(Latest) Comm Sys Lab Mannual

Communication Systems laboratory (EE 321)
Lab. Manual Handout
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
Instructor: Noman Aftab
Department of Electrical Engineering,

UET Faisalabad
Lab. Manual Handout
Experiment
#
Experiment Title:
Low Pass Filters

Objective:
To understand the basic principals of low pass filters and make
Bode plot of low pass filters
High Pass Filters
Objectives:
To study the operation of high pass filters and implementation of
(i)
Bode plot of high pass filters
(ii)
Examining phase and frequencies
Band Pass Filters
Objectives:
To study the basic principles of band pass filters and
implementation of
(i)
frequency response of band pass filter
(ii)
Bandwidth, Centre frequency, Critical frequency
(iii)
Bode Diagram
Band Stop Filters
Objectives:
To study the basic principals of band stop filters and
implementation of
(i)
Frequency response of band stop filter
(ii)
Bandwidth, Notch frequency, Critical frequency
(iii)
Bode Diagram
(iv)
Notch filter
Amplitude Modulation
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
Frequency Modulation
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
Frequency demodulation
Page
No
4
13
22
30
41
56
List of Experiment

UET Faisalabad

Pulse amplitude Modulation
7 Objective:
To design and test a Pulse Amplitude Modulator generator circuit.
Lab. Manual Handout

66

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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
Theory and Procedure:

1.1 Introduction
A low-pass filter is a filter that passes low-frequency signals but attenuates (reduces
the amplitude of) signals with frequencies higher than the cutoff frequency. The actual amount of
attenuation for each frequency varies from filter to filter. It is sometimes called a high-cut filter,
or treble cut filter when used in audio applications. A low-pass filter is the opposite of a highpass filter, and a band-pass filter is a combination of a low-pass and a high-pass.
Low-pass filters exist in many different forms, including electronic circuits (such as a hiss
filter used in audio), digital filters for smoothing sets of data, acoustic barriers, blurring of
images, and so on. The moving average operation used in fields such as finance is a particular
kind of low-pass filter, and can be analyzed with the same signal processing techniques as are
used for other low-pass filters. Low-pass filters provide a smoother form of a signal, removing
the short-term fluctuations, and leaving the longer-term trend.

UET Faisalabad
Lab. Manual Handout
Low Pass Filter Output Response
1.2 Examples of Low Pass Filter:

Acoustic
A stiff physical barrier tends to reflect higher sound frequencies, and so acts as a low-pass filter
for transmitting sound. When music is playing in another room, the low notes are easily heard,
while the high notes are attenuated.
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
similar manner. See current divider discussed in more detail below.

Electronic low-pass filters are used to drive subwoofers and other types of loudspeakers,
to block high pitches that they can't efficiently broadcast.

Radio transmitters use low-pass filters to block harmonic emissions which might cause
interference with other communications.

The tone knob found on many electric guitars is a low-pass filter used to reduce the
amount of treble in the sound.
1.3 Ideal Filters:

An ideal low-pass filter completely eliminates all frequencies above the cutoff frequency while
passing those below unchanged: its frequency response is a rectangular function, and is a brickwall filter. The transition region present in practical filters does not exist in an ideal filter. An
ideal low-pass filter can be realized mathematically (theoretically) by multiplying a signal by the
rectangular function in the frequency domain or, equivalently, convolution with its impulse
response, a sinc function, in the time domain.
1.4 Real Filters:
Real filters for real-time applications approximate the ideal filter by truncating
and windowing the infinite impulse response to make a finite impulse response; applying that
filter requires delaying the signal for a moderate period of time, allowing the computation to a

UET Faisalabad
Lab. Manual Handout
little bit into the future. This delay is manifested as phase shift. Greater accuracy in
approximation requires a longer delay
1.5 Low Pass Filter (Connection Diagram):
1.6 Low Pass Filter Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout
1.7 Readings and Measurements by Oscilloscope: (Draw Yourself)

1) Frequency (f = 2kHz)
Oscilloscope Time Base = 2 ms
T: 2 ms/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT
Oscilloscope Time Base = 500us

UET Faisalabad
Lab. Manual Handout
T: 500 s/DIV
CHN A [2 V/DIV] DC
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
Frequency (f = 6kHz) Oscilloscope Time Base = 1000us

UET Faisalabad
Lab. Manual Handout
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):

UET Faisalabad
Lab. Manual Handout
2.2 Bode Plot Determination of Low Pass Filter:

Determine the limit frequency f3dB of the circuit where the output voltage has fallen by 3dB
compared to the input voltage. What is the phase angle at f = f3dB?
2.3 Typical Results
(Draw yourself)

UET Faisalabad
150
120
|F | [d B ]
p h i [D e g ]
180
Lab. Manual Handout
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
The limit frequency is approximately
kHz, the phase angle is approx.
60
2.4 Quantitative measurement of the limit frequency of the (Low pass Filter)
First, calculate the limit frequency, f3dB.
Typical result:

UET Faisalabad

1.940
Lab. Manual Handout
Hz
Now, calculate the value of the output voltage for f = f3dB

Result:
14.142
VPP ;
where U1 is the value of input voltage as peak value VPP .

