INSTITUTE OF INDUSTRIAL ELECTRONICS

ENGINEERING, (PCSIR) KARACHI

Lab. Manual

IE484: Communication Systems

Third Year Spring Semester

Submitted By

Submitted To
Engr. Dr. Farah Haroon
Associate Professor

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 INDEX
Lab.No Title Pg. No

1. Processing of Continuous Time Signals in MATLAB 3

2. Huffman Coding 5

3. Parity Generation and Checking 6

4. Linear Block Codes 7

5. AM Modulation- Time and Frequency Domain Analysis 8

6. DSB Modulation and Demodulation 12

7. Implementation of AM Modulator 13

8. Basic Operation of Spectrum Analyzer 17

9. AM Demodulators- Envelope and Product Detector 21

10. FM Modulation 24

11. Implementation of Sampling Theorem 28

12. White Noise 30

13. Basics of Transmission Lines 31

14. Proposal of Design Project related to Communication Systems 34

15. Implementation of Design Project 35

16. Demonstration and Submission of Design Project 36

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Communication Systems PROCESSING OF CONTINOUS TIME SIGNALS IN MATLAB IIEE
6th Semester Institute of Industrial Electronics Engineering LAB1

OBJECTIVES:
To get acquainted with MATLAB environment for processing continuous time signals.

BRUSH UP:
MATLAB is a software program that allows you to do data manipulation and visualization,
calculations, math and programming. It can be used to do very simple as well as very sophisticated
tasks. It is also recognized as the interactive program for the design and analysis of Communication
tasks.
The modeled signals and systems in Communication are mostly analog (continuous-time) while the
simulation environment is digital (discrete-time). So using Matlab, we have certain constraints.
 A continuous time signal xc(t) is represented by a vector, [xc(t0) xc(t1) ... xc(tN-1)],
whose length is N, i.e. the signal is truncated and sampled.
 Time interval tN-1-t0 and sampling interval Ts=ti+1 - ti depend on the signal.
 Time interval is chosen large enough to obtain enough information. Sampling frequency
typically chosen much higher than the Nyquist frequency, 2*fmax.
 Sampling frequency Fs=1/Ts is chosen so large that the sampled signal “looks like” analog
signal in Matlab. The rule of thumb is 10 times the maximum frequency component.

STEPS TO BE TAKEN:
1. Try out commands (preceded with >>) in your Matlab command window

Creating scalar
>> s=2

Creating matrix
>> a=[1 3;6 9]

Creating vector
>> v=[1 5 9]

Summation
>> a+5

Multiplication
>> b=s*v

Element-by-element multiplication
>> v.*b

Checking length of a vector

>> length(v)

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Checking size of a matrix
>> size(a)

Accessing certain element of a matrix

>> a(1,2)

Accessing certain elements of a vector

>> v(1:2)

Creating time vector from 0 to 0.5s with 0.1 step-size (begin:stepsize:end)

>> t=0:0.1:0.5
OR
>> t=linspace(0,0.5,6)

2. Generate and plot a 50 Hz sinusoidal signal in M file.

3. Justify the criteria for selection of step size of time vector.
4. Conclude your observations and submit the report with M file and proper commenting.

LAB LEARNING OUTCOMES:

After the completion of this Lab, student would be able to
i. Perform basic matrix manipulations using MatLab.
ii. Select proper sampling interval for generation of discrete time signal.
iii. Generate and display discrete time Sine wave in MatLab.

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 Communication Systems Huffman Coding IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 2

OBJECTIVES: Analyze the characteristics and data compression employing Huffman source coding.

BRUSH UP:
Source coding is essential for an efficient transmission of data. It can be employed to reduce the
redundancy of source data, hence is utilized in data compression techniques. Huffman coding is one of
source coding techniques which allocates an unequal efficient code lengths to discrete source symbols of
unequal probabilities.

STEPS TO BE TAKEN:

1. Generate an independent source signal vector of length (1,15) of discrete five symbols.
2. Assign the respective probabilities and create the Huffman code directory.
3. Encode the generated data using Huffman encoder.
4. Calculate and display the respective entropy and average code length.
5. Decode and display the data and compare the results.
6. Submit yours findings and investigations in the form of proper report.

LAB LEARNING OUTCOMES:

After the completion of this Lab, students would be able to

i. Understand the significance of source coding in digital data transmission that how
unequal code length reduces redundancy.
ii. Encode and decode the source data using Huffman code.
iii. Differentiate between entropy and average code length and calculate efficiency.

