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CS Lab Manual Final

The document outlines the syllabus for the Communication Systems Laboratory course (EC 3461) offered to second-year Electronics and Communication Engineering students at Trichy Engineering College. It includes course objectives, a list of experiments, course outcomes, and the vision and mission of the department. The document also details program educational objectives, program specific objectives, and various assessment criteria for evaluating student performance in the lab.

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0% found this document useful (0 votes)
31 views96 pages

CS Lab Manual Final

The document outlines the syllabus for the Communication Systems Laboratory course (EC 3461) offered to second-year Electronics and Communication Engineering students at Trichy Engineering College. It includes course objectives, a list of experiments, course outcomes, and the vision and mission of the department. The document also details program educational objectives, program specific objectives, and various assessment criteria for evaluating student performance in the lab.

Uploaded by

rajiece.19
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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TRICHY ENGINEERING COLLEGE

( A Unit of SS Group of Institutions)


(Approved by AICTE & Affiliated to Anna University, Chennai)
Sivagnanam Nagar, Konalai, Tiruchirappalli - 621 132.

DEPARTMENT OF ELECTRONICS AND COMMUNICATION


ENGINEERING

Regulation-2021

CLASS : II YEAR-ECE

SEMESTER : IV SEM

SUBJECT CODE: EC 3461

SUBJECT : COMMUNICATION SYSTEMS LABORATORY


SYLLABUS

COURSE OBJECTIVES:
● To study the AM & FM Modulation and Demodulation.
● To learn and realize the effects of sampling and TDM.
● To understand the PCM & Digital Modulation.
● To Simulate Digital Modulation Schemes.
● To Implement Equalization Algorithms and Error Control Coding Schemes.

LIST OF EXPERIMENTS
1. AM- Modulator and Demodulator
2. FM - Modulator and Demodulator
3. Pre-Emphasis and De-Emphasis.
4. Signal sampling and TDM.
5. Pulse Code Modulation and Demodulation.
6. Pulse Amplitude Modulation and Demodulation.
7. Pulse Position Modulation and Demodulation and Pulse Width Modulation and Demodulation.
8. Digital Modulation – ASK, PSK, FSK.
9. Delta Modulation and Demodulation.
10. Simulation of ASK,FSK, and BPSK Generation and Detection Schemes.
11. Simulation of DPSK, QPSK and QAM Generation and Detection Schemes.
12. Simulation of Linear Block and Cyclic Error Control coding Schemes.
TOTAL: 45 PERIODS

COURSE OUTCOMES:

At the end of the laboratory course, the student will be able to understand the:

1. Design AM,FM & Digital Modulators for specific applications.


2. Compute the sampling frequency for digital modulation.
3. Simulate & validate the various functional modules of Communication system.
4. Demonstrate their knowledge in base band signaling schemes through implementation of
digital modulation schemes.
5. Apply various channel coding schemes & demonstrate their capabilities towards the
improvement of the noise performance of Communications
INDEX
EXP. DATE NAME OF THE EXPERIMENT MARKS SIGNATURE
NO
Vision of the Department

To become a centre of excellence in training, research, consultancy and innovation in the field of
Electronics and Communication Engineering.

Mission of the Department

1. To impart value- and soft skills-oriented curriculum to meet national and global challenges.

2. To foster critical thinking and creativity through research and experimentation.

3. To prepare our students based technical education with a strong foundation on analytical

thinking, design, application to be a lifetime professional with creativity and leadership.

Program Educational Objective (PEOs)

PEO 1 – To provide the students with a strong foundation in the required sciences in order to pursue
studies in Electronics and Communication Engineering

PEO 2 -To gain adequate knowledge to become good professional in electronic and communication
engineering associated industries, higher education and research

Program Specific Objective (PSOs)

PSO 1 – To Design, develop and analyze electronic systems through application of relevant electronics,
mathematics and engineering principles
PSO 2 - To Design, develop and analyze communication systems through application of fundamentals from
communication principles, signal processing, and RF System Design & Electromagnetics
PSO 3 – To Adapt to emerging electronics and communication technologies and develop innovative
solutions for existing and newer problems
PO1 - Engineering knowledge: Apply the knowledge of mathematics, science, engineering
fundamentals, and an engineering specialization to the solution of complex engineering problems.
PO2 - Problem analysis: Identify, formulate, review research literature, and analyze complex
Program Outcomes
engineering problems reaching substantiated conclusions using first principles of mathematics, natural
sciences, and engineering sciences.
PO3 - Design/development of solutions: Design solutions for complex engineering problems and
design system components or processes that meet the specified needs with appropriate consideration
for the public health and safety, and the cultural, societal, and environmental considerations.
PO4 - Conduct investigations of complex problems: Use research-based knowledge and research
methods including design of experiments, analysis and interpretation of data, and synthesis of the
information to provide valid conclusions.
PO5 - Modern tool usage: Create, select, and apply appropriate techniques, resources, and modern
engineering and IT tools including prediction and modeling to complex engineering activities with an
understanding of the limitations.
PO6 - The engineer and society: Apply reasoning informed by the contextual knowledge to assess
societal, health, safety, legal and cultural issues and the consequent responsibilities relevant to the
professional engineering practice.
PO7 - Environment and sustainability: Understand the impact of the professional engineering
solutions in societal and environmental contexts, and demonstrate the knowledge of, and need for
sustainable development.
PO8 - Ethics: Apply ethical principles and commit to professional ethics and responsibilities and
norms of the engineering practice.
PO9 - Individual and team work: Function effectively as an individual, and as a member orleader in
diverse teams, and in multidisciplinary settings.
PO10 - Communication: Communicate effectively on complex engineering activities with the
engineering community and with society at large, such as, being able to comprehend and write
effective reports and design documentation, make effective presentations, and give and receive clear
instructions.
PO11 - Project management and finance: Demonstrate knowledge and understanding of the
engineering and management principles and apply these to one’s own work, as a member and leader in
a team, to manage projects and in multidisciplinary environments.
PO12 - Life-long learning: Recognize the need for, and have the preparation and ability to engage in
independent and life-long learning in the broadest context of technological change.
EC3461 COMMUNICATION SYSTEMS LABORATORY

LTPC
COURSE OBJECTIVES: 0 0 3 1.5
● To study the AM & FM Modulation and Demodulation.
● To learn and realize the effects of sampling and TDM.
● To understand the PCM & Digital Modulation.
● To Simulate Digital Modulation Schemes.
● To Implement Equalization Algorithms and Error Control Coding
Schemes.

LIST OF EXPERIMENTS
1. AM- Modulator and Demodulator
2. FM - Modulator and Demodulator
3. Pre-Emphasis and De-Emphasis.
4. Signal sampling and TDM.
5. Pulse Code Modulation and Demodulation.
6. Pulse Amplitude Modulation and Demodulation.
7. Pulse Position Modulation and Demodulation and Pulse Width Modulation and Demodulation.
8. Digital Modulation – ASK, PSK, FSK.
9. Delta Modulation and Demodulation.
10. Simulation of ASK, FSK, and BPSK Generation and Detection Schemes.
11. Simulation of DPSK, QPSK and QAM Generation and Detection Schemes.
12. Simulation of Linear Block and Cyclic Error Control coding Schemes.
TOTAL: 45 PERIODS

COURSE OUTCOMES:

At the end of the laboratory course, the student will be able to understand the:

1. Design AM, FM & Digital Modulators for specific applications.


2. Compute the sampling frequency for digital modulation.
3. Simulate & validate the various functional modules of Communication system.
4. Demonstrate their knowledge in base band signaling schemes through implementation of
digital modulation schemes.
5. Apply various channel coding schemes & demonstrate their capabilities towards the
improvement of the noise performance of Communications
EC3461 COMMUNICATION SYSTEMS LABORATORY

NBA CODE FOR THE SUBJECT: EC


SEMESTER: IV (AY: 22-23 EVEN)
CO-Code Course Outcome Description
CO 1
Design AM,FM & Digital Modulators for specific applications

CO 2 Compute the sampling frequency for digital modulation.