Increase the frequency until the output voltage reaches its maximum value as calculated above.
What is the value of frequency?
Result:
f3dB =
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:

UET Faisalabad
Lab. Manual Handout
(Low pass filter as an itegrator)

Q# 2: How a low pass filter is beneficial while performing rectification process?
Answer:
Low pass filter reduces the amplitudes of all alternating components in the rectified waveform and
possesses the DC component.
Q#3. How we can improve the efficiency of a low pass filter?

Answer:
The Efficiency of low pass filter can be improved by increasing its order.
Conclusions and Comments:

The basic RC series circuit for a low pass filter is given as:
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)

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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
Theory and Procedure:

1.1 Introduction
A high-pass filter, or HPF, is an LTI filter that passes high frequencies well but attenuates (i.e.,
reduces the amplitude of) frequencies lower than the filter's cutoff frequency. The actual amount
of attenuation for each frequency is a design parameter of the filter. It is sometimes called a lowcut filter or bass-cut filter
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,
where fc is in hertz, is in seconds, R is in ohms, and C is in farads.

UET Faisalabad
Lab. Manual Handout
High Pass Filter Output Response

1.2 Applications / Examples of High Pass Filter
Acoustic and Audio
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.

UET Faisalabad
Lab. Manual Handout
1.3 Ideal Filters

An ideal high-pass filter completely eliminates all frequencies below the cutoff frequency while
passing those above unchanged: its frequency response is a rectangular function, and is a brickwall filter. The transition region present in practical filters does not exist in an ideal filter.
1.4 Real Filters
Real filters for real-time applications approximate the ideal filter by truncating
and windowing the infinite impulse response to make a finite impulse response; applying that
filter requires delaying the signal for a moderate period of time, allowing the computation to a
little bit into the future.
1.5 High Pass Filter (Connection Diagram)
1.6 High Pass Filter Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout
Readings and Measurements by Oscilloscope: (Draw yourself)

T: 2 ms/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT

UET Faisalabad
Lab. Manual Handout
T: 500 s/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT

UET Faisalabad
Lab. Manual Handout
T: 1 ms/DIV
CHN A [2 V/DIV] DC
CHN B [2 V/DIV] DC
XT
2. Bode Diagram
2.1 Bode Plot Determination (Connection Diagram):

UET Faisalabad
Lab. Manual Handout
2.2 Bode Plot Determination of High Pass Filter:

2.3 Typical Results (Draw yourself)

UET Faisalabad
180
150
120
|F | [ d B ]
p h i [D e g ]
Lab. Manual Handout
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
The limit frequency is approximately
kHz, the phase angle is approx.
30
2.4 Quantitative measurement of the limit frequency of (High pass Filter)

First, calculate the limit frequency, f3dB.
Typical result:
1.940
kHz
Now, calculate the value of the output voltage for f = f3dB.

Result:

UET Faisalabad
1.414
Lab. Manual Handout
VPP ;
where U1 is the value of input voltage as peak value VPP.

Set the frequency to 4kHz and decrease the frequency until the output voltage reaches the value
as calculated above. What is the value of frequency, now ?
Result:
f3dB =
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:
Q .2: How a high pass filter can be used in industrial instruments?

Answer:
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.
These can be used for the elimination of unwanted signals
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.

UET Faisalabad
Lab. Manual Handout
Q. 3: How delay time of a high pass filter can be reduced?

Answer:
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,
Hence delay time can be reduced by increasing the value of either capacitor or resistor or by reducing the
cut-off frequency.
Conclusions and Comments :

The basic RC series circuit for a low pass filter is given as:
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)

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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)
(vii) Critical frequency
(viii) Bode Diagram
Theory & Procedure:

A band pass filter is a filter that allows to pass a certain range of frequencies and attenuates the
frequencies above and below this range.
A band pass filter is basically a combination of high pass and low pass filter. The output of the
high pass filter is the input to the low pass filter and output is taken across low pass filter. In this
way band is formed. The frequencies below and above this band range are attenuated and only
those frequencies which are in the range of this band are allowed to pass .
1.1 Filter Roll Off:

An ideal band pass filter would have a completely flat pass band (e.g. with no gain/attenuation
throughout) and would completely attenuate all frequencies outside the pas band. Additionally,
the transition out of the pass band would be instantaneous in frequency. In practice, no bandpass
filter is ideal. The filter does not attenuate all frequencies outside the desired frequency range
completely; in particular, there is a region just outside the intended passband where frequencies
are attenuated, but not rejected. This is known as the filter roll off

UET Faisalabad
Lab. Manual Handout
Ideal Band Pass Filter Output Response
Band Pass Filter Output Response
1.2 Examples of Band Pass Filter:

Wireless transmission
Band pass filters are used primarily in wireless transmitters and receivers. The main function of
such a filter in a transmitter is to limit the bandwidth of the output signal to the minimum
necessary to convey data at the desired speed and in the desired form. In a receiver, a band pass
filter allows signals within a selected range of frequencies to be heard or decoded, while
preventing signals at unwanted frequencies from getting through. A band pass filter also
optimizes the signal to noise ratio.
In both transmitting and receiving applications, well-designed band pass filters, having the
optimum bandwidth for the mode and speed of communication being used, maximize the number