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Communication Systems Parity Generation and Checking IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 3

OBJECTIVE:
To design and implement 4-bit even and odd parity circuits.

BRUSH UP:
Error can occur as digital codes are transferred from one point to another within the digital systems.
Errors occur as the bits may alter due to noise or component malfunction. The parity method
involves attaching the parity bit to the information bits in order to make total number of 1s either
even or odd.
A parity checker compares a parity bit from a parity generator with a parity bit created from the
original data applied to the generator. A parity generator can be modified to create a parity checker.

STEPS TO BE TAKEN:

1. What is the difference between Even and Odd parity systems.

2. Design and implement a 4-bit even parity generator and checker circuit.
3. Confirm the operation using Truth Table.
4. Repeat the above two steps for Odd parity circuit.
5. When does the parity method fail? Suggest some alternate solution.
6. Explain the checksum method of error detection.

LAB LEARNING OUTCOMES:

After the completion of this Lab, students would be able to
i. Understand the significance and limitations of using parity bit for error checking.
ii. Design and implement Even and Odd parity systems.
iii. Investigate other error checking mechanisms for digital data.

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Communication Systems Linear Block Codes IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 4

OBJECTIVE:
 To apply error-control coding techniques using Linear Block Codes.
 To design and generate Linear Block Codes.
BRUSH UP:
Error-control coding techniques are used to detect and/or correct errors that occur in the
message transmission in a digital communication system. The transmitting side of the error-control
coding adds redundant bits or symbols to the original information signal sequence. The receiving
side of the error-control coding uses these redundant bits or symbols to detect and/or correct the
errors that occurred during transmission. The transmission coding process is known as encoding,
and the receiving coding process is known as decoding.
Block code is an important class of error control coding. In block coding, successive blocks
of K information (message) symbols are formed. The coding algorithm then transforms each block
into a codeword consisting of n symbols where n>k. This structure is called an (n,k) code. The ratio
k/n is called the code rate.
STEPS TO BE TAKEN:
1. Assume that the generator matrix for a (6,3) linear block code is

0 1 1 1 0 0
G  1 0 1 0 1 0 
 
1 1 0 0 0 1
2. Specify the generator matrix as
G = [[0 1 1;1 0 1 ;1 1 0], eye(3)]
[k,n] = size(G)
3. Construct the message vector.
4. Use linear block code to produce the codewords pertaining to each message as
code0 = encode(msg0,N,K,'linear',G)
5. To recover the message vector, the codeword vector needs to be decoded as
dmsg0 = decode(code0,N,K,'linear',G)
6. To compute the number of errors and its error ratio.
[number, ratio] = biterr(msg0,dmsg0)
7. Conclude your results

LAB LEARNING OUTCOMES:

After the completion of this Lab, students would be able to
i. Find the hamming distance between two code words.
ii. Understand the significance of generator and parity check matrix. Encode and decode the
linear block code.
iii. Comprehend the error detection and correction capability of linear block code.

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Communication Systems AM Modulation- Time and Frequency Domain Analysis IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 5

OBJECTIVES:

Understand the phenomena of Amplitude Modulation and the effect of modulation index on the
modulated output.
Simulate the results both in time and frequency domain.

BRUSH UP:

In AM, the information signal varies the amplitude of the carrier in accordance with the amplitude
and frequency variations of the modulating signal.

Modulation index represented as m defines the depth of modulation. It is simply the ratio of the
modulating signal to the carrier voltage.
m = Vm/ Vc
Where, Vm and Vc are the peak values of the modulating and carrier signal.

From the modulated waveform, it can also be calculated as

m= Vmax - Vmin
Vmax + Vmin

Where, Vmax and Vmin are the max and min values of the envelope of the modulated o/p.

The spectrum of modulated output will consists of the carrier and sideband frequencies.

ASSIGNMENTS:

1. Using simulink create the following model of AM.

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2. Set the properties of the modulating signal as

3. Set the parameters of the carrier signal as

4. With simulation time 1000, run the model and display the modulated output on the scope. From
the displayed waveform , measure the modulation index and compare with the set one.

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5. What is the depth of modulation?

6. Now for the case of under modulation with carrier amplitude equal to 1, change the amplitude of
the modulating signal to 0.2 and 0.5 and repeat step4.