CO 3
Simulate & validate the various functional modules of Communication system

CO 4 Demonstrate their knowledge in base band signaling schemes through


implementation of digital modulation schemes.
CO 5 Apply various channel coding schemes & demonstrate their capabilities
towards the improvement of the noise performance of Communications

Subject Code : EC3461 NBA Code: EC408 Title: EC3461 COMMUNICATION SYSTEMS
LAB Semester : 4

CO PO PO PO PO PO PO PO PO PO PO PO PO PS PS PS
Code 1 2 3 4 5 6 7 8 9 10 11 12 O1 O2 O3

EC408.1 3 3 3 3 2 2 3 2 2

EC408.2 3 3 3 2 2 2 3 2 2

EC408.3 3 3 3 2 3 1 2 2 1

EC408.4 3 3 3 2 2 1 2 2 1
DEPARTMENT OF ELECTRONICS AND COMMUNICATION
ENGINEERING
Name of Lab Course EC3461 COMMUNICATION SYSTEMS LAB
Semester & Year IV\II
Name of the Student
Registration No
Name of the Evaluator
Marks scored out of 10
RUBRIC ASSESSMENT FOR LAB COURSE
Performance
Level 1 (0-1) Level 2 (1-2) Level 3 (2-3)
Indicators
Explanation to, the Pre Explanation to, the Pre lab Explanation to, the Pre lab
lab questions and questions and objective of questions and objective of
objective of the the experiment, is, where the experiment, is, where
Pre Lab experiment, is, where compared to the compared to the
Questions, compared to the expectation of the faculty expectation of the faculty
Objectives (P –I) expectation of the is partially satisfactory. is highly satisfactory.
faculty is not
satisfactory.
Explanation to the Explanation to the Explanation to the
procedure of the procedure of the procedure of the
experiment, is, where experiment, is, where experiment, is, where
Procedures (P-II) compared to the compared to the compared to the
expectation of the expectation of the faculty expectation of the faculty
faculty is not is partially satisfactory. is highly satisfactory.
satisfactory.
Calculation of the Calculation of the Calculation of the
Data / observed values and observed values and observed values and
Observations (P- validation of the results validation of the results of validation of the results of
III) of the experiment the experiment the experiment precise.
inaccurate. approximate.
Explanation to the post Explanation to the post lab Explanation to the post lab
lab questions and questions and conclusions questions and conclusions
Post Lab conclusions of the of the experiments is, of the experiments, is,
Questions, experiments, is, where where compared to the where compared to the
Conclusions (P- compared to the expectation of the faculty expectation of the faculty
IV) expectation of the partially satisfactory. is highly satisfactory.
faculty is not
satisfactory.
Register NO:
Assessment Sheet

P-I P-II P-III P-IV Total


S.No Date Experiment
(2) (2) (3) (3) (10)

10

11

12

TOTAL (OUT OF 10)


AM TRANSMITTER (MODULATOR)
1. AM MODULATOR AND DEMODULATOR

AIM:
To transmit a modulating signal after amplitude modulation using AM transmitter and
receive the signal back after demodulating using AM receiver.
APPARATUS REQUIRED:
S.No. Name of the Equipment/ Component Range Quantity
1. AM Transmitter kit - ACL 01 - 1
2. AM Receiver kit - ACL 01 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required
THEORY:
MODULATION THEORY:
Modulation is defined as the process by which some characteristics of a carrier signal is
varied in accordance with a modulating signal. The base band signal is referred to as the
modulating signal and the output of the modulation process is called as the modulation signal.
The carrier frequency fc must be much greater than the highest frequency components fm of the
message signal m (t) i.e. fc>>fm. The modulation index must be less than unity. If the
modulation index is greater than unity, the carrier wave becomes over modulated.
The modulating, carrier and modulated signals are given by
Vm(t) =Vmsinωmt ; VC(t) = VCsinωCt ; VAM(t) = VC (1+ma sinωmt) sinωCt
The modulation index is given by, ma = Vm / VC.
Vm = Vmax – Vmin and VC = Vmax + Vmin
The amplitude of the modulated signal is given by,
Where Vm = maximum amplitude of modulating signal, VC = maximum amplitude of
carrier signal, Vmax = maximum variation of AM signal, Vmin = minimum variation of AM
signal
DEMODULATION THEORY:
Demodulation is the reverse process of modulation. The detector circuit is employed to
separate the carrier wave and eliminate the side bands. Since the envelope of an AM wave has
the same shape as the message, independent of the carrier frequency and phase, demodulation
can be accomplished by extracting envelope. The depth of modulation at the detector output
greater than unity and circuit impedance is less than circuit load (Rl>Zm) results in clipping
of negative peaks of modulating signal. It is called “negative clipping “.
TABULATION:

Amplitude Time Period Frequency


Parameter
(V) in seconds in Hz

Message signal

Carrier signal

Modulated signal

Demodulated signal

Calculation of modulation index:


Practical calculation

Theoretical calculation
MODEL GRAPH:

PROCEDURE:
1. The circuit wiring is done as shown in diagram
2. A modulating signal input given to the Amplitude modulator
3. Now increase the amplitude of the modulating signal to the required level.
4. The amplitude and the time duration of the modulating signal are observed using CRO.
5. Finally the amplitude modulated output is observed from the output of amplitude
modulator stage and the amplitude and time duration of the AM wave are noted down.
6. Calculate the modulation index by using the formula and verify them.
7. The final demodulated signal is viewed using CRO at the output of audio power
amplifier stage. Also the amplitude and time duration of the demodulated wave are noted
down.

VIVA QUESTIONS:

1. What is the need of modulation and demodulation?


2. What is the range of frequency in commercial AM broadcasting?
3. What is analog modulation various techniques?
4. What is the difference between detector and demodulator?
5. Define the term Modulation index.
6. What are the main components of a RF receiver?
7.What is the Difference between Coherent and Non-coherent Demodulation?
8. Why the Intermediate Frequency Should is Carefully Chosen As?
9. What is the function of an Automatic Gain Control of the AM
Receiver? 10.Define Amplitude Modulation?

RESULT:
Thus the AM signal was transmitted using AM trainer kit and the AM signal
detected using AM detector kit. The calculated modulation index ma = .
FM MODULATOR

FM DEMODULATOR
2. FM MODULATION AND DEMODULATION
AIM:
To plot the modulation characteristics of FM modulator and demodulator and also to

Observe and measure frequency deviation and modulation index of FM.

APPARATUS REQUIRED:
S.No. Name of the Equipment/ Component Range Quantity
1. FM Transmitter kit - ACL 03 - 1
2. FM Receiver kit - ACL 04 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required

THEORY:
Frequency modulation is a type of modulation in which the frequency of the
high frequency (carrier) is varied in accordance with the instantaneous value of
the modulating signal.
FREQUENCY MODULATION GENERATION:
The circuits used to generate a frequency modulation must vary the frequency of
a high frequency signal (carrier) as function of the amplitude of a low frequency signal
(modulating signal). In practice there are two main methods used to generate FM.
DIRECT METHOD
An oscilloscope is used in which the reactance of one of the elements of the
resonant circuit depends on the modulating voltage. The most common device with
variable reactance is the Varactor or Varicap, which is a particular diode which capacity
varies as function of the reverse bias voltage. The frequency of the carrier is established
with AFC circuits (Automated frequency control) or PLL (Phase locked loop).
INDIRECT METHOD:
The FM is obtained in this case by a phase modulation, after the modulating
signal has been integrated. In this phase modulator the carrier can be generated by a
quartz oscillator, and so its frequency stabilization is easier. In the circuit used for the
exercise, the frequency modulation is generated by a Hartley oscillator, which
frequency is determined by a fixed inductance and by capacity (variable) supplied by
varicap diodes.
FREQUENCY DEVIATION f and MODULATION INDEX fm:
The frequency deviation f represents the maximum shift between the modulated
signal frequency, over and under the frequency of the carrier.
TABULATION

Time Period
Parameter Amplitude (V) Frequency in Hz
in seconds

Message signal

Carrier signal
Tmin = Fmin =
Modulated signal Tmax = Fmax =
Demodulated signal

MODEL GRAPH
We define modulation index m f the ratio between f and the modulating frequency f.

PROCEDURE:
1. Connect the power supply with proper polarity to the kit. While connecting this
ensures that the Power supply is OFF.
2. Switch on the power supply and carry out the following presetting as shownin circuit
Diagram.
3. In the FM modulator set the level about 2Vpp and frequency knob to the minimumand
switch on 1500 KHz.
4. Observe the FM modulated waveform from the RF/FM output of the FM modulator
measure frequency deviation and modulation index of FM.
5. For demodulation switch on the demodulator and carry out thefollowing demodulation
connection as shown in circuit diagram.
6. Observe the demodulated waveform and plot the graph.
VIVA QUESTION

1. Why frequency modulation is better than amplitude modulation?


2. What is the application of FM?
3. What is PLL?
4. What is FM modulation?
5. What is FM demodulation?
6. Define frequency deviation in FM?
7. Difference between narrowband and wideband FM?
8. What are the advantages of FM over AM?
9. What are the disadvantages of FM over AM?
10. Mention the types of digital modulation.

