To design and test a Pulse Amplitude Modulator .
Apparatus Required:
Sr. No.
1.
2.
3.
4.
5.
6.
7.
Equipment
NPN Transistor (BC107)
Resistor (100 K, 4.7 K, 1 K
Capacitor (0.001F)
AFO with dc shift (0-1MHz)
CRO (0-20MHz)
RPS (0-30v)
Bread Board & Connecting Wires
Quantity
2
2
2
1
1
1
1
SPECIFICATIONS:
BC107- 50V, 1A, 3W, 300MHz
All resistors are 1/4watt carbon film resistors.
Capacitor :0.001F-ceramic capacitor.
Theory:
1.1 PULSE AMPLITUDE MODULATION (PAM):
Vcc
12Vdc
4.7k
R1
1
R3
100k
C1
0.001u
R2
100k
C2
R4
2
4.7k
0.001u
Q1
Q2
BC107
R5
(2Vpp,dc)
OUTPUT (CRO)
BC107
1k
Message Signal
1.3 Waveforms
Procedure:
1. The circuit connections are made as shown in figure.
2. The free running frequency of the astable multivibrator is measured using CRO.
3. The input sine wave (dc) is given from the AFO.
4.The PAM waveform is noted from the CRO and plotted.
Instructor: Noman Aftab
Lab Exercise:
Q.1: Differentiate between PAM, PCM & PPM.
Ans:
PAM (pulse Amplitude Modulation):
It encodes information in the amplitude of a sequence of signal pulses.
PCM (Pulse Code modulation):
It has many types like PAM, PPM and PWM etc.
PWM (Pulse Width Modulation):
It results in variation of average waveform.
Name
Reg. No
Marks / Grade
Asad Ahmad
2012-EE-408
EXPERIMENT # 9
Pulse Width Modulator
Objective:
To design and test a Pulse Width Modulator (PWM) generator circuit.
Theory:
1.1 PULSE WIDTH MODULATION (PWM):
Pulse width modulation is defined as an analog modulation technique in which the width
of each pulse is made proportional to the instantaneous amplitude of the signal at the sampling
instant.
Pulse Width modulator circuit shown is basically a monostable multivibrator with a
modulating input signal applied at pin-5. By the application of continuous trigger at pin-2, a
series of output pulses are obtained, the duration of which depends on the modulating input at
pin-5. The modulating signal applied at pin-5 gets superimposed upon the already existing
voltage (2/3) Vcc at the inverting input terminal of UC. This in turn changes the threshold level
of the UC and the output pulse width modulation takes place. The modulating signal and the
output waveform are drawn in fig. It may be noted from the output waveform that the pulse
duration, that is, the duty cycle only varies, keeping the frequency same as that of the continuous
input pulse train trigger.
Apparatus Required:
IC 555
-1No
Resistor (5.5K)
-1No
Capacitor (0.01F)
-1No
-1No
-1 No
Trigger source
-1 No
=1.1RC
0.06 ms
0.06 x 1000
1.1 x 0.01 F
0.011
= 5.45 K =5.5K
Specifications:
IC 555: 4 to 18V, -55 to 125 C
All resistors are 1/4watt carbon film resistors.
Capacitor: 0.01F-ceramic capacitor.
Circuit Diagram:
Vcc
5v
Trigger source
2
0.08ms
0.02ms
R1
5.6k
6
7
IC555
0.01u
C1
1
AFO
Vm
Instructor: Noman Aftab
(200Hz,dc,2Vpp)
IC Pin Diagram:
Ground
Trigger
8 Vcc
7 Discharge
555
Output
Reset
6 Threshold
5 Control voltage
Procedure:
1. The circuit connections are made as shown in figure.
2. The Ton and Toff of the monostable multivibrator is measured using CRO.
3. The input sine wave (dc) is given from the AFO.
4. The PWM waveform is noted from the CRO and plotted
= 0.1ms
TON
= 0.08ms; TOFF
=0.02ms
TLOW = 0.69RBC
0.02ms=0.69 x RB x 0.01F
Instructor: Noman Aftab
RB
0.02 ms
0.69 x 0.01 F
= 2.898 K ~ 3K
0.08ms
0.69 x 0.01F
RA
= 11.59K-3K
= 8.59K
Lab. Exercise:
Q.1: Differentiate between PAM, PWM & PPM.