7. Now for the case of over modulation with modulating signal amplitude equal to 1, change the
amplitude of the carrier to 0.2 and 0.5 and display the waveforms.

8. Now create the next model again with simulink as

9. Set the amplitude of both the modulating signal and the carrier to 1 and sample time to
1/20000.

10. Set the frequency of the modulating signal to 1KhZ and the frequency of the carrier to
10KhZ.

11. Set the parameters of B-FFT as

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 12. Run and display the spectrum. Measure the carrier and side frequencies and compare with the
expected values.

LAB LEARNING OUTCOMES:

After the completion of this Lab, students would be able to
1. Understand and demonstrate the amplitude modulation technique both in time and frequency
domain.
2. Apply various values of modulation index and analyze its effect on AM modulation.
3. Configure and use Spectrum Analyzer in MatLab

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 Communication Systems DSB Modulation and Demodulation IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 6
OBJECTIVES:
To understand the phenomenon of DSB modulation and demodulation
Analyze the output both in time and frequency domain

BRUSHUP:
The drawback of conventional AM is its power wastage in the carrier signal. The solution to this is
DSBSC or DSB in which the spectrum consists of only two side frequencies with suppressed carrier
frequency.

The message signal can be recovered back from DSBSC using coherent detection. In this technique
the modulated signal is mixed with the carrier having the same frequency and phase as the carrier at
the transmitter. The mixed output is then passed through a LPF which recovers the message back.

ASSIGNMENTS:
1. Using simulink to create the following model with 1K message and 100K carrier.

2. Using high order LPF display and draw the oscillogram of modulated and demodulated
output.
3. Identify the type of demodulation and verify the results using calculations.
4. Using BFFT scope, analyze and draw the spectrum of modulated and demodulated output.
5. Conclude your results.

LAB LEARNING OUTCOMES

After the completion of this Lab, student would be able to
1. Compare and DSB and AM techniques both in time and frequency domain.
2. Demonstrate coherent detection of DSB.
3. Use BFFT scope for frequency analysis.

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 Communication Systems Demonstration of AM Modulator IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 7a

OBJECTIVES:
Demonstrate an Amplitude Modulator using KL 900A communication trainer.

BRUSH UP:

AM modulator circuit using MC 1496 is shown in Figure. Carrier and audio signals are single-ended
inputs and are respectively connected to pin 10 and to pin 1. The gain of entire circuit is determined
by the R8 value. The R9 determines the amount of bias current. Adjusting the amount of VR1 or the
audio amplitude can change the percentage modulation.

Fig: Amplitude modulator using MC1496

The gain of entire circuit is determined by the R8 value. The R9 determines the amount of bias
current. Adjusting the amount of VR1 or the audio amplitude can change the percentage modulation.

Modulation index represented as m defines the depth of modulation. It is simply the ratio of the peak
modulating signal to the peak carrier voltage.

m = Vm/ Vc (1)

With Vmax and Vmin as the max and min values of the envelope of the modulated o/p, m
can also be calculated as

m= Vmax - Vmin (2)

Vmax + Vmin

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ASSIGNMENTS:

1. Locate AM modulator circuit on Module KL-93002. Insert connect plugs in J1 and J3 to set
R8 =1k: and R9 =6.8kΩ.
2. Connect a 0.4Vp-p, 1 kHz sine wave to the audio input, and a 0.4Vp-p, 500 kHz sine wave to
the carrier input.
3. Connect the vertical input of the oscilloscope to the AM output (O/P). Observe the output
waveform. Calculate the percentage modulation of output signal using Eq (2) and compare
with Eq (1).
4. Repeat the steps 2-3 for an audio amplitude of 0.2 Vp-p.
5. Adjust the VR1 and mention its operation.
6. What is the effect of modulation index on the modulated waveform.

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 Communication Systems Implementation of AM Modulator IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 7b

OBJECTIVES:
To generate amplitude modulated wave and determine the percentage modulation.

BRUSH UP:
Amplitude Modulation is defined as a process in which the amplitude of the carrier wave c(t) is
varied linearly with the instantaneous amplitude of the message signal m(t).The standard form of an
amplitude modulated (AM) wave is defined by

Where
Ka is a constant called the amplitude sensitivity of the modulator.

The Modulation Index is defined as, m = Emax - Emin

Emax + Emin

Where Emax and Emin are the maximum and minimum amplitudes of the modulated wave.