UET Faisalabad
Lab. Manual Handout
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.
1.3 Band Pass Filter (Connection Diagram):
Band Pass Filter Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout

Oscilloscope Time Base = 2ms
T: 2 ms/DIV
CHN A [10 V/DIV] DC
CHN B [2 V/DIV] DC
2) Frequency (f = 3.1kHz)
XT

UET Faisalabad
Lab. Manual Handout
T: 100 s/DIV
CHN A [10 V/DIV] DC
CHN B [2 V/DIV] DC
XT
2.1 Bode Diagram:

Bode Plot Determination (Connection Diagram):

UET Faisalabad
Lab. Manual Handout
Bode Plot Determination of Band Pass Filter: (Draw yourself)

Determine the centre (mid-) frequency, fm of the circuit and the bandwidth, f.

UET Faisalabad
p h i [D e g ]
180
150
120
|F | [ d B ]
Lab. Manual Handout
10
90
60
30
0
-5
-30
-60
-10
-90
-120
-15
-150
-180
-20
1E1
1E2
1E3
1E4
1E5
1E6
f [Hz]
The centre (mid-) frequency, fm is approx.___1.55__ kHz.

The bandwidth f is defined as
__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.

UET Faisalabad
Lab. Manual Handout
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.

UET Faisalabad
Lab. Manual Handout
Q.2: How a band pass filter can be used in industrial instruments?

Answer:
Band pass filters are used in all kinds of instrumentation, as well, in Sonar, even medical applications, for
example, electrocardiograms, EEGs and such. They are also widely used in optics, such as with lasers etc.
Q.3: How bandwidth of a band pass filter can be increased?

Answer:
Band width is actually a distance between maximum frequency and minimum frequency. By changing
time constant of both low pass filter and high pass filter we will either increased or decreased bandwidth
of band pass filters.
Comments & Conclusions:

(Write down any three industrial applications where u can use the circuit)
They are also widely used in optics, such as with lasers.

Color filtering is actually a band pass function.
Telephone service uses band pass filters; the audio side is roughly 250Hz-5kHz.
If you get DSL, technicians install band pass filters to split the digital and audio signals.

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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)
(xi)
Critical frequency
(xii) Bode Diagram
Theory & Procedure:

1.1 Band-stop Filter
A band-stop filter or band-rejection filter is a filter that passes most frequencies unaltered,
butattenuates those in a specific range to very low levels. It is the opposite of a band-pass filter.
A notch filter is a band-stop filter with a narrow stopband (high Q factor).
Narrow notch filters (optical) are used in raman spectroscopy, live sound reproduction (Public
Address systems, also known as PA systems) and in instrument amplifier (especially amplifiers
or preamplifiers for acoustic instruments such as acoustic guitar, mandolin, bass instrument
amplifier, etc.) to reduce or prevent feedback, while having little noticeable effect on the rest of
the frequency spectrum (electronic or software filters). Other names include 'band limit filter', 'Tnotch filter', 'band-elimination filter', and 'band-reject filter'.

UET Faisalabad
Lab. Manual Handout

UET Faisalabad
Lab. Manual Handout

UET Faisalabad
Lab. Manual Handout
1.2 Band Stop Filter (Connection Diagram):
1.3 Band Stop Filter Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout

T: 2 ms/DIV
CHN A [10 V/DIV] DC
CHN B [2 V/DIV] DC
XT

UET Faisalabad
Lab. Manual Handout
T: 500 s/DIV
CHN A [10 V/DIV] DC
CHN B [2 V/DIV] DC
XT
T: 500 s/DIV
CHN A [10 V/DIV] DC
CHN B [2 V/DIV] DC
XT

UET Faisalabad
Lab. Manual Handout
T: 1 ms/DIV
CHN A [10 V/DIV] DC
CHN B [2 V/DIV] DC
XT
2.1 Bode Diagram:

response gain, and a Bode phase plot, expressing the frequency response phase.
Bode Plot Determination (Connection Diagram):

UET Faisalabad
Lab. Manual Handout
Bode Plot Determination of Low Pass Filter:

Typical Results (Draw Yourself)

UET Faisalabad
150
120
|F | [ d B ]
p h i [D e g ]
180
Lab. Manual Handout
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?

UET Faisalabad
Lab. Manual Handout
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:

UET Faisalabad
Lab. Manual Handout
Q.3: How efficiency of a band stop filter can be increased?

Answer:
If we increased order of low pass filter and increased time constant of high pass filter we will improve
efficiency of band stop filters.
Comments & Conclusion :

The Band Stop filter is also called band-elimination, band-reject, or notch filters, this kind of filter passes
all frequencies above and below a particular range set by the component values. It can be made out of a
low-pass and a high-pass filter, just like the band-pass design, except that this time we connect the two
filter sections in parallel with each other instead of in series.
(System level block diagram of a band-stop filter.)