Ans:
PAM (pulse Amplitude Modulation):
It encodes information in the amplitude of a sequence of signal pulses.
PPM (Pulse Position modulation):
It change position of modulated signal.
PWM (Pulse Width Modulation):
It results in variation of average waveform.
Name
Reg. No
Marks / Grade
Omer Farooq
2012-EE-431
EXPERIMENT # 10
Pulse Position Modulator
Objective:
To design and test a Pulse Position Modulator (PPM) generator circuit.
Theory:
Pulse-position modulation (PPM) is a form of signal modulation in which M message bits
are encoded by transmitting a single pulse in one of 2M possible time-shifts. This is repeated
every T seconds, such that the transmitted bit rate is M/T bits per second. It is primarily useful
for optical communications systems, where there tends to be little or no multipath interference.
Procedure:
1.
Waveforms:
Circuit Diagram:
Lab Exercise:
Q.1: Differentiate between PAM, PWM & PPM.
Ans:
PAM (pulse Amplitude Modulation):
It encodes information in the amplitude of a sequence of signal pulses.
PPM (Pulse Position modulation):
It change position of modulated signal.
PWM (Pulse Width Modulation):
It results in variation of average waveform.
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EXPERIMENT # 10
Introduction to Matlab
Objective:
To study MATLAB and Communication Tool Box
Theory:
Attention, the Universe! By kingdom, right wheel! This prophetic phrase is the first telegraph
message on record, sent over a 16km line by Samuel F. B. Morse in 1838. The era of electrical
communication began. In this lab, we are going to learn about the basic principles and building
blocks of telecommunication. There are many telecommunication technologies today, including
optical fibers, shielded cables, telephone wires and wireless RF transmission, in the order of
decreasing data capacity. They all have similar structures when we consider the block diagram
design.
Channel
Transmitter
Receiver
The channel will carry the
electrical signal.
The transmitter drives
The receiver gets both the signal
Unfortunately, it also
electrical signals on the
and noise. It needs to filter out
attenuates the signal (the
antenna or
the noise and amplify the signal
signal becomes weaker)
communication cables.
for further processing. One of
and introduces noise,
In our calculations, we
the easiest ways to understand
Instructor: Noman Aftab
Department of Electrical Engineering,
interference and
can assume the antenna
filtering and amplification in
distortion. UET Faisalabad
circuits is by working with the
looks like a 50 load
frequency components of the
(from howstuffworks.com)
The signal, in the forms of communication described above, is a time-varying quantity, such as
voltage (or current). We can express it in the time domain as v(t) (or i(t)). For electromagnetic
waves in a wire or in air, the signal can be expressed as a sinusoid, v(t)=Av cos(2ft + v), where
Av is the signal amplitude (in volts), f is the wave frequency (in Hertz), and v is the phase (in
radians or degrees). Waves which travel at different frequencies can be superimposed on the
same channel and still distinguished from each other. We can code information by changing
either the amplitude, Av (amplitude modulation), or the phase, v (phase modulation). Phase
modulation is sometimes interpreted as frequency modulation since a time-varying phase is
equivalent to a small change in frequency. In communication systems, it is more convenient to
express the signal expressed in the frequency domain or spectrum. Many functional blocks, such
as filters, are designed according to their frequency characteristics (also called a transfer
function). The radio spectrum is shown above. The wavelength (in meter) of each frequency
component follows the straightforward relation of =c/f, where c is the speed of light in a media.