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1. Construct the circuit as in Fig 1.Switch on + 12 volts VCC supply.
2. Apply sinusoidal signal of 1 KHz frequency and amplitude 2 Vp-p as modulating signal, and
carrier signal of frequency 11 KHz and amplitude 15 Vp-p.
3. Now slowly increase the amplitude of the modulating signal up to 7V and note down values
of Emax and Emin.
4. Calculate modulation index using equation.
5. Repeat step 5 by varying the amplitude of the modulating signal.

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 Communication Systems Basic Operation of Spectrum Analyzer IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 8

OBJECTIVES:
Understand the basic operation of GSP-730 by using its parameter settings such as frequency,
amplitude and markers.
Measurement and analysis of a baseband signal.

BRUSH UP:
Spectrum analyzers are mainly used to measure physical quantities such as the frequency and
amplitude of a signal. For basic operation, the frequency range must be set first, then the reference
level amplitude.

Fig 1:Snapshot of GSP-730 Spectrum Analyzer

In Fig 1, the horizontal setting is frequency and the vertical axis is amplitude. Therefore a spectrum
analyzer is basically used to perform frequency and amplitude-related measurements.

Range of frequency, span and amplitude can be set using the keyboard shown in Fig 2.

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Fig2: The Frequency, Amplitude and Span keys as well as the keypad and unit keys.

STEPS TO BE TAKEN:

a. Connect the Sinewave baseband signal 0.4 Vpp of frequency 1MHz from the function
generator to the input terminal of the GSP-730 using the RF wire.

b. Set the GSP 730 as follows.

 Center frequency: 2.5MHz
 Start frequency: 0kHz,
 Stop frequency: 5MHz
 Reference level: 10dBm
 RBW: Auto

c. Follow the following steps. When the first step is done, steps 2 and 3 (below) will have
already beenautomatically set. Students may do steps 2 and 3 here mentioned for reference
only.

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 d. Utilize the Marker function on the spectrum analyzer

A function signal generator can also be used as a signal source in the above measurement, but be
aware that the amplitude of the output signal can’t be too high.
dBm is a power unit that is referenced to 1mW. The formula for

X dBm = 10*log(Px/1mW).

Because the output voltage of a signal generator is often used expressed as a voltage into a 50 ohm
load, you must convert voltage to power. A few common values are listed below:

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Note: If voltage is measured without a load on an oscilloscope, the Vpp and Vm values should be
multiplied by 2. For instance, when we get a measured value of 4Vpp into no load, it is the
equivalent of 2Vpp into 50 ohms, or 10dBm after conversion.

e. Draw the spectrum showing the harmonics.

f. What is the spectrum of a theoretical sine wave and why is it different with the actual
measured one?
g. What is the frequency domain feature of the analyzed signal?

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 Communication Systems AM Demodulators IIEE
Envelope and Product Detector
6th Semester Institute of Industrial Electronics Engineering LAB 9

OBJECTIVES:
Understand the principle of amplitude demodulation.
Implement an amplitude demodulator with diode.
Implement an amplitude demodulator with a product detector.

BRUSH UP:
To recover the audio signal in receiver, it is necessary to extract the audio signal from an
AM signal. The process of extracting a modulating signal from a modulated signal is called
demodulation or detection. Detectors can be categorized into two types:

Fig 1: Block Diagram of Diode Detector

Fig 2: Diode Detector Circuit

Demodulation for AM signal can be also accomplished coherently as shown in Fig 3.

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 Fig 3: Block Diagram of Coherent Detection

Fig.4 provides the internal circuit of MC1496 balanced modulator also called as
synchronous detector or product detector.

Fig 4: Product Detector Circuit

STEPS TO BE TAKEN:
Diode Detector Circuit:
1. The AM signal source in this experiment is from the AM modulator output using
carrier of 250mVp-p, 200 kHz sine wave, and the audio signal of 150mVp-p, 3 kHz
sine wave.
2. Adjust the VR1 of AM modulator to get maximum amplitude of AM signal output.
Connect the AM signal output to the input (I/P) of diode detector.
3. Observe the output waveforms of the amplifier and the diode detector, and record the
results

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Product Detector:
4. Repeat step 1 and 2.
5. Connect the output of AM modulator to the input of AM signal (I/P2) of the product
detector located on the bottom of Module KL-93002, and connect the same carrier to
the carrier input (I/P1).
6. Observe the output waveform of the product detector, and record the results.
7. Now connect another carrier with the same amplitude and frequency and observe and
justify the output.