RESULT:
Thus the modulation characteristics of FM modulator and demodulator are observed and
plotted.
PRE- EMPHASIS CIRCUIT

DE-EMPHASIS CIRCUIT
3. PRE-EMPHASIS AND DE-EMPHASIS.

AIM:
To design a pre emphasis circuit to boost the input signal level for a FM transmitter for a cut off
frequency of 1KHz. Attenuate the boosted high frequency signals at the receiver side using a
deemphasis circuit with a cutoff frequency of 1.6KHz. Analyze the frequency response
characteristics of pre emphasis and de emphasis circuits.

APPARATUS REQUIRED:

S.No. Name of the Equipment/ Component Range Quantity


1. FM Transmitter kit - ACL 03 - 1
2. FM Receiver kit - ACL 04 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required

THEORY:
Pre‐emphasis refers to boosting the relative amplitudes of the modulating voltage for higher audio
frequencies. Pre‐emphasis is done at the transmitting side of the frequency modulator. Signals
with higher modulation frequencies have lower SNR. In order to compensate this, the high
frequency signals are emphasised or boosted in amplitude at the transmitter section of a
communication system prior to the modulation process. That is, the pre ‐ emphasis network allows
the high frequency modulating signal to modulate the carrier at higher level, this causes more
frequency deviation.
The cut off frequency is given by the formula
fc = R/(2π L)
DE‐EMPHASIS:
De‐emphasis is the complement of pre‐emphasis, in the antinoise system called emphasis. This
circuit is used to attenuate the high frequency signal that is boosted at the transmitter section. The
circuit is placed at the receiving side. It acts as a low pass filter.
The cut off frequency is given by the formula
fc = 1/(2πRC)

VIVA QUESTIONS

1. What is meant by threshold effect?


2. What is pre‐emphasis?
3. How the threshold effect can be avoided?
4. What is fidelity?
5. What is sensitivity and selectivity?
TABULATION

Gain in
Time
Parameter Amplitude Frequency in db=20log(Vo/Vi)
Period in
(V) Hz
seconds

Message signal

Carrier signal
Tmin = Fmin =
Modulated signal Tmax = Fmax =
Demodulated
signal
Pre-emphasis output

De-emphasis output

MODEL GRAPH
RESULT:

Thus the frequency response characteristics of pre emphasis and de emphasis circuits were
generated and verified.
CONNECTION DIAGRAM

NATURAL SAMPLING

SAMPLE AND HOLD

FLAT – TOP SAMPLING


SIGNAL SAMPLING AND RECONSTRUCTION

AIM:
To sample and reconstruct the given signal using natural sampling, sample - hold and flat
top sampling techniques using DCL-01Kit.
APPARATUS REQUIRED:

S.No. Name of the Equipment/ Component Range Quantity


1. DCL-01 trainer kit - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required
THEORY
Sampling is the process of splitting the given analog signal into different samples of
equal amplitudes with respect to time. There are two types of sampling namely natural
sampling, flat top sampling. Sampling should follow strictly the Nyquist Criterion i.e. the
sampling frequency should be twice higher than that of the highest frequency signal.
fs ≥ 2fm
Where,
fs= Minimum Nyquist Sampling rate (Hz), fm= Maximum analog input frequency (Hz).
In natural sampling, the top of the sampled pulse follows the shape of the original signal.
Since the natural sampling increases the system complexity, the flat top sampling is mostly
preferred in practical case. The flat top sampling is achieved with the help of sample and hold
circuit. Sample and Hold circuits are used internally in analog to digital conversion. Here, the
sampled signal obtained at each sampling instant is hold until the next sampling instant. The
sampling process must follow the sampling theorem for proper signal reconstruction. In other
words, the sampling frequency must be equal to twice that of the highest frequency component
present in the original signal.
The reconstructed signal is the succession of sine pulses weighted by x (nTs) these
pulses are interpolated with the help of a LPF. It is also called reconstruction filter or
interpolation filter Natural sampling is chopper sampling because the waveform of the sampled
signal appears to be chopped off from the original signal waveform. The top of the samples
remains constant and equal to instantaneous value of x(t) at start of sampling fs = 1/Ts.
TABULATION

Amplitude Time Period


Parameters
in Volts in Seconds
Input Signal
Ton =
Sampling Signal
Toff=
Ton =
Natural sampling Output
Toff=
Ton =
Flat-top sampling Output
Toff=
Sample and Hold output

Reconstructed Signal

MODEL GRAPH

D = Ton / (Ton + Toff) = %


PROCEDURE:

1. Give the connections as per the connection diagram.


2. Put the duty cycle selector switch in position 50%.
3. Connect the modulating signal of 1 KHz frequency to BUFIN and measure its amplitude
and time period.
4. Connect the sampling frequency clock in the internal mode INT CLK using
the sampling signal selector switch SW4.
5. Set the sampling frequency to 8 KHz and note down the amplitude and time period of
the sampling signal.
6. Observe the natural sampling output at OUT terminal and note down the amplitude
and time period of the sampled signal.
7. Give the sampled signal to the 2nd order low pass filter circuit and observe
the reconstructed signal.
8. Note down the amplitude and time period of the reconstructed signal.
9. Repeat the same procedure for flat top sampling, Sample and hold and note down
the readings.
10. Plot the readings in the graph.
VIVA QUESTIONS:

1. What is the need of sampling?


2. What is sampling rate?
3. What is Nyquist rate of sampling?
4. Mention the types of sampling.
5. State the sampling theorem.
6. What is natural sampling?
7. Differentiate three types of sampling techniques.
8. What is the function sample & hold circuit?
9. What is aliasing effect and how it can be eliminated?
10. State the role of duty cycle in sampling circuit.
11. What is the function reconstruction filter?
12. What is the use of sampling and reconstruction technique in
communication engineering?

RESULT:
Thus the given signal is sampled and reconstructed for natural sampling, sample and hold
and flat top sampling also plotted the observed waveforms.
CONNECTION DIAGRAM

MODEL GRAPH
TIME DIVISION MULTIPLEXING

AIM:
To perform four channel Time Division multiplexing and De multiplexing using DCL –02
trainer kit.
APPARATUS REQUIRED:

S.No. Name of the Equipment/ Component Range Quantity


1. DCL-02 trainer kit - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required

THEORY:
Time Division Multiplexing (TDM) is a technique of transmitting different source signals
on the same channel at different time slots. That is several information can be transmitted over a
single channel by sending samples from different information sources at different moments.
TDM is widely used in digital communication systems to increase the efficiency of the
transmitting medium. TDM can be achieved by electronically switching the samples such that
they interleave sequentially at correct instant in time without mutual interference. A
major problem in any TDM system is the synchronization of the transmitter and receiver timing
circuits. The transmitter and receiver must switch at the same time and frequency.TDM based
on analog modulation, the time slots are separated by guard slots to prevent crosstalk between
the channels.
In PAM, PPM the pulse is present for a short duration and for most of the time between the
two pulses no signal is present. This free space between the pulses can be occupied by pulses
from other channels. Thus, time division multiplexing makes maximum utilization of the
transmission channel. Each channel to be transmitted is passed through the low pass filter. The
outputs of the low pass filters are connected to the rotating sampling switch (or) commutator.
It takes the sample from each channel per revolution and rotates at the rate of f s. Thus the
sampling frequency becomes fs the single signal composed due to multiplexing of input
channels. These channels signals are then passed through low pass reconstruction filters. If the
highest signal frequency present in all the channels is fm, then by sampling theorem, the
sampling frequency fs must be such that fs ≥2fm. Therefore, the time space between successive
samples from any one input will be Ts=1/fs, and Ts≤ 1/2fm.
TABULATION:

Amplitude Time Period


Parameters
in Volts in Seconds
Channel 0 (CH 0)

Channel 1 (CH 1)

Channel 2 (CH 2)

Channel 3 (CH 3)

TxD

RxD

Output 0

Output 1

Output 2

Output 3
PROCEDURE:

1. Give the connections as per the connection diagram.


2. Connect the power supply with proper polarity to the kit DCL-02 and switch it ON.
3. Connect 250 Hz, 500 Hz, 1 KHz, 2 KHz sine wave signal from the function generator to
the multiplexer input channel CH0, CH1, CH2 and CH3 by means of connecting cords.
4. Connect the multiplexer OUT TXD to the transmitter section to the de-multiplexer
input RXD of the receiver section.
5. Connect the output of the receiver section CH0, CH1, CH2 and CH3 to the IN0, IN1,
IN2 and IN3 of the filter section.
6. Connect the sampling clock TXCLK and the channel identification clock SYNC of the
receiver section respectively.
7. Set the amplitude of the input sine wave as desired.
8. Take the observations and draw the required graphs.
VIVA QUESTIONS:

1. What is meant by multiplexing technique?


2. What are the two types of multiplexing?
3. When do you prefer TDM to FDM?
4. When would you prefer FDM to TDM?
5. Differentiate TDM and FDM.
6. Define crosstalk effect in PAM/TDM system.
7. What is the transmission bandwidth of a PAM/TDM channel?
8. State the advantages of TDM.
9. State the disadvantages of TDM.
10. Why are the bands limiting filters used at the PAM-TDM transmitter?
11. In TDM, multiple signals share a channel by transmitting in different ……..
12. The minimum sampling rate for a voice channel is………..