For air or free space, c=3108m/sec. For example, a 1GHz (109 Hz) signal has wavelength of
30cm. Here are some examples of frequency usage in common household wireless
communication (mostly in the VHF and UHF bands):
Garage door openers, alarm systems, etc. - Around 40 MHz
Standard cordless phones: Bands from 40 to 50 MHz
The choice of these frequencies is quite arbitrary and is dependent on history and politics. Since
we all share the free space spectrum, it is usually controlled and regulated by a government
agency. In the United States, this is the responsibility of the FCC (Federal Communications
Commission), which reports directly to Congress. If you set up a wire or fiber network, the
Instructor: Noman Aftab
usage of frequency domain is not restricted since the signal wave is confined in the wire.
(Usually the operation frequency is chosen for minimal loss in the wire or fiber). Multiple
signals can be sent through the same channel, as long as there is some method of receiving the
signals separately. FDMA (frequency-division multiple access) allots separate frequencies to
each signal. CDMA (code-division multiple access) uses header codes to divide signals in a
wireless network. WDM (wavelength division multiplexing) uses different wavelengths of light
for each signal traveling on an optical fiber.
We will restrict ourselves to the communication of digital information (the binary system of
0 and 1, a number of 2 is represented as 10 2, 9 as 1012, and 67 as 10000112). (Should we
include something here about how to convert between binary and decimal?) This is not due to
popularity, but because of the great improvements allowed by error correction of sequential
digital information. A string of binary bits can be transmitted over a channel with errors, and
then recovered at the receiving end without error!! This is achieved using the family of
Hamming codes, and you can search further for information on how it is done. It is very
beautiful
In order to have two-way communication, we need both a transmitter and a receiver in each
participating unit (that is why they are called transceivers). Since we assume digital data (data
which is represented only by 0s and 1s) modulation and demodulation (modem) are
necessary to convert the signal to an appropriate form for transmission. We will not discuss
modulation methods, but will just go ahead with the bandwidth concepts in good faith.
Bandwidth is the difference between the upper and lower frequencies in a signal.
There are many sources of noise, interference and distortion. Lets focus on the channel for now.
One source of noise is from ambient conditions. This type of noise is usually white, i.e., the
noise power has no preference for any frequency. It is called white since it consists of all
frequencies, similar to white light which also consists of all frequencies. A non-spam channel
usually has around V noise while the signal is around 1-10mV. Another type of noise is caused
by another signal interfering the same channel, even if it is using another frequency for
transmission. Due to geometrical shielding, a wired channel has much less noise than a wireless
channel (i.e. air).
Gain (dB)
Interferin
g signal
Desired
signal
High-pass filter
White noise
In the receiver, we need to distinguish the signal from noise, and then amplify the signal for
further use. The receiver includes filters and amplifiers. Filters select a frequency range to pass.
In general, the more selective (or the higher order) the filter is, the more power is needed. The
gain of the amplifier is also proportional to power consumption. Gain, a unitless number, is
Instructor: Noman Aftab
often represented by dB (decibel), which is 10log 10(A) with A being the amplification factor. For
example, a two-time amplification is about 10log102 = 3dB. A four-time amplification is about
10log104 = 6dB. A ten-time amplification is by definition 10dB.
Communication systems alone are a specialized tradeThe component and system design for
wireless, wired and fiber networking combine to make up an industry that is close to a trillion
dollars a year (a trillion dollars in what business spec, exactly?_ (depending on how you do the
accounting). As a quantitative benchmark, a 1Gbit/sec (data rate) wireless modem in a 3cm by
3cm package, operating at 2.4GHz (operating frequency) will consume 200mW power and can
transmit signals to a receiver up to 100m away. For our VERY simple purposes, you will
estimate the bandwidth and power consumption of a communication module using the following
three rules, with the benchmark example of the wireless modem as a reference point:
The antenna rule: The antenna size (and hence the system size) needs to be larger than
one tenth of the wavelength, i.e, the lower the frequency of choice, the larger the antenna
needs to be.
The power rule: Power consumption is proportional to the frequency and proportional to
the square-root of the data rate. For example, if you use only 1000bit/sec in your design
(106 reduction from the example), the power consumption is 10 3 times smaller (i.e.,
0.2mW) at the same frequency (2.4GHz). Or if you decide to use 24GHz signal (for a
smaller system size), the power consumption will be 10 times larger for the same data
rate.