The response of bandstop filter is given as under:
(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:

UET Faisalabad
Lab. Manual Handout
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)

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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
Theory & Procedure:

1.1 Amplitude Modulation
The term amplitude modulation is used to define a form of signal modulation where the
amplitude of a high frequency carrier signal is varied by a low frequency modulating signal. A
basic circuit by a transistor is shown in the picture below.

UET Faisalabad
Lab. Manual Handout
Amplitude modulation is a form of oscillation modulation. The principle of oscillation

modulation is based on the fact that the parameters of a sinusoidal carrier oscillation are varied
by a transmitted signal or also useful signal, whereby the magnitude of these variations can be
defined.
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

UET Faisalabad
Lab. Manual Handout
transmission of information. Therefore, with suitable circuits (e.g. filters) the carrier is
suppressed and only the upper (USB) and lower (LSB) sidebands remain.
1.2 Double SideBand Modulation:

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.

UET Faisalabad
Lab. Manual Handout
1.3 Amplitude Modulation (Connection Diagram):
1.4 Amplitude Modulation Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout
Readings and Measurements by Oscilloscope: (Draw Yourself)

1)
f = 455kHz
UOszil = 100 mVpp
2)
f= 25kHz
UAF= 160 mVpp

UET Faisalabad

T: 20 s/DIV
Lab. Manual Handout

dT: 40.0362 s
f: 24.9774 kHz
dUA: 159.255 mV
CHN A [100 mV/DIV] AC
CHN B [1 V/DIV] AC
XT
1.5 Degree of Modulation

Retain the settings that were used in the last part of the previous exercise. Slowly reduce the
amplitude of the AF signal and then slowly increase it again.
Set the AF signal back to its initial value. Adjust the oscilloscope for the X-Y mode of display.
Connect test channel B to the signal at the output of the modulator and test channel A to the AF
signal

UET Faisalabad
Lab. Manual Handout

UET Faisalabad
Lab. Manual Handout

UET Faisalabad
Lab. Manual Handout
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:

UET Faisalabad
Lab. Manual Handout
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.
Q.3: Can we categories a modulation frequency as Pitch of voice?

Answer:
Yes, we can categories a modulation frequency as Pitch of voice. Modulating your voice means moving
the pitch up and down. By moving your pitch up and down to adjust your voice can help to raise your
voice. It adds variety, and then affects your sentence.

Modulation in which the amplitude of a carrier wave is varied in accordance with some characteristic of
the modulating signal. Amplitude modulation implies the modulation of a coherent carrier wave by
mixing it in a nonlinear device with the modulating signal to produce discrete upper and lower sidebands,
which are the sum and difference frequencies of the carrier and signal. The envelope of the resultant
modulated wave is an analog of the modulating signal. The instantaneous value of the resultant modulated
wave is the vector sum of the corresponding instantaneous values of the carrier wave, upper sideband, and
lower sideband. Recovery of the modulating signal may be by direct detection or by heterodyning.
Its has the following common types:
Single and Double SideBand Modulation:
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.

UET Faisalabad
Lab. Manual Handout
Single Sideband Suppressed Carrier Modulation

Objectives:
To study:
(i)
SSB Modulation
(ii)
SSB Demodulation
Theory & Procedure:

single-sideband modulation (SSB) or single-sideband suppressed-carrier (SSB-SC) is a refinement
of amplitude modulation that more efficiently uses transmitter power and bandwidth. Amplitude
modulation produces an output signal that has twice the bandwidth of the original baseband signal.
Single-sideband modulation avoids this bandwidth doubling, and the power wasted on a carrier, at the
cost of increased device complexity and more difficult tuning at the receiver.
Single side band generation
The two methods of SSB generation are (i) frequency discrimination method and (ii) the phase
discrimination method. The frequency discrimination method of SSB generation given in figure 1, is
based on suppressing one of the sidebands from the double-side-band suppressed carrier (DSB-SC)
modulated waveform. For a perfect SSB to be generated using this method, the band pass filter (BPF),
should have sharp cut-off, which is a difficult constraint for practical implementation, especially when
the message signal has significant components near the zero frequency.

UET Faisalabad
Lab. Manual Handout
1.3 Amplitude Modulation(SSB) (Connection Diagram):
1.4 Amplitude Modulation(SSB) Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout
Readings and Measurements by Oscilloscope:

1) f =5kHz
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

UET Faisalabad

T: 200 s/DIV
Lab. Manual Handout
dT: 221.014 s
f: 4.52459 kHz
dUB: 949.942 mV
CHN A [200 mV/DIV] AC CHN B [2 V/DIV] AC
XT
Double Side Band - Suppressed Carrier

(DSB-SC) Modulation
In AM modulation, transmission of carrier consumes lot of power. Since, only the side bands contain the
information about the message, carrier is suppressed. This results in a DSB-SC wave. A DSB-SC wave
s(t) is given by:
s(t) = m(t)Ac cos(ct)
S() = Ac/2(M( c) +M( + c))
Modulation in DSB-SC:

UET Faisalabad

Lab. Manual Handout
Here also product modulator is used , but the carrier is not added. Figure 6 shows
the spectrum of the DSB-SC signal.
If, the demodulator has constant phase, the original signal is reconstructed by passing v(t) through an
LPF.
1.2 Amplitude Modulation(DSB) (Connection Diagram):