RESULT:

Thus the four channel Time division multiplexing and de-multiplexing using DCL-02 kit
was performed and obtained its waveform.
CONNECTION DIAGRAM:

MODEL GRAPH:
5. PULSE CODE MODULATION AND DEMODULATION

AIM: To generate a PCM signal using PCM modulator and detect the message signal from
PCM signal by using PCM demodulator.

APPARATUS REQUIRED:

S.No. Name of the Equipment/ Range Quantity


Component
1 PCM Trainer kit - DCL 03, DCL - 1
. 04
2 CRO 30 MHz 1
.
3 Power supply 5V, 1
. ±12V
4 Patch chords - Required

THEORY:
Pulse code modulation is known as digital pulse modulation technique. It is the process in
which the message signal is sampled and the amplitude of each sample is rounded off to the nearest
one of the finite set of allowable values. It consists of three main parts transmitter, transmitter path
and receiver. The essential operation in the transmitter of a PCM system are sampling, quantizing
and encoding. The band pass filter limits the frequency of the analog input signal. The sample and
hold circuit periodically samples the analog input signal and converts those to a multi-level PAM
signal. The ADC converts PAM samples to parallel PCM codes which are converted to serial binary
data in parallel to serial converter and then outputted on the transmission line as serial digital pulse.
The transmission line repeaters are placed at prescribed distance to regenerate the digital pulse.
In the receiver serial to parallel converter converts serial pulse received from the transmission
line to parallel PCM codes. The DAC converts the parallel PCM codes to multi- level PAM signals.
The hold circuit is basically a Low Pass Filter that converts the PAM signal back to its original
analog form.
ADVANTAGES:
1. Secrecy
2. Noise resistant and hence free from channel interference
DISADVANTAGE:
1. Requires more bandwidth
APPLICATION:
1. Compact DISC for storage

2. Military Applications.
PROCEDURE:
1. Refer the block diagram and carry out the following connections and switch settings.
2. Connect the power supply with power polarity to the kit and switch it ‘ON’.
3. Put the switch sw1 to ‘FAST’ mode.
4. Select 500Hz and 1 KHz sine wave signals generated on board.
5. Connect the signals to ‘CH0’ and ‘CH1’ of sample and hold circuit.
6. The output of sample and hold circuit ‘OUT0’ and ‘OUT1’ are given to the multiplexer, then
to the pulse code modulation logic.
7. The output ‘TXDATA’ is connected to ‘RXDATA’ of pulse code demodulation logic.
8. The output ‘DAC OUT’ is given to ‘IN’ of ‘DEMUX’.
9. Then the output of ‘DEMUX’ –‘OUT0’ and ‘OUT1’ are given to ‘IN0’ and ‘IN1’ of filter.
10. Observe the pulse code demodulated output at ‘OUT0’ and ‘OUT1’.

Time period in Sec


Parameters Amplitude in V TON TOFF
Input Signal 1(500Hz)
Input Signal 1(1KHz)
Sample & Hold OUT 0
Sample & Hold OUT 1
Multiplexer CLK1
Multiplexer CLK1
Multiplexed Data
MUX OUT
TX Data
TX CLK
TX SYNC
RX Data
RX CLK
RX SYNC
DAC OUT
Demultiplexer CLK1
Demultiplexer CLK2
Demultiplexed Data CH 0
Demultiplexed Data CH 1
Received Signal OUT 0
Received Signal OUT 1
VIVA QUESTIONS:
1. What is PCM?
2. How bits are needed to encode N different levels?
3. Define step size?
4. How to calculate Step size in PCM?
5. Define Quantization error.
6. What is the max value of Quantization error?
7. What are the applications of PCM?
8. What are the disadvantages of Pulse code modulation?
9. Define Differential pulse code modulation?
10. Why DPCM is better than PCM?

RESULT
Thus the PCM signal was generated using PCM modulator and the message signal was
detected from PCM signal by using PCM demodulator.
CONNECTION DIAGRAM
6. PULSE AMPLITUDE MODULATION AND DEMODULATION

AIM:
To generate a PAM signal using PAM modulator and detect the message signal from
PAM signal by using PAM demodulator.

APPARATUS REQUIRED:

S.No. Name of the Equipment/ Range Quantity


Component
1 PAM Trainer kit - DCL 08 - 1
.
2 CRO 30 MHz 1
.
3 Power supply 5V, 1
. ±12V
4 Patch chords - Required

THEORY:

In Pulse Amplitude Modulation, the signal is sampled at regular intervals and the amplitude of
each sample is made proportional to the amplitude of the signal at that instant of sampling. This
amplitude of each sample is hold for the sample duration to make pulses flat top. The pulse
amplitude demodulator consists of active Low pass butter worth filter. It filters out the sampling
frequency and their harmonics from the modulated signal and recovers the base band by integrated
action.

TABULATION

Time period in Sec


Parameters Amplitude in V
TON TOFF
Input Signal 1(1 KHz)

Clock Signal (16 KHz)

PAM Output

Amplifier Output

Filter Output
MODEL GRAPH

VIVA QUESTIONS:
1. What is PAM?
2. What are the applications of PAM?
3. What are the advantages of PAM or pulse amplitude modulation?
4. What are the disadvantages of pulse amplitude modulation?
5. What factors affect pulse amplitude modulation?
6. State and explain the types of pulse amplitude modulation.
7. What are the advantages of PAM or pulse amplitude modulation?

RESULT
Thus the PAM signal was generated using PAM modulator and the message signal was detected
from PAM signal by using PAM demodulator.
CONNECTION DIAGRAM
PULSE WIDTH MODULATION AND DEMODULATION

AIM:
To generate a PWM signal using PWM modulator and detect the message signal from
PWM signal by using PWM demodulator.

APPARATUS REQUIRED:

S.No. Name of the Equipment/ Range Quantity


Component
1 PWM Trainer kit - DCL 08 - 1
.
2 CRO 30 MHz 1
.
3 Power supply 5V, ±12V 1
.
4 Patch chords - Required

THEORY:

The PWM and the carrier signals are connected to the inputs of a product detector, and then a sequence of
pulses having the width inversely proportional to the width of PWM pulse presents at output. When the Va
signal passes through the low-pass filter, a demodulated signal is obtained. The input signal is PWM so the
ON time of the signal. In this demodulation technique during the ON time of PWM signal one counter is
enabled. The output of DAC corresponds to the amplitude of input signal. Thus at the output we get the
original modulating signal extracted from PWM wave.

TABULATION

Time period in Sec


Parameters Amplitude in V
TON TOFF
Input Signal 1(1 KHz)

PWM Output

Buffer Output

PWM Demodulation Output

Filter Output
MODEL GRAPH

VIVA QUESTIONS:
1. What is PWM?
2. What are the applications of PWM?
3. What are the advantages of PWM or pulse amplitude modulation?
4. What are the disadvantages of pulse width modulation?
5. What factors affect pulse width modulation?
6. State and explain the types of pulse width modulation.
7. What are the advantages and disadvantages of PWM

RESULT
Thus the PAM signal was generated using PWM modulator and the message signal was detected
from PWM signal by using PWM demodulator.
CONNECTION DIAGRAM
PULSE POSITION MODULATION AND DEMODULATION

AIM:
To generate a PPM signal using PPM modulator and detect the message signal from
PPM signal by using PPM demodulator.