The distance rule: Power consumption is proportional to the square-root of the
transmission distance. For example, if you only need to transmit 1m instead of 100m, the
power consumption can be reduced 10 times.
We will not deal with the further complications of technology and component design. This is
just to give you a taste, and remember that communication systems take a long time to learn.
However, to give you a simple example of the components in communication systems, we will
use Simulink for receiver design practice. Simulink is an extension package to MATLAB which
uses a graphical interface for constructing block diagram representations of dynamic processes.
Block diagrams are graphical representations of processes, composed of inputs, systems, and
outputs. Simulink numerically solves the underlying equations governing such processes and
allows the user to display graphical results with ease. Engineers use computer-aided design
(CAD) software frequently to help tackle the immense design complexity.
The process you are simulating in this project is a transmitter-receiver system across a given
channel. The input and output waveforms will be examined with the Simulink scope at various
stages of the system (i.e. before and after filtering the signal). In addition, the waveforms can be
heard on the speakers, where effects such as high and low frequency noise can be discerned.
Procedure:
In this lab, you will simulate a transceiver system. You will be in control of many parameters in
your system. Almost all of the components consume some amount of power (not directly
corresponding to the benchmark of the example above, but the ideas are similar). You have to be
careful not to use more power then you have available. You can control the cutoff frequencies of
Instructor: Noman Aftab
the various filters available. In the filters, the order affects the power consumption of the filter.
A higher order filter consumes more power, but also exhibits a cleaner cutoff at the desired
frequency. In the amplifier, increased power results in increased amplification, but also in
increased noise. The final result should be a system that faithfully recreates the input signal at
the output without exceeding the power budget (or even better, by using as little power as
possible). By simulating this system, you will learn tradeoffs between power and noise, and how
system components affect the design of the overall system.
Instructions:
1. After starting MATLAB enter startup in the MATLAB command window. Simulink
should load; afterwards the CURIE library blocks and a template file should appear. The
CURIE library contains all the blocks which will be used in the project and is seen in
Figure 1.
parameter to the block and is accessible by double-clicking the block after it is onto the
template. The total amount of power consumed and available is shown on the upper left
hand corner of the template.
6. Specifying the power consumption affects dependant parameters of a given block. For
example, gain in an amplifier is a function of the power consumption. Specifying the
power determines the gain, which is visible on the block itself as seen in Figure 2.
Lab. Exercise:
Q.1: Differentiate between Matlab and other simulaton softwares
Ans:
Simulink is a graphical tool for building models and then running them. it interfaces with matlab, shares
the workspace, can be run from Matlab scripts, the models can be modified from matlab scripts if you
want to. The blocks that it provides do most of the same things that you can do in matlab, but with lots
less programming.
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EXPERIMENT # 11
Amplitude Modulation in Matlab
Objective:
To verify the principles of amplitude modulation (AM) and demodulation in Matlab
Theory:
In general, we use modulation to give the transmitted signal properties which are best
suited to the transmission channel or environment. Specifically, modulation is the process of
imparting the source information onto a bandpass signal with a carrier frequency, fc, by the
introduction of amplitude or phase perturbations or both. This bandpass signal is called the
modulated signal and the baseband source signal is called the modulating signal. At the receiver
a means to translate the higher frequencies back to the audio range is implemented and this is
demodulation.
1.1 Amplitude Modulation (AM)
Audio signals at most occupy the frequency range 0-20kHz (minimum 15km
wavelength). This range of frequencies is too low to transmit directly as electromagnetic
radiation, particularly due to the prohibitive sizes of the transmitter and receiver antennas which
would be required. (Antennas must have lengths of the order of the wavelength of the EM
radiation of interest.) Higher frequencies permit much more effective and practical transmission,
Instructor: Noman Aftab
however these lie outside the audio range. For example, AM radio broadcasting occurs at
frequencies of the order of 1MHz (e.g. frequency of 1053kHz = 1.053MHz).