UET Faisalabad
Lab. Manual Handout
1.4 Amplitude Modulation(DSB) Wiring Diagram (Draw Yourself):
Readings and Measurements by Oscilloscope:

f = 44kHz
Vpp = 1.9mV

UET Faisalabad

T: 20 s/DIV
Lab. Manual Handout
dT: 22.8261 s
f: 43.8095 kHz
dUA: 1.92782 V
CHN A [500 mV/DIV] AC CHN B [500 mV/DIV] AC
Name
XT
Awais Khalid
UET Faisalabad
Reg. No
Marks / Grade
Lab. Manual Handout
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
Theory & Procedure:

1.1 Frequency Modulation:
Frequency modulation is a type of modulation where the frequency of the carrier is varied in
accordance with the modulating signal. The amplitude of the carrier remains constant.
The information-bearing signal (the modulating signal) changes the instantaneous frequency of
the carrier. Since the amplitude is kept constant, FM modulation is a low-noise process and
provides a high quality modulation technique which is used for music and speech in hi-fidelity
broadcasts.
In addition to hi-fidelity radio transmission, FM techniques are used for other important
consumer applications such as audio synthesis and recording the luminance portion of a video
signal with less distortion. There are several devices that are capable of generating FM signals,
such as a VCO or a reactance modulator.

UET Faisalabad
Lab. Manual Handout
The FM can be described mathematically as follows:
1.2 Modulation index:

The frequency of the carrier signal is varied in dependence on the modulating signal.
The ratio of the frequency shift to modulation frequency is termed modulation index.
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

UET Faisalabad
Lab. Manual Handout
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.

UET Faisalabad
Lab. Manual Handout
An audio signal (top) may be carried by an AM or FM radio wave.
Control Elements And Sockets of FM Modulator/Demodulator
FM Modulator Wiring Diagram (Draw Yourself)

UET Faisalabad
Lab. Manual Handout
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.

UET Faisalabad
Lab. Manual Handout
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

UET Faisalabad
Lab. Manual Handout
in the case of high frequencies.?
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:

UET Faisalabad
Lab. Manual Handout
A rectangular AF signal is very suitable for the case in hand because...?
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:

UET Faisalabad
Lab. Manual Handout
This means that...

Answer:
Modulation index is actually relation between frequency deviation and modulation frequency.
How does the modulation index influence the value of bandwidth necessary for
transmitting a frequency-modulated signal?
The bandwidth for transmitting frequency-modulated signals is calculated with the formula:
Therefore, as the modulation index increases...
The Band width also increases..
In practice, the amplitude of the modulating signal is identical with the volume, and the
modulation frequency is a measure of the pitch of a voice or music signal.
This means that for the transmission of louder signals...

Answer:
Have low pitch of voice and greater amplitude of modulating signal.
FM Demodulator Wiring Diagram (Draw Yourself):

UET Faisalabad
Lab. Manual Handout
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.
Comments & Conclusions :

Frequancy modulation is a method of modulating a carrier wave where by modulating audio
signal cause the frequency of carrier to change.

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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):

UET Faisalabad
Lab. Manual Handout
Pulse amplitude modulation is defined as an analog modulation technique in which

the signal is sampled at regular intervals such that each sample is proportional to the amplitude
of the signal, at the instant of sampling.
Vcc
1.2 CIRCUIT DIAGRAM:
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

UET Faisalabad
Lab. Manual Handout
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.

UET Faisalabad
Lab. Manual Handout
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.
Q.2: What is the efficicency of PAM signal?

Ans:
PAM low bandwidth requirements resulting in a minimal carrier frequency which minimize the power
dissipation in a switching power amplification stage. Unfortunately, PAM is limited by requirement for
pulse amplitude accuracy.
Q.3 : What are the advantages of Time Division Multiplexxing?

Ans:
Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals over a
common signal path by means of synchronized switches at the each end of the transmission line so that
each signal appears on the line only a fraction of time in an alternating pattern.

In pulse amplitude modulation (PAM) the amplitude of train of constant width pulse is varied in
proportion to the sample value of modulating (message) signal. The pulses are usually spaced at equal
time interval.

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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
AFO with dc shift (0-1MHz)
-1No

UET Faisalabad
Lab. Manual Handout
DSO(Digital Storage Oscilloscope) -1No

RPS (0-30v)
-1 No
Trigger source
-1 No
Connecting wires and breadboard

Design for Monostable:
T
=1.1RC
0.06ms = 1.1 x R x 0.1F

R
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
(200Hz,dc,2Vpp)

UET Faisalabad
Lab. Manual Handout
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
Design for Astable (trigger source):

T
= 0.1ms
TON
= 0.08ms; TOFF
=0.02ms
TLOW = 0.69RBC
0.02ms=0.69 x RB x 0.01F

UET Faisalabad
RB
Lab. Manual Handout
0.02 ms
0.69 x 0.01 F
= 2.898 K ~ 3K
THIGH = 0.69 ( RA+RB)C

0.08ms = 0.69 x 0.01F( RA+RB)
(RA+RB) =
0.08ms
0.69 x 0.01F
RA
= 11.59K-3K
= 8.59K
Lab. Exercise:
Q.1: Differentiate between PAM, PWM & PPM.
Ans:
PPM (Pulse Position modulation):
It change position of modulated signal.
Q.2: What is the efficicency of PWM signal?