APPARATUS REQUIRED:

S.No. Name of the Equipment/ Range Quantity


Component
1 PPM Trainer kit - DCL 08 - 1
.
2 CRO 30 MHz 1
.
3 Power supply 5V, ±12V 1
.
4 Patch chords - Required

THEORY:

Pulse Position Modulation (PPM) is an analog modulating scheme in which the amplitude and width of the
pulses are kept constant, while the position of each pulse, with reference to the position of a reference pulse
varies according to the instantaneous sampled value of the message signal. The transmitter has to send
synchronizing pulses (or simply sync pulses) to keep the transmitter and receiver in synchronism. These sync
pulses help maintain the position of the pulses. Due to the set and reset signals applied to the flip-flop, we get
a PWM signal at its output. The PWM signal can be demodulated using the PWM demodulator.

TABULATION

Time period in Sec


Parameters Amplitude in V
TON TOFF
Input Signal 1(1 KHz)

PPM Output

Buffer Output

PPM Demodulation Output

Filter Output
MODEL GRAPH
VIVA QUESTIONS:

1. What is PPM?
2. What are the applications of PPM?
3. What are the advantages of PPM or pulse position modulation?
4. What are the disadvantages of pulse position modulation?
5. What factors affect pulse position modulation?
6. State and explain the types of pulse position modulation.
7. What are the advantages and disadvantages of PPM?

RESULT
Thus the PPM signal was generated using PPM modulator and the message signal was detected from
PPM signal by using PPM demodulator.
CONNECTION DIAGRAM
8. DIGITAL MODULATION OF ASK, PSK, FSK

AIM:
To study the carrier modulation techniques using ASK, PSK, FSK methods.

APPARATUS REQUIRED:

S.No. Name of the Equipment/ Range Quantity


Component
1 DCL-06 - 1
.
2 CRO 30 MHz 1
.
3 Power supply 5V, ±12V 1
.
4 Patch chords - Required

THEORY

Amplitude Shift Keying

ASK In this form of modulation the sine carrier takes 2 amplitude values, determined by the binary data
signal. Usually the modulator transmits the carrier when the data bit is "1", it completely removes it when the
bit is "0" (fig 2.1). There are also ASK shapes called "multi-level", where the amplitude of the modulated
signal takes more than 2 values. The demodulation can be coherent or non-coherent. In the first case, more
complex as concerns the circuits but more effective as against the noise effect, a product demodulator
multiplies the ASK signal by the locally regenerated carrier. In the second case the envelope of the ASK
signal is detected via diode. In both cases the detector is followed by a low pass filter, which removes the
residual carrier components, and a threshold circuit which squares the data signal.

Phase Shift Keying

In this kind of modulation, the sine carrier takes 2 or more phase values, directly determined by the binary
data signal (2-phase modulation) or by the combination of a certain number of bits of the same data signal(N-
phase modulation). In 2-phase PSK modulation, called 2-PSK, or Binary PSK (BPSK), or Phase Reversal
Keying (PRK), the sine carrier takes 2 phase values, determined by the binary data signal.

Frequency Shift Keying

FSK is a scheme of frequency modulation. The output of a FSK modulated wave is high in frequency for a
binary High input and is low in frequency for a binary Low input. The binary 1s and 0s are called Mark and
Space frequencies. Frequency-shift keying (FSK) is commonly used over telephone lines for caller ID
(displaying callers' numbers) and remote metering applications.
CONNECTION DIAGRAM
CONNECTION DIAGRAM

VIVA QUESTIONS

1. Compare ASK, FSK,PSK


2. Discuss the operation of PSK
3. Give the applications of shift keying signals.
4. What is the need of shift keying in digital communication?
5. What are the applications of amplitude shift keying?
RESULT:

Thus the carrier modulation signal of ASK, FSK, PSK has been generated and verified.
CONNECTION DIAGRAM
9. DELTA MODULATION AND DEMODULATION

AIM:
To transmit an analog message signal in its digital form and again reconstruct back the
original analog message signal at receiver by using Delta modulator.
APPARATUS REQUIRED:
S.No Name of the Equipment/ Component Range Quantity
Delta modulation/ demodulation
1. - 1
trainer kit, DCL 07
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required
THEORY:
Delta modulation uses a single bit PCM code to achieve digital transmission of analog
signal. With conventional PCM, each code is a binary representative of both the sign and
magnitude of a particular sample. The algorithm of delta modulation is simple if the current
sample is smaller than the previous sample a logic0 is transmitted. If the current sample is larger
than the previous sample logic 1 is transmitted.
ADVANTAGES:
1. Simple system/circuitry - Cheap
2. Single bit encoding allows us to increase the sampling rate or to transmit
more information at some sampling rate for the given system BW.
DISADVANTAGES:
1. Noise and distortion.
2. Major drawback is that it is unable to pass DC information.
APPLICATIONS:
Digital voice storage, Voice transmission, Radio communication devices such TV remotes.
Adaptive delta modulation
is delta modulation system where the step size of DAC is
automatically varied, depending on the amplitude characteristics of the analog input signal. A
common algorithm for an adaptive delta modulator is when three consecutive 1s or 0soccurs the
step size of the DAC is increased or decreased by a factor of 1.5
APPLICATION:
Audio communication syste
MODEL GRAPH

TABULATION

Time Period
Parameter Amplitude (V) Frequency in Hz
in seconds
Input signal

Integrator 1 output

Sampler output

Integrator 1 output

Filter output

Demodulated output
PROCEDURE:
1. Refer to the block diagram and carry out the following connections
2. Connect the power supply with proper polarity to the kit DCL-07 and switch it ON.
3. Select the sine wave input 250Hz or through port P1 and connect port 250Hz to
port IN of input buffer.
4. Connect the output of buffer ‘OUT’ to digital samples input port ‘IN1’.
5. Then select clock rate of 8 KHz by pressing switch S1. Selected clock is indicated
by LED glow.
6. Keep switch S2 in (∆) delta position.
7. Connect output of a digital sampler port ‘OUT’ to input port ‘IN’ of integrator.
8. Connect output of integrator 1 port ‘OUT’ to input port ‘IN2’ of digital sampler.
9. The digital sampler ‘OUT’ is given to the input of the output buffer.
10. The output of the buffer is given to the second order input of the fourth
order Butterworth filter.
VIVA QUESTIONS:
1. Define Delta modulation? Why it is better?
2. What is granular noise? Define slope overload?
3. When granular noise and slope overload occur in Delta modulation?
4. What is Adaptive Delta Modulation and what are the advantages?
5. Compare all Digital pulse modulation techniques (PCM, DPCM, DM and ADM)
6. In digital transmission, the modulation technique that requires minimum
bandwidth is …………
7. In Delta Modulation, the bit rate is ………….
8. What are the applications of DM?
9. Mention the advantage of DM?
10. What is temporal waveform coding?

RESULT:
Thus the analog message signal in its digital form was transmitted and again the
original analog message signal was reconstructed at receiver by using Delta
modulator and Demodulator.
SIMULATION WAVEFORM
SIMULATION OF ASK GENERATION AND DETECTION SCHEME

AIM: To generate and demodulate amplitude shift keyed (ASK) signal using MATLAB.

Theory

Generation of ASK

Amplitude shift keying - ASK - is a modulation process, which imparts to a sinusoid two or
more discrete amplitude levels. These are related to the number of levels adopted by the
digital message. For a binary message sequence there are two levels, one of which is typically
zero. The data rate is a sub-multiple of the carrier frequency. Thus the modulated waveform
consists of bursts of a sinusoid. One of the disadvantages of ASK, compared with FSK and
PSK, for example, is that it has not got a constant envelope. This makes its processing (eg,
power amplification) more difficult, since linearity becomes an important factor. However, it
does make for ease of demodulation with an envelope detector.
Demodulation

ASK signal has a well defined envelope. Thus it is amenable to demodulation by an


envelope detector. Some sort of decision-making circuitry is necessary for detecting the
message. The signal is recovered by using a correlator and decision making circuitry is used
to recover the binary sequence.
Algorithm