In standard AM the audio signal is shifted in amplitude by adding a DC component and then
multiplied by a sinusoid at the carrier frequency, fc. The carrier frequency is much higher than
the audio frequency band.
1.2 Amplitude Demodulation:
There are a number of available techniques for demodulating AM signals. We will be using two
techniques in this laboratory. The first technique we shall use in this laboratory is envelope
detection.The advantage of an envelope detector is its simplicity. In terms of hardware the
envelope detector consists of a diode and a low pass filter.
The second technique is synchronous or coherent detection (also called product detection, and
depends crucially on the carrier sinusoid in the receiver being as close as possible in frequency
(within 10Hz or so) to the original carrier. If the two sinusoids are too different distortion will be
heard in the demodulated signal. In this technique the message signal is recovered in two stages.
In the first stage intermediary signal with a baseband component and a high frequency
component is obtained by multiplying the received signal by a sinusoid of the same frequency as
the original carrier (see the trigonometric identities). In the second stage a low pass filter is used
to remove the non-audio (high frequency) component of the intermediary signal. Thus the
resulting output signal is a reconstruction of the audio frequency message signal.
1.3 Creating M-files:
What is an m-file?
An m-file, or script file, is a simple text file where you can place Matlab commands. When the
file is run, Matlab reads the commands and executes them exactly as it would if you had typed
each command sequentially at the Matlab prompt. All m-file names must end with the extension
'.m' (e.g. plot.m). If you create a new m-file with the same name as an existing m-file, Matlab
will choose the one which appears first in the path order (help path for more information). To
make life easier, choose a name for your m-file which doesn't already exist. To see if a
filename.m exists, type help filename at the Matlab prompt.
Why use m-files?
For simple problems, entering your requests at the Matlab prompt is fast and efficient. However,
as the number of commands increases or trial and error is done by changing certain variables or
values, typing the commands over and over at the Matlab prompt becomes tedious. M-files will
be helpful and almost necessary in these cases.
Instructor: Noman Aftab
Procedure:
a) Using Command Windows
1- Create a new M-file.
2- Enter the following
clear
[y,fs,n]=wavread('hugo');
fs
n
disp('paused.
pause;
sound(y);
3- Save the file as Test.m.
4- At Matlab command prompt enter 'test'.
Amplitude Modulation
In Amplitude modulation the following complex envelope is used.
g (t ) Ac [1 m(t )]
c 2f c
What are the peak values (min and max) for m(t)?
Ans:
Peak value of m (t) is +1 and -1.
What would be the % modulation, %positive modulation and %negative modulation for this m(t)
given Ac=1?
Ans:
Modulation is a process that cause a shift in range of frequencies in signal and modulation index is ratio
of m and Ac.When modulation index is going to multiply with 100% its became % modulation.when m
is greater than 1 its positive modulation otherwise negative.
plot(s);
now, plot the AM signal.
What happen if you multiply m by 0.5. What is the %modulation in this case?
Ans:
Modulation index tell about percentage modulation and it is ratio of m and A (carrier amplitude), if we
multiply m by 0.5 then percentage modulation is 50%.
What happen if you multiply m by 1.5? What is the %modulation in this case?
Ans:
Modulation index tell about percentage modulation and it is ratio of m and A (carrier amplitude), if we
multiply m by 1.5 then percentage modulation is 150%.
Create a M-file and plot 4 figures, one for your signal m(t), one for carrier c(t),
One for the modulated signal s(t), and the last one for the spectrum for s(t).
Hint: use subplot to divide your figure.
Amplitude Modulation DSB-SC (Double sideband suppressed carrier)
Make a copy of the AM modulation M-file "DSBSC.m" from Blackboard
Make the following changes.
m cos(2 * pi * mf * t ) 2 * cos( 2 * pi * 2 * mf * t pi / 4)
s Ac * m. * m. * c
To start Simulink, first start Matlab then type 'simulink' you will got two windows one for the
library of blocks
2- Double click on the Signal wave. Change the Frequency to 5 (rad/sec) and change sample
time to 1/100.
3- Do the same for the carrier but with Frequency =1000.