Ans:
PWM perform sampling in time rather than in amplitude in PAM. Bandwidth for PWM are typically close
to an order of magnitude higher than PAM.it only required synthesis of a few discrete output level which
can easily realized by simple high efficiency switching power stage.
Q.3: What are the advantages of Time Division Multiplexxing?

Ans:


UET Faisalabad
Lab. Manual Handout
Pulse Width Modulation (PWM):

In PWM system the width of the pulse is varied in accordance with the
instantaneous level of the modulating signal.

UET Faisalabad
Name
Reg. No
Marks / Grade
Lab. Manual Handout
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.
Connections are made as shown in the circuit diagram.
2. Check the working of 555 timer as a monostable multivibrator by giving an unmodulated

PWM signal. Verify the pulse width of output signal for the designed value.
3. By applying the PWM signal note the change in the position of the pulses i.e. PPM signal.
4. Critical amplitude of the modulating signal is that value of m(t) at which the
pulse in PPM just disappears.

UET Faisalabad
Lab. Manual Handout
Waveforms:
Circuit Diagram:

UET Faisalabad
Lab. Manual Handout
Lab Exercise:
Q.1: Differentiate between PAM, PWM & PPM.
Ans:
PPM (Pulse Position modulation):
It change position of modulated signal.
Q.2: What is the efficicency of PPM signal?

Ans:
Pulse Position Modulation (PPM) differs from PWM in that the value of each instantaneous
sample of a modulating wave is caused to vary the position in time of a pulse, relative to its
non-modulated time of occurrence. Each pulse has identical shape independent of the
modulation depth. This is an attractive feature, since a uniform pulse is simple to reproduce
with a simple switching power stage.
Q.3: What are the advantages of Time Division Multiplexxing?

Ans:

Pulse Position Modulation (PPM):
In PPM System, the position of the pulse relative to the zero reference level is

UET Faisalabad

varied in accordance with the instantaneous level of the modulating
Lab. Manual Handout
Name
Reg. No
Marks / Grade
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
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
Lab. Manual Handout
(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
Baby monitors: 49 MHz
Radio controlled airplanes: Around 72Mhz, which is different from...
Radio controlled cars: Around 75 MHz
Wildlife tracking collars: 215 to 220 MHz
Cell phones: 824 to 849 MHz
New 900-MHz or 2.4GHz cordless phones
Air traffic control radar: 960 to 1,215 MHz
Global Positioning System: 1,227 and 1,575 MHz
Deep space radio communications: 2290 MHz to 2300 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

UET Faisalabad
Lab. Manual Handout
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
Frequency (Hz) in log

scale
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

UET Faisalabad
Lab. Manual Handout
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

UET Faisalabad
Lab. Manual Handout
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.
Figure 1: CURIE library contains all the blocks required

2. To use a block simply drag and drop it onto the template file, which will contain the
transmitter-receiver system. If you have difficulty with this step, or any other step, ask a
TA. This lab was not created to be frustrating, but fun!
3. Each block has specific parameters associated with its functionality. After placing the
block onto the template file, double-click it to view the parameters. For example, the
transmitter has frequency and power consumption as changeable parameters.
IMPORTANT: after modifying a variable, DO NOT close the box with the enter key.
You must click the OK button with the mouse so that the power displays will be correctly
updated.
4. Once placed onto the template file, the blocks can be connected by signal wires. Simply
drag and hold the mouse over one output port to another input port. If you want to split a
signal, for example to send it to a scope as well as output the sound to the workspace,
hold the control key and click the wire to make it branch off. Then release the mouse
where you want the branch to connect to.
5. All active elements will consume power. Blocks containing active elements display the
amount of power currently being consumed. This power consumption is specified as a

UET Faisalabad
Lab. Manual Handout
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.
Figure 2: Power consumption affects parameters such as gain

7. To view the waveforms at any stage of the system, drag and drop the scope onto the
template. Connect the signal to be viewed to the input port of the scope.
8. After all connections have been made and all parameter values are valid check to see if
the available power is not less than 0.0 mW. The system will not simulate if the power
budget is exceeded. Press the black play button on the top toolbar to begin the simulation.
9. If the simulation is successful, double-click on the scopes in the template to view the
output waveforms. Effects such as noise and attenuation should be visible on the scope
outputs. Refer to the block details on the sound block to listen to the signal transmitted.
10. Tweak the parameters and power consumption levels to achieve the desired results at the
output. With a proper combination of filter and amplifier, the output signal should
resemble the original transmission signal fairly closely. The noise characteristics of the
channels are different and require different filtering techniques. Note that as long as the
noise is not too strong at the transmission frequency, the original signals can be recovered
regardless of how distorted the signal looks on the output end of the transmission
channel.
Block Details:
Transmitter: transmits a modulated sine wave signal at a specified frequency
frequency: frequency of the signal to be transmitted
power consumption: power consumed by transmitter; directly affects the amplitude
amplitude: output amplitude of the transmitter in Volts