Questions:
8. The recovered output from which detector circuit is better? Give reason.
9. In the product detector circuit of Fig.4, if the carrier signal and the AM signal are
asynchronous, what is the output signal?
10. What is the function of R9, C7 or C9 in Fig.4?
11. What is the function of VR1 or VR2 in Fig.4?
12. What is the function of R5 or R6 in Fig.4?
Conclude your observations and submit the report.

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 Communication Systems FM Modulation IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 10a

OBJECTIVES:
To generate FM using VCO technique.
To generate a FSK signal

BRUSH UP:
FM can be generated by using several techniques. Use of VCO is quite easy. In this experiment, we
will use a famous chip, the LM 566C to generate the FM signal. The LM 566 is a linear voltage-to-
frequency converter which can generate an FM signal up to 1 MHz and for a +/- 10% deviation from
the center frequency, it has an FM distortion of less than 0.2%. The center frequency (fo) is set by a
resistor (Ro) and a capacitor (Co), where:
2.4  V  0.75 V  V  V


fo RoC1o 5 V 


5


2 K  Ro  20 K
Notice that V5 must be between 0.75V+ and V+ for proper operation of the LM 566, and
2 KΩ < Ro < 20 KΩ.
The LM 566 delivers either an FM square wave or an FM triangular wave. However, this is not
important, since the square/triangular wave can be converted into a “good” sine wave with the use of
a simple RC filter. The input impedance for the modulation signal is 1 MΩ and the output
impedance of the square/triangular wave is 50 Ω.
In both cases above, if the modulating signal is a digital waveform, then the resulting FM signal is a
frequency-shift-keying signal (FSK).

STEPS TO BE TAKEN:
1. Connect the LM 566C as shown below. The center frequency is set by Ro, Co, V5/V+. Set V+
(Vcc) to 10 V.

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2. Measure the output voltage Vo (pin #4). It should be a triangular wave with frequency of 100-
110KHz and Vppk around 1.8 V.
3. Connect the signal generator to the modulation input (Vs) and set it to a sine wave with f = 5
KHz and Vppk= 200 mV. Basically, the voltage at pin #5 is now varying by +/- 0.1 V.
The output frequency of the LM 566 will vary by +/- 10 KHz (this is the maximum deviation).
This is the case of mf = 2 (since fm=5 KHz). Notice how the amplitude of the FM signal remains
constant
4. Change from a sinewave to a square wave with Vppk = 400 mV. You will have now only two
frequencies, one around 80 KHz and one around 120 KHz. This is an FSK signal which is
generated by the VCO.
5. Go into the time domain. The scope will lock to the FSK signal using the external trigger.
Measure exactly these frequencies by expanding the timebase, stopping the scope, and using the
Horizontal/delay knob.
6. Note down all the readings and submit the results along with the waveforms.

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 Communication Systems FM Modulation IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 10b

OBJECTIVES:
To generate FM using VCO technique.
To generate a FSK signal

BRUSH UP:
FM can be generated by using several techniques. Use of VCO is quite easy. LM 566C is used to
generate the FM signal. The LM 566 is a linear voltage-to-frequency converter which can generate
an FM signal up to 1 MHz and for a +/- 10% deviation from the center frequency, it has an FM
distortion of less than 0.2%.
If the SW1 is open, this circuit is a typical VCO whose output frequency is determined by the values
of C3 and VR1, and the audio input voltage. If the values of C3 and VR1 are fixed, the output
frequency is directly proportional to the voltage difference between pins 8 and 5, (V8-V5). An
increase in audio input voltage (V5) causes a decrease in the value of (V8-V5) and a decrease in the
output frequency. Conversely, decreasing the audio input voltage (V5) will cause the output
frequency to increase. Values of C3 and VR1 can also determine the output frequency, which is
inversely proportional to the product of VR1 and C3. The center frequency (fo) is
where:

( )
If the SW1 is closed, the voltage divider constructed by R1 and R2 provides a dc level to the audio
input (pin 5). By adjusting the VR1, we can easily tune the VCO center frequency fo. When an audio
signal is applied to the audio input, the output frequency will generate frequency deviations around
fo in the variations of audio amplitude. Thus, a frequency-modulated signal is obtained.