Now we are done with the input. Let's link them together. We will need some math
operation so from the library browser.
4- Go to Simulink > Math Operations. Drag one Product block and one Sum block.
5- Now link the Signal and the Constant 1 to the Sum operation inputs.
6- Double click in the Product block. Make the number of inputs 3 instead of two so we can
multiply 3 blocks in one process.
7- Now link the Carrier, the AC constant and the output of Sum to the input of the
Product block.
Now wee need to add a block to show the output
8- Go to Simulink > Sinks. Drag one Scope.
This is a modified version of our system I've added a scope to see our original signal and
spectrum scope to see the spectrum of our modulated signal.
Lab. Exercise:
i.
Simulation 1
1. Visualize the spectrum output (BFFT). It can be seen that the output consists of just two
side bands at 9 kHz and 11 kHz why?
2. Effect of Variations in Modulating and Carrier frequencies on DSB SC signal
ii.
Simulation 2
The figure below show the experiment of an amplitude modulation for modulation index a = 1
and 0.5. The equation of this AM is given by:
s( t ) k m [1 a.m(t )] cos( wc )
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EXPERIMENT # 12
Instructor: Noman Aftab
Theory:
1.1 Single Sideband Modulation (SSB-AM)
Single Sideband is an improvement of the Amplitude modulation. It is more efficient in
the usage of power and bandwidth. In the Double sideband AM modulation, the output signal has
twice the bandwidth of the modulated signal. While Using Single sideband modulation we will
use bandwidth similar to the original signal.
As we know that Amplitude modulation uses two frequencies copies of the modulated signal.
(Upper and Lower sidebands). So, instead of wasting our bandwidth in sending two side bands,
SSB modulation technique will apply a filter to filter-out one of the sidebands. It also removes
the carries signal that will reduce our bandwidth and power used to tend out signal.
In the other hand, SSB modulation system will be more complex then DSB-AM. In the
transmitter we will include a new filter to filter-out one side band.
At the receiver, we will use a complex envelope to generate the original signal.
From this we conclude that we improved the AM at the cost of extra complexity.
1.2 Frequency and Phase Modulation (FM & PM)
Frequency modulation is a Simple but powerful method of modulating a signal yet it requires a
wider bandwidth than Amplitude Modulation. Frequency Modulation uses a sine wave carrier
with frequency is modulated according to the waveform of the modulating signal.
s t Ac m t cos c t m t sin c t
if m(t)= sine(2*pi*fm*t)
Where fm=1000, fs=10000;
m(t) = sin(2*pi*fm*t); and
m(t) hat is its -90 degree phase shift = sin(2*pi*fm*t pi/2)
The oscillator is = cos(2*pi*fs*t) = sin(2*pi*fs*t+pi/2)
And oscillator hat is its -90 degree phase shift =cos(2*pi*fs*t-pi/2) = sin(2*pi*fs*t)
The system sample time = 1/(4*10000*pi)
s t Ac cos 2f c t t
For phase modulation t Dp m t
Write a Matlab code (m-file) for this problem given,
m t cos(2f m t )
Ac=2;
fc=10000;
fm=1000;
Dp=5;
First set these system parameters
d=0.004;
fs = 1000000;
ns=d*fs
t=(1:ns)/fs;
Plot the modulating signal, modulated signal and the spectrum;
To generate mf(t) from mp(t) we will differentiate mp and we will get mf sin( 2f m t )
We will use the complex envelope g t e j t e j*Dp*sin( 2* * fm*t )
sf =Ac*real(g.*phase);
3- What is the relation between m(t), and f(t)?
Ans:
m(t) is message signal and f(t) is frequency modulated signal its to vary the carrier frequency
within some small range about its original value.
4- What happen when you change frequency deviation Df (take values 2,5,10)?
Ans:
Frequency deviation (f) is used in FM Radio to describe the maximum instantaneous difference
between an FM modulated frequency and the carrier frequency.
when frequency deviation is small than its bandwidth then Carson approximation about
bandwidth satisfied.