UET Faisalabad
Lab. Manual Handout
Amplifier: amplifies the input signal by an amount specified by the gain

power consumption: power consumed by amplifier; directly affects the gain
gain: ratio of the input to output amplitudes of the signals. Note power consumption of 0
results in a gain of 1
High/Low Pass Filter: attempts to remove the frequency component of the input signal below /
above a certain cutoff frequency
cutoff frequency: frequency at which lower / higher frequencies are removed by the filter
power consumption: power consumed by filter; directly affected by the order
order: higher order results in steeper slopes at the cutoff frequency. Higher order results
in stronger distinction between frequency to be passed and frequency to be removed
Band Pass Filter: attempts to remove the frequency content of the input signal outside a
specified range
Upper / lower cutoff frequency: the range of frequencies the filter passes
Power consumption: power consumed by filter; directly affected by the order
Order: higher order results in steeper slopes at the cutoff frequency. Higher order results
in stronger distinction between frequency to be passed and frequency to be removed
Scope: displays the signal at any point
By default, Matlab limits the number of data points shown on the scope. To change this,
double click the scope, then click the button next to the printer button to bring up the
properties. Click the Data History tab and uncheck the box next to Limit data points
to last
To print the image of the signal, simply click the printer icon in the scope window.
Output sound to workspace: outputs a signal vector to the workspace for listening
After placing in the template, double click and enter the name for the sound in the top
dialogue box.
To play the sound, in the Matlab command window, type sound(name), where name is
the name of the sound variable and without the quotes.
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.
Q.2: What we prefer simulink in Matlab?

Ans:

UET Faisalabad

Lab. Manual Handout
MATLAB is the programming environment, you need to program in the command window or m files.
SIMULINK is used to do simulations, it has many blocks ,you just need to drag and connect them as you
need.
Q.3: What are the advantages of using Matlab?

Ans:
Advantages:
- It's very easy for a beginner in computer programming
- It comes with well-written manuals.
- Large user community, sharing free codes.

MATLAB is a high-level language and interactive environment for numerical computation, visualization,
and programming. Using MATLAB, you can analyze data, develop algorithms, and create models and
applications. The language, tools, and built-in math functions enable you to explore multiple approaches
and reach a solution faster than with spreadsheets or traditional programming languages, such as C/C++
or Java. You can use MATLAB for a range of applications, including signal processing and
communications, image and video processing, control systems, test and measurement, computational
finance, and computational biology. More than a million engineers and scientists in industry and academia
use MATLAB, the language of technical computing.

UET Faisalabad
Lab. Manual Handout
Name
Reg. No
Marks / Grade
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,

UET Faisalabad
Lab. Manual Handout
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.

UET Faisalabad
Lab. Manual Handout
How to create, save or open an m-file?

To create an m-file, choose New from the File menu and select m-file. This procedure brings up
a text editor window in which you can enter Matlab commands.
To save the m-file, simply go to the File menu and choose Save (remember to save it with the
'.m' extension). To open an existing m-file, go to the File menu and choose Open .
How to run the m-file?
After the m-file is saved with the name filename.m in the Matlab folder or directory, you can
execute the commands in the m-file by simply typing filename at the Matlab prompt.
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.
Press any key to play the wave file');
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 )]
Where Ac is a constant and m(t) is our signal.


UET Faisalabad
Lab. Manual Handout
The carrier is:

c(t ) cos(2f c t )
c 2f c
fc is the carrier frequency. The AM signal will be:

g (t ) c (t ) Ac [1 m(t )] cos( wc )
Create an M-file name it AM.m.

Type the following.
fs = 2000;
% this is the sampling frequency
d = 0.05;
% the duration of the signal.
n = fs * d;
% total number of samples
t = (1:n) / fs; %time vector
First we create our signal.
It wlll be a sinusoidal signal with frequency mf= 100hz
mf = 100;
m = cos(2 * pi * mf * t);
plot(t,m);
Save you file and type AM at Matlab commend prompt
Now create the carrier. Create another sinusoidal signal c(t) with cf =1000hz. Call it c
And plot it.

UET Faisalabad
Lab. Manual Handout
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.
Now it is time for the modulation.

Ac=1;
s = Ac*(1+m) .* c;
use .* to multiply vectors elements
plot(s);
now, plot the AM signal.

UET Faisalabad
Lab. Manual Handout
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%.

UET Faisalabad
Lab. Manual Handout
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
Type DSBSC at Matlab command prompt.

b) Using Simulink
Simulink is an extension to Matlab. It allows us to use icons and block to represent our
processes. Instead of writing codes for our process we can use a graphical interface where we
drag and drop out block into the working space. And link the blocks to gather. Then we run it.
This is a simple example of Simulink
which you can select your input.
to multiply two numbers. You got a big library from
To start Simulink, first start Matlab then type 'simulink' you will got two windows one for the
library of blocks

UET Faisalabad
Lab. Manual Handout
Amplitude Modulation with Simulink
First start new model.