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ASSIGNMENTS:

1. Locate the LM566 Modulator circuit on Module KL-93004. Insert the connect plug
in J2 to set the capacitor to C3 (0.1 µF).

2. Connect a 3.6Vdc to the dc voltage input (pin 5) and adjust the VR1 to obtain an
output frequency of 2 kHz. This frequency is the center frequency fo.

3. Change the dc voltages at pin 5 to 2.7V, 3.0V, 3.3V, 3.9 V, 4.2 V and 4.5V
sequentially. Observe the output frequencies corresponding to the dc voltage inputs
and record the results in Table.

4. Using the results of Table, plot the frequency vs voltage characteristic curve of
LM566.

5. Locate the LM566 FM Modulator circuit on Module KL-93004. Insert connect plugs
in J1 and J3 to set the capacitor to C3 (0.01µF). Turn the VR1 to get the output
frequency of 20 kHz.

6. Connect a 1Vp-p, 1kHz sine wave to the audio i/p. Using the oscilloscope observe
the output waveform (O/P) and record the result.

7. Change the audio frequencies to 3kHz and 5kHz sequentially and observe the output
respective output waveform and record the result.

8. What is the function of R1 and R2 when SW1 is closed?

9. Conclude your results and submit the report.

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 Communication Systems Implementation of Sampling Theorem IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 11

OBJECTIVE:
 To verify the sampling theorem, through an implementation of sample and hold circuit.
 To measure the droop rate of the sample and hold IC with different values of holding
capacitor.

BRUSH UP:
Analog signals, such as music, are often converted to, and stored in, digital form. To do this,
the analog signal is periodically sampled (its instantaneous value is acquired by a circuit), and that
constant value is then held and is converted to a digital word. For example, music signals are often
sampled at 44 kHz, which means there are just a bit more than 20 microseconds available to sample,
hold, and then convert the sampled value to a digital word. Then, the whole process of
sample/hold/convert happens again.
So, any type of analog-to-digital converter must contain, or be preceded by, a circuit which holds the
voltage at the input to the ADC converter constant during the entire conversion time. Conversion
times vary widely, from nanoseconds (for flash ADCs) to microseconds (for successive
approximation ADCs) to hundreds of milliseconds (for dual-slope integrator ADCs).
Droop rate is the rate at which the output voltage drops if the clock input has been removed,
i.e sampling pulses are stopped. Droop rate is inversely related with the value of the holding
capacitor.

STEPS TO BE TAKEN:

I. Sample and Hold Circuit Output Voltage

1. Construct the circuit shown. For the
CHOLD capacitor, quality is very
important. Use a capacitor with a mica,
polystyrene or polypropylene dielectric.
2. Observe and record the voltage on the
clock input (pin 8).
3. Adjust Vin to be a 4 Vpp sinusoid at 10
Hz. Observe it on channel 1, and make it the
trigger source. Observe Vo on channel 2,
changing the frequency slightly, as
needed, so that the display is stable.
Record Vin and Vo.
4. Repeat step 3, with Vin = 100 Hz.
5. Repeat step 3, with Vin = 1,000 Hz.
6. Now change Vin to a 4 Vpp, 50 Hz triangle waveform. Record Vin and Vo.

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.

II. Droop Rate Determination

1. Let Vin = 1 VDC from a DC power supply (be sure to disconnect the function generator).
2. Measure Vo with a DC voltmeter while the 2 kHz differentiated square wave is clocking the
LF398 IC.
3. Disconnect the clock input, and observe Vo on a DC voltmeter for 60 seconds. Record Vo at
the end of 60 seconds.
4. Calculate the droop rate in volts/sec.
5. Replace CHOLD with a 0.1µF capacitor. Repeat II.2, II.3 and II.4.

Lab Report:
1. What is the time constant of the RC network providing the sample commands to the logic
input (pin 8 of the LF398)? Why was this network used (instead of connecting the
squarewave to pin 8 directly)?
2. In parts I.3, I.4 and I.5, was the Nyquist criterion satisfied? Was the output voltage a good
representation of the input voltage for all three input frequencies (10, 100 and 1000 Hz)?
3. Compare the droop rate you measured in part II with the manufacturer’s specified value.