Initialization commands

ASK modulation

1. Generate carrier signal.

2. Start FOR loop

3. Generate binary data, message signal(on-off form)

4. Generate ASK modulated signal.

5. Plot message signal and ASK modulated signal.

6. End FOR loop.

7. Plot the binary data and carrier.


ASK demodulation

1. Start FOR loop

2. Perform correlation of ASK signal with carrier to get decision variable

3. Make decision to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’

4. Plot the demodulated binary data.

Program
%ASK Modulation

clc;
clear all;
close all;
%GENERATE CARRIER SIGNAL
Tb=1; fc=10;

t=0:Tb/100:1;
c=sqrt(2/Tb)*sin(2*pi*fc*t);
%generate message signal

N=8;
m=rand(1,N);
t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2]
if m(i)>0.5

m(i)=1;
m_s=ones(1,length(t));

else
m(i)=0;
m_s=zeros(1,length(t));

end
message(i,:)=m_s;
%product of carrier and
message ask_sig(i,:)=c.*m_s;
t1=t1+(Tb+.01); t2=t2+
(Tb+.01);
%plot the message and ASK signal
subplot(5,1,2);axis([0 N -2 2]);plot(t,message(i,:),'r');
title('message signal');xlabel('t--->');ylabel('m(t)');grid on hold on
subplot(5,1,4);plot(t,ask_sig(i,:));
title('ASK signal');xlabel('t--->');ylabel('s(t)');grid on hold on

end

hold off

%Plot the carrier signal and input binary data subplot(5,1,3);plot(t,c);


title('carrier signal');xlabel('t--->');ylabel('c(t)');grid on
subplot(5,1,1);stem(m);
title('binary data bits');xlabel('n--->');ylabel('b(n)');grid on

% ASK Demodulation

t1=0;t2=Tb for i=1:N t=[t1:Tb/100:t2]


%correlator x=sum(c.*ask_sig(i,:));
%decision device if x>0 demod(i)=1;
else demod(i)=0; end t1=t1+
(Tb+.01); t2=t2+(Tb+.01); end
%plot demodulated binary data bits
subplot(5,1,5);stem(demod);
title('ASK demodulated signal'); xlabel('n--->');ylabel('b(n)');grid on

Result
The program for ASK modulation and demodulation has been simulated in MATLAB and
necessary graphs are plotted.
SIMULATION WAVEFORM
SIMULATION OF BPSK GENERATION AND DETECTION SCHEMES

Aim:

To generate and demodulate Binary phase shift keyed (BPSK) signal using MATLAB

Generation of PSK signal

PSK is a digital modulation scheme that conveys data by changing, or modulating, the phase
of a reference signal (the carrier wave). PSK uses a finite number of phases, each assigned a
unique pattern of binary digits. Usually, each phase encodes an equal number of bits. Each
pattern of bits forms the symbol that is represented by the particular phase. The demodulator,
which is designed specifically for the symbol-set used by the modulator, determines the
phase of the received signal and maps it back to the symbol it represents, thus recovering the
original data.

In a coherent binary PSK system, the pair of signal S1(t) and S2 (t) used to represent binary
symbols 1 & 0 are defined by

S1 (t) = √2Eb/ Tb Cos 2πfct

S2 (t) =√2Eb/Tb (2πfct+π) = - √ 2Eb/Tb Cos 2πfct where 0 ≤ t< Tb and

Eb = Transmitted signed energy for bit

The carrier frequency fc =n/Tb for some fixed integer n.

Antipodal Signal:

The pair of sinusoidal waves that differ only in a relative phase shift of 180° are
called antipodal signals.

BPSK Transmitter
Program
% BPSK modulation
clc;
clear all;
close
all;
%GENERATE CARRIER
SIGNAL Tb=1;
t=0:Tb/100:Tb; fc=2;
c=sqrt(2/Tb)*sin(2*pi*fc
*t);
%generate message signal
N=8;
m=rand(1,
N);
t1=0;t2=Tb
for i=1:N
t=[t1:.01:t]
if m(i)>0.5 m(i)=1;
m_s=ones(1,length(t
)); else
m(i)=0;
m_s=-1*ones(1,length(t));
end
message(i,:)=m
_s;
%product of carrier and message
signal bpsk_sig(i,:)=c.*m_s;
%Plot the message and BPSK modulated signal
subplot(5,1,2);axis([0 N -2 2]);plot(t,message(i,:),'r');
title('message signal(POLAR form)');xlabel('t---
>');ylabel('m(t)'); grid on; hold on;
subplot(5,1,4);plot(t,bpsk_sig(i,:));
title('BPSK signal');xlabel('t--->');ylabel('s(t)');
grid on; hold on;
The input binary symbols are represented in polar form with symbols 1 & 0 represented
by constant amplitude levels √Eb& -√Eb. This binary wave is multiplied by a sinusoidal carrier
in a product modulator. The result in a BSPK signal.
BSPK Receiver:

The received BPSK signal is applied to a correlator which is also supplied with a
locally generated reference signal c1 (t). The correlated o/p is compared with a threshold of
zero volts. If x>
0, the receiver decides in favour of symbol 1. If x< 0, it decides in favour of symbol 0.
Algorithm
Initialization commands
BPSK modulation
1. Generate carrier signal.
2. Start FOR loop
3. Generate binary data, message signal in polar form
4. Generate PSK modulated signal.
5. Plot message signal and PSK modulated signal.
6. End FOR loop.
7. Plot the binary data and carrier.
BPSK demodulation
1. Start FOR loop
2. Perform correlation of PSK signal with carrier to get decision variable
3. Make decision to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’
4. Plot the demodulated binary data.
t1=t1+1.01; t2=t2+1.01;
end
hold off
%plot the input binary data and carrier
signal subplot(5,1,1);stem(m);
title('binary data bits');xlabel('n--->');ylabel('b(n)');
grid on;
subplot(5,1,3);plot(t,c);
title('carrier signal');xlabel('t--->');ylabel('c(t)');
grid on;
% PSK Demodulation
t1=0;t2=
Tb for
i=1:N
t=[t1:.01:
t2]
%correlator
x=sum(c.*bpsk_sig(i,:
));
%decision
device if x>0
demod(i)=1;
else
demod(i)
=0; end
t1=t1+1.0
1;
t2=t2+1.0
1; end
%plot the demodulated data bits
subplot(5,1,5);stem(demod);
title('demodulated data');xlabel('n--->');ylabel('b(n)');
grid on;
Result
Thus the program for BPSK modulation and demodulation has been simulated in
MATLAB and necessary graphs are plotted.
SIMULATION WAVEFORM
SIMULATION OF FSK GENERATION AND DETECTION SCHEME

Aim

To generate and demodulate frequency shift keyed (FSK) signal using MATLAB

Theory

Generation of

FSK

Frequency-shift keying (FSK) is a frequency modulation scheme in which digital


information is transmitted through discrete frequency changes of a carrier wave. The
simplest FSK is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit
binary (0s and 1s) information. With this scheme, the "1" is called the mark frequency and
the "0" is called the space frequency.

In binary FSK system, symbol 1 & 0 are distinguished from each other by transmitting one of
the two sinusoidal waves that differ in frequency by a fixed amount.

Si (t) = √2E/Tb cos 2πf1t 0≤ t ≤Tb

0 elsewhere

Where i=1, 2 &Eb=Transmitted energy/bit

Transmitted freq= ƒi = (nc+i)/Tb, and n = constant (integer), Tb = bit interval

Symbol 1 is represented by S1 (t) Symbol 0 is represented by S0 (t)

BFSK Transmitter

The input binary sequence is represented in its ON-OFF form, with symbol 1 represented by
constant amplitude of √Eb with & symbol 0 represented by zero volts. By using inverter in
the lower channel, we in effect make sure that when symbol 1is at the input, The two
frequency f1& f2 are chosen to be equal integer multiples of the bit rate 1/Tb. By summing
the upper & lower channel outputs, we get BFSK signal.
Program

% FSK Modulation

clc;

clear all;

close all;

%GENERATE CARRIER SIGNAL

Tb=1; fc1=2;fc2=5;

t=0:(Tb/100):Tb;

c1=sqrt(2/Tb)*sin(2*pi*fc1*t);

c2=sqrt(2/Tb)*sin(2*pi*fc2*t);

%generate message signal

N=8;

m=rand(1,N);

t1=0;t2=Tb

for i=1:N

t=[t1:(Tb/100):t2]

if m(i)>0.5

m(i)=1;

m_s=ones(1,length(t));

invm_s=zeros(1,length(t));

else

m(i)=0;

m_s=zeros(1,length(t));

invm_s=ones(1,length(t));

end

message(i,:)=m_s;
%Multiplier

fsk_sig1(i,:)=c1.*m_s;

fsk_sig2(i,:)=c2.*invm_s;

fsk=fsk_sig1+fsk_sig2;

%plotting the message signal and the modulated signal

subplot(3,2,2);axis([0 N -2 2]);plot(t, message(i,:),'r');

title('message signal');xlabel('t >');ylabel('m(t)');grid on;hold on;