We need to build the following function.
s (t ) Ac [1 m(t )] cos( wc )
Where m(t) is a sinusoidal function. So we got four inputs.

AC and 1 which are constants, and two sinusoidals for our message and carrier.
1- From our library browser we go to Simulink > Sources, we drag two constants and two
sine waves. And name them with appropriate names.
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.

UET Faisalabad
Lab. Manual Handout
9- Link the output of the Product to the Scope.

10- Now run your model and double click on the scope. You will see your modulated signal.
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.

UET Faisalabad
Lab. Manual Handout
Lab. Exercise:
i.
Simulation 1
1. Double Side-Band Suppressed Carrier Modulation

The Figure below shows the implementation of a DSB-SC signal. The Signals are at 1 kHz and 10 kHz.
Implement and simulate this modulation.

UET Faisalabad
Lab. Manual Handout
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
a. Vary the 10 KHz carrier frequency. What is the expected result?

b. Vary the modulating frequency. What is the expected result?
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 )
m(t ) cos(2 1000t )

and c 2 (10000)
Represent the signal s(t) in both time-domain and frequency domain when km=1 for a=1 and
a=0.5.

UET Faisalabad
Lab. Manual Handout
Name
Reg. No
Marks / Grade
EXPERIMENT # 12

UET Faisalabad
Lab. Manual Handout
Single Sideband and Frequency Modulation in Matlab

Objectives:
To verify the principles of Single Sideband (SSB), Frequency and Phase Modulation
(FM, PM).
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.

UET Faisalabad
Lab. Manual Handout
Phase Modulation is a special case of Frequency modulation. It is uses a carrier wave by

variation of its phase according to the waveform of the modulating signal. But the different is
that in Phase Modulation the phase is directly proportional to the modulating signal. While in
Frequency Modulation the phase is proportional to the integral of the modulation signal. Both
Frequency and phase modulation are called Angle Modulation.
1.3 Single Side Band using Simulink:

For single sideband, the modulated signal is
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)

UET Faisalabad
Lab. Manual Handout
1- What happen when you change the constant +- to -1?

Ans:
When we change constant it change output from positive amplitude to negative and shift signal.
2- What happen when you change fm, fs?

Ans:
When we change fm and fs then output signal shifted to new frequency range and side band have
new change frequency.
1.5 Phase and Frequency modulation:

In this procedure we will start with phase modulation.
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;

UET Faisalabad
Lab. Manual Handout
1- What is the relation between m (t), and sp(t)?

Ans:
m(t) is message signal spectrum and sp(t) is output signal that is product of message
signal and carrier signal and having output either lower and upper side band having
whole information of baseband signal.
2- What happen when you change phase sensitivity Dp (take values 2,5,10)?
Ans:
Dp is constant that change phase of signal and became phase modulation.
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 )
Where sf (t ) REAL e j 2* pi* fs*t g t e j 2* pi* fs*t * e j*Dp*sin( 2* * fm*t )
We got the following.

g = exp(j*Df*sin(2*pi*fm*t));
phase = exp(j*2*pi*fs*t);

UET Faisalabad
Lab. Manual Handout
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.
((((( ---------------------------- )))))

UET Faisalabad
Lab. Manual Handout

UET Faisalabad

(Latest) Comm Sys Lab Mannual

Hochgeladen von

Dokumentinformationen

Copyright

Verfügbare Formate

Dieses Dokument teilen

Dokument teilen oder einbetten

Freigabeoptionen

Stufen Sie dieses Dokument als nützlich ein?

Sind diese Inhalte unangemessen?

Copyright:

Verfügbare Formate

(Latest) Comm Sys Lab Mannual

Hochgeladen von

Copyright:

Verfügbare Formate

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Instructor: Noman Aftab

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Low Pass Filters

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Instructor: Noman Aftab

Lab. Manual Handout

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Theory and Procedure:

Instructor: Noman Aftab

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Low Pass Filter Output Response

1.2 Examples of Low Pass Filter:

similar manner. See current divider discussed in more detail below.

to block high pitches that they can't efficiently broadcast.

interference with other communications.

1.3 Ideal Filters:

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

1.5 Low Pass Filter (Connection Diagram):

1.6 Low Pass Filter Wiring Diagram (Draw Yourself):

Instructor: Noman Aftab

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

1.7 Readings and Measurements by Oscilloscope: (Draw Yourself)

Oscilloscope Time Base = 2 ms

Instructor: Noman Aftab

Oscilloscope Time Base = 500us

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Oscilloscope Time Base = 500us

Frequency (f = 6kHz) Oscilloscope Time Base = 1000us

Instructor: Noman Aftab

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Instructor: Noman Aftab

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

2.2 Bode Plot Determination of Low Pass Filter:

Instructor: Noman Aftab

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

The limit frequency is approximately

kHz, the phase angle is approx.

Department of Electrical Engineering,

Communication Systems laboratory (EE 321)

Lab. Manual Handout

Now, calculate the value of the output voltage for f = f3dB

where U1 is the value of input voltage as peak value VPP .

Instructor: Noman Aftab