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 Communication Systems White Noise IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 12

OBJECTIVES: Investigation and analysis of simplest communication AWGN channel and analyze
its spectral characteristics.
BRUSH UP:
Besides thermal, many other types of noise sources are Gaussian and have flat spectral density over
a wide range of frequencies. An AWGN channel adds white Gaussian noise to the signal with
normal pdf exhibiting zero mean and finite variance. They are termed as white noise sources with
the totally uncorrelated samples and the power spectrum P(f) almost independent of frequency.
Signal-to-noise ratio (SNR) per sample is also an important parameter for an AWGN channel.

STEPS TO BE TAKEN:

1. Analyze the simplest communication channel with an additive Gaussian noise (AWGN) in
Matlab.
2. Generate a triangular signal and add random AWGN in m file.
3. Vary SNR in dB and plot the channel input and output signals in each case.
4. Generate and examine the normal pdf with varying mean and variance.
5. Given a sine wave signal (amplitude and frequency of your choice and 100 samples on x-
axis).
6. Apply auto-correlation of the sine wave signal and plot the output.
7. Generate noise signal of equal number of samples (100). Plot the noise signal.
8. Apply auto-correlation of noisy signal and plot the output. Write your observation regarding
the output.
9. Now, add the sine wave signal (generated previously) to the noisy signal and plot the graph
showing the corrupt signal.
10. Now, apply auto-correlation of the sine + noise signal and plot the output. What do you
observe? Write your conclusion.
Submit yours findings and investigations in the form of proper report.

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 Communication Systems Basics of Transmission Lines IIEE
6th Semester Institute of Industrial Electronics Engineering LAB 13

OBJECTIVES:
 Measure the characteristics impedance of a transmission line.
 Measure the attenuation of a line

BRUSH UP:

Transmission lines are a means of conveying signals or power from one point to another.
Experimental Training Board is available to study the various parameters in Transmission Line. It
consists of the Coaxial Cables of 100 meters in the coiled form of 25m each. The lines are straight
and it is done for space saving.
Equivalent circuit of any transmission line is

Characteristic impedance of transmission line is its input impedance if it was infinitely long given as

And is shown as

The basic properties used in the trainer are

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The attenuation of RG 174 cable is 40dB/100 m at 200 MHz. But due to coiled form it reduces to
3 dB/100 m.

ASSIGNMENTS:
1. Below Fig shows the modalities for the measurements to be formed.

2. Make a connection as shown in

3. Both the inductance and the ohmic resistance of the line are measured in series by short
circuiting end of the line and connecting the measuring instruments to the start of the line.
The capacitance and the conductance are measured in parallel by operating on the line.

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4. The resistance R and the conductance G can be measured with an ohmmeter is required
which is able to perform resistance measurements with a range greater than 100M.
5. At high frequencies characteristic impedance is approximated as

6. The ohmic resistance R and conductance G are responsible for energy dissipation in the form
of heat. These loses contribute to attenuation a is given by

7. To measure attenuation, adjust Ri and RL for 18 and 68 respectively with the help of
DMM.

8. Set the sine wave frequency to approx.. 100 KHz and level to 0.4Vpp.
9. Make connections as in above fig.
10. Measure signal level at input, 25, 50, 75 and 100m length and calculate attenuation in dB at
various lengths. The attenuation is approximately -2dB at 100m.
11. Conclude your results and submit the report.

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 Communication Systems Proposal of Design Project I.I.E.E
6th Semester Institute of Industrial Electronics Engineering Lab 14

OBJECTIVES:
To submit the proposal of Design Project related to Communication Systems.

ASSIGNMENTS:
Every group is required to submit the proposal and get it approved.

The proposal should include the following

1. Introduction
2. Literature Review
3. List of Objectives
4. Methodology
5. Conclusion
6. Timeline

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 Communication Systems Implementation of Design Project I.I.E.E
6th Semester Institute of Industrial Electronics Engineering Lab 15

OBJECTIVE:
Implementation of the proposed Design Project

ASSIGNMENTS:
1. Implement the circuit of proposed Designi Project.
2. Verify the results with the simulations.

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 Communication Systems Demonstration and Submission of Design Project I.I.E.E
6th Semester Institute of Industrial Electronics Engineering Lab 16

OBJECTIVE:
Demonstration and submission of the implemented Design Project

ASSIGNMENTS:
1. Demonstrate the working of implemented Mini Project.
2. The project should be submitted in proper casing with terminals and casing. It should be
usable as a
trainer in future.
3. Submit the report comprising of Objectives, Theory, Simulation and theoretical results,
Calculations, Waveforms, Specifications, Manual-How to use it and Conclusion.

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