BFSK Receiver

The receiver consists of two correlators with common inputs which are supplied
with locally generated coherent reference signals c1(t) and c2 (t).
The correlator outputs are then subtracted one from the other, and the resulting
difference x is compared with a threshold of zero volts. If x >0, the receiver decides in
favour of symbol 1 and if x <0, the receiver decides in favour of symbol 0.
Algorithm
Initialization
commands FSK
modulation
1. Generate two carriers signal.
2. Start FOR loop
3. Generate binary data, message signal and inverted message signal
4. Multiply carrier 1 with message signal and carrier 2 with inverted message signal
5. Perform addition to get the FSK modulated signal
6. Plot message signal and FSK modulated signal.
7. End FOR loop.
8. Plot the binary data and carriers.
FSK demodulation
1. Start FOR loop
2. Perform correlation of FSK modulated signal with carrier 1 and carrier 2 to get
two decision variables x1 and x2.
3. Make decision on x = x1-x2 to get demodulated binary data. If x>0, choose ‘1’ else
choose ‘0’.
4. Plot the demodulated binary

data. subplot(3,2,5);plot(t,fsk(i,:));

title('FSK signal');xlabel('t >');ylabel('s(t)');grid on;hold on;

t1=t1+(Tb+.01); t2=t2+(Tb+.01);

end

hold off

%Plotting binary data bits and carrier signal

subplot(3,2,1);stem(m);
title('binary data');xlabel('n >');
ylabel('b(n)');grid on;

subplot(3,2,3);plot(t,c1);

title('carrier signal-1');xlabel('t >');


ylabel('c1(t)');grid on;

subplot(3,2,4);plot(t,c2);

title('carrier signal-2');xlabel('t >');


ylabel('c2(t)');grid on;

% FSK Demodulation

t1=0;t2=Tb for i=1:N t=[t1:(Tb/100):t2]

%correlator

x1=sum(c1.*fsk_sig1(i,:));
x2=sum(c2.*fsk_sig2(i,:)); x=x1-x2;

%decision device

if x>0 demod(i)=1;
else demod(i)=0; end
t1=t1+(Tb+.01); t2=t2+(Tb+.01);
end
%Plotting the demodulated data bits
subplot(3,2,6);stem(demod);

title(' demodulated data');xlabel('n >');ylabel('b(n)'); grid on;

Result

Thus the program for FSK modulation and demodulation has been simulated in MATLAB
and necessary graphs are plotted.
SIMULATION WAVEFORM
11. SIMULATION OF DPSK, QPSK GENERATION SCHEMES

AIM: To simulate DPSK and QPSK Generation Schemes using MATLAB.

SOFTWARE REQUIRED: MATLAB

PROGRAM FOR DPSK GENERATION SCHEME:

clc;

clear all;

rng default

M = 6; % Alphabet size

dataIn = randi([0 M-1],1011,1);

% Random message

txSig = dpskmod(dataIn,M);

% Modulate

rxSig =

txSig*exp(2i*pi*rand());

dataOut = dpskdemod(rxSig,M);

errs = symerr(dataIn,dataOut)

errs = symerr(dataIn(2:end),dataIn(2:end))

figure subplot(2,2,1)

plot(dataIn)

title('DATA')

subplot(2,2,2)

plot(txSig)

title('DPSK SIGNAL')

subplot(2,2,3)

plot(rxSig)

title('Received DPSK')
subplot(2,2,4)

plot(dataOut)

title('DATA RECEIVED')
SIMULATION WAVEFORM
PROGRAM FOR QPSK GENERATION SCHEME: QPSK

clc;

clear all;

t=0:0.0001:0.25;

m=square(2*pi*10*t);

c1=sin(2*pi*60*t);

c2=sin(2*pi*60*t+180);

for i=1:2500

if(mod(i,1000))<500 s(i)=c1(i);

else

s(i)=-c2(i);

end

end

subplot(4,1,1);

plot(t,m,'k','linewidth',5);

title('polor representation of message 1 0 1 0 1 0');

xlabel('time'); ylabel('amplitude') subplot(4,1,2); plot(c1); title('frequency 1');

xlabel('time'); ylabel('amplitude'); subplot(4,1,3); plot(c2); title('frequency

2'); xlabel('time'); ylabel('amplitude'); subplot(4,1,4); plot(s);

title('quadrature phase shift keying');

xlabel('time');

ylabel('amplitude');

RESULT:

Thus the DPSK and QPSK generation schemes were simulated using MATLAB.
BPSKCONSTELLATION:
OBSERVATION (SIMULATION) OF SIGNAL CONSTELLATIONS OF
BPSK, QPSK AND QAM

Aim:

To plot the constellation diagram of digital modulation system BPSK, QPSK & QAM

using MATLAB.

APPARATUS REQUIRED:

1. PC

2. MATLAB SOFTWARE

THEORY:

A constellation diagram is a representation of a signal modulated by an arbitrary


digital modulation scheme. It displays the signal as a two dimensional scatter diagram in the
complex plane at symbol sampling instants. It can also be viewed as the possible symbols
that may be selected by a given modulation scheme as points in the complex plane.

PROGRAM: BPSK

clc;

clear all; close all; M=2; k=log2(M); n=3*1e5;

nsamp=8;

X=randint(n,1);

xsym = bi2de(reshape(X,k,length(X)/k).','left-msb');

Y_psk= modulate(modem.pskmod(M),xsym);

Ytx_psk = Y_psk;

EbNo=30;

SNR=EbNo+10*log10(k)-10*log10(nsamp);

Ynoisy_psk = awgn(Ytx_psk,SNR,'measured');

Yrx_psk = Ynoisy_psk; h1=scatterplot(Yrx_psk(1:nsamp*5e3),nsamp,0,'r.');

hold on;

scatterplot(Yrx_psk(1:5e3),1,0,'k*',h1);

title('constellation diagram BPSK');


legend('Received signal' ,'signal

constellation'); axis([-5 5 -5 5]);

hold off;

QPSKCONSTELLATION:

QAM CONSTELLATION:
Program for QPSK & QAM:

clc;

clear all;

close all;

M=16; k=log2(M);

n=3*1e5; nsamp=8;

X=randint(n,1);

xsym = bi2de(reshape(X,k,length(X)/k).','left-msb');

Y_qam= modulate(modem.qammod(M),xsym);

Y_qpsk= modulate(modem.pskmod(M),xsym);

Ytx_qam = Y_qam;

Ytx_qpsk = Y_qpsk; EbNo=30;

SNR=EbNo+10*log10(k)-10*log10(nsamp);

Ynoisy_qam = awgn(Ytx_qam,SNR,'measured');

Ynoisy_qpsk =

awgn(Ytx_qpsk,SNR,'measured'); Yrx_qam =

Ynoisy_qam;

Yrx_qpsk = Ynoisy_qpsk;

h1=scatterplot(Yrx_qam(1:nsamp*5e3),nsamp,0,'r.');

hold on;

scatterplot(Yrx_qam(1:5e3),1,0,'k*',h1);

title('constellation diagram 16 QAM');

legend('Received signal' ,'signal

constellation'); axis([-5 5 -5 5]);

hold off;

h2=scatterplot(Yrx_qpsk(1:nsamp*5e3),

nsamp,0,'r.');

hold on;
scatterplot (Yrx_qpsk(1:5e3),1,0,'k*',h2);

title('constellation diagram 16 PSK');

legend('Received signal' ,'signal

constellation'); axis([-5 5 -5 5]);

hold off;

title('constellation diagram 16 PSK');

legend('Received signal' ,'signal constellation');

axis([-5 5 -5 5]);

hold off;

RESULT:

Thus the constellation diagrams of digital modulation system BPSK, QPSK & QAM
are simulated & plotted in MATLAB.
OUTPUT

Enter the Generator Matrix: [1 0 0 0 1 0 1; 0 1 0 0 1 1 1; 0 0 1 0 1 1 0; 0 0 0 1 0 1 1]

g=

1 0 0 0 1 0 1

0 1 0 0 1 1 1

0 0 1 0 1 1 0

0 0 0 1 0 1 1

G =The Order of Linear block Code for given Generator Matrix is:

n =7 k =4

The Possible Code words are:

c=

0 0 0 0 0 0 0

0 0 0 1 0 1 1

0 0 1 0 1 1 0

0 0 1 1 1 0 1

0 1 0 0 1 1 1

0 1 0 1 1 0 0

0 1 1 0 0 0 1

0 1 1 1 0 1 0

1 0 0 0 1 0 1

1 0 0 1 1 1 0

1 0 1 0 0 1 1

1 0 1 1 0 0 0
1 1 0 0 0 1 0

1 1 0 1 0 0 1

1 1 1 0 1 0 0

1 1 1 1 1 1 1
12. SIMULATION OF LINEAR BLOCK CODING SCHEME

AIM:

To simulate and study the error control coding scheme of linear block code using MATLAB

ALGORITHM:

STEP 1: Give the generator matrix

STEP 2: Find the order of the linear block code for the given generator matrix

STEP 3: Obtain the possible code words

STEP 4: Find the minimum hamming distance

STEP 5: Give the received code word

STEP 6: Calculate the syndrome vector and compare it with transpose of


hamming matrix.
STEP 7: Find the error bit position and display the corrected code word

MATLAB CODE:
% Input Generator Matrix

g=input('Enter The Generator Matrix: ')

disp ('G = ')

disp ('The Order of Linear block Code for given Generator Matrix is:') [n,k] =
size(transpose(g))

fori = 1:2^k forj = k:-1:1

ifrem(i-1,2^(-j+k+1))>=2^(-j+k)

u(i,j)=1; else u(i,j)=0; end

end end

disp('The Possible Codewords are :')

c = rem(u*g,2)

disp('The Minimum Hamming Distance dmin for given Block Code is= ')

d_min = min(sum((c(2:2^k,:))'))
The Minimum Hamming Distance dmin for given Block Code is=

d_min = 3

Enter the Received Code Word:[1 0 0 0 1 0 0]

r=1 0 0 0 1 0 0

Hammimg Code ht =

1 0 1

1 1 1

1 1 0

0 1 1

1 0 0

0 1 0

0 0 1

Syndrome of a Given Code word

is : s = 0 0 1

The Error is in bit: 7

The Corrected Code word is:

1 0 0 0 1 0 1

% Code Word

r = input('Enter the Received Code Word:')


p = [g(:,n-k+2:n)];
h = [transpose(p),eye(n-k)];
disp('HammimgCode')
ht = transpose(h)
disp('Syndrome of a Given Codeword is :')
s= rem(r*ht,2)
for
i = 1:1:size(ht)
if(ht(i,1:3)==s)
r(i) = 1-r(i);
break;
end
end
disp('The Error is in
bit:') disp(i);
disp('The Corrected Codeword is
:') disp(r);

RESULT:
Thus the error control coding scheme of linear block code is studied using
MATLAB.
OUTPUT

COMPUTATION OF CODE VECTORS FOR A CYCLIC CODE

Msg=

1001

1010

1011

Code =

1101001

0111010

0001011

SYNDROME DECODING

Recd=

1011110

Syndrome=7(decimal), 1 1 1(binary

Parmat=

1001011

0101110

0010111

Corrvect=

0000010

Correctedcode=

1011100
13. SIMULATION OF ERROR CONTROL USING

CYCLICCODE AIM:

a. To generate parity check matrix & generator matrix for a (7,4) Hamming code.
b. To generate parity check matrix given generator polynomial g(x) = 1+x+x3.

c. To determine the code vectors.

d. To perform syndrome decoding

PROGRAM:

Generation of parity check matrix and generator matrix for a (7, 4) Hamming

code. [h,g,n,k] = hammgen(3);

Generation of parity check matrix for the generator polynomial g(x) = 1+x+x3.

h1 = hammgen(3,[1011]);

Computation of code vectors for a cyclic code

clc;

close all;

n=7;

k=4;

msg=[1 0 0 1; 1 0 1 0; 1 0 1 1];

code = encode(msg,n,k,'cyclic');

msg code

Syndrome decoding

clc;

close all; q=3; n=2^q-1; k=n-q;

parmat = hammgen(q);

% produce parity-check matrix trt = syndtable(parmat);

% produce decoding table

recd = [1 0 1 1 1 1 0 ]

%received vector syndrome = rem(recd * parmat',2);


syndrome_de = bi2de(syndrome, 'left-msb');

%convert to decimal

disp(['Syndrome = ',num2str(syndrome_de),.....

' (decimal), ',num2str(syndrome),' (binary) ']);

corrvect = trt(1+syndrome_de, :);

%correction vector

correctedcode= rem(corrvect+recd,2);

parmat corrvect

correctedcode

CONVOLUTIONAL CODING
12.b. SIMULATION OF CONVOLUTIONAL CODING SCHEME

AIM

To simulate convolutional coding scheme using MATLAB Simulink tool.

APPARATUS REQUIRED:

PC with MATLAB Software

ALGORITHM:

Simulate the link by following these steps:


1. Generate binary data.
2. Encode the data with a rate 2/3 convolutional code.
3. Modulate the encoded data.
4. Pass the signal through an AWGN channel.
5. Demodulate the received signal.
6. Decode the demodulated signal by using a Viterbi decoder.
7. Collect the error statistics.

THEORY:

Generating Random Data


The Bernoulli Binary Generator block produces the information source for this
simulation. The block generates a frame of three random bits at each sample time. The
Samples per frame parameter determines the number of rows of the output frame.
Convolutional Encoding with Puncturing
The Convolutional Encoder block encodes the data from the Bernoulli Binary
Generator. This example uses the same code as described in Soft-Decision Decoding.
The puncture pattern is specified by the Puncture vector parameter in the mask.
The puncture vector is a binary column vector. A 1 indicates that the bit in the
corresponding position of the input vector is sent to the output vector, while a 0
indicates that the bit is removed.
For example, to create a rate 3/4 code from the rate 1/2, constraint length 7
convolutional code, the optimal puncture vector is [1 1 0 1 1 0].' (where the.' after the
vector indicates the transpose). Bits in positions 1, 2, 4, and 5 are transmitted, while bits in
positions 3 and 6 are removed. Now, for every 3 bits of input, the punctured code
generates 4 bits of output (as opposed to the 6 bits produced before puncturing). This makes
the rate 3/4.
In this example, the output from the Bernoulli Binary Generator is a column vector
of length 3. Because the rate 1/2 Convolutional Encoder doubles the length of each vector,
the length of the puncture vector must divide 6.
BIT ERROR RATE (BER)

Transmitting Data
The AWGN Channel block simulates transmission over a noisy channel. The
parameters for the block are set in the mask as follows:
The Mode parameter for this block is set to Signal to noise ratio (Es/No).
The Es/No parameter is set to 2 dB. This value typically is changed from one
simulation run to the next.
The preceding modulation block generates unit power signals so the
Input signal power is set to 1 Watt.
The Symbol period is set to 0.75 seconds because the code has rate 3/4.
Demodulating
In this simulation, the Viterbi Decoder block is set to accept unquantized inputs.
As a result, the simulation passes the channel output through a Simulink® Complex
to Real- Image block that extracts the real part of the complex samples.
Viterbi Decoding of Punctured Codes
The Viterbi Decoder block is configured to decode the same rate 1/2 code
specified in the Convolutional Encoder block.
In this example, the decision type is set to Unquantized. For codes without
puncturing, you would normally set the Traceback depth for this code to a value close
to 40. However, for decoding punctured codes, a higher value is required to give the
decoder enough data to resolve the ambiguities introduced by the punctures. Since the
punctured bits are not transmitted, there is no information to indicate their values. As a
result they are ignored in the decoding process.
The Puncture vector parameter indicates the locations of the punctures or the
bits to ignore in the decoding process. Each 1 in the puncture vector indicates a
transmitted bit while each 0 indicates a puncture or the bit to ignore in the input to
the decoder. In general, the two Puncture vector parameters in the Convolutional
Encoder and Viterbi Decoder must be the same.
Calculating the Error Rate
The Error Rate Calculation block compares the decoded bits to the original
source bits. The output of the Error Rate Calculation block is a three-element
vector containing the calculated bit error rate (BER), the number of errors
observed, and the number of bits processed.
In the mask for this block, the Receive delay parameter is set to 96,
because the Traceback depth value of 96 in the Viterbi Decoder block creates a delay
of 96. If there were other blocks in the model that created delays, the Receive delay
would equal the sum of all the delays.
BER simulations typically run until a minimum number of errors have occurred,
or until the simulation processes a maximum number of bits. The Error Rate
Calculation block uses its Stop simulation mode to set these limits and to control the
duration of the simulation.
OUTPUT (AM)
RESULT:

Thus the convolution coding scheme was simulated using MATLAB Simulink
tool

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