4EC4-21 ADC Lab AS-16a

TABLE OF CONTENTS
S.No. Contents Page No.

1 Course Objectives 2
2 Course Outcomes 2
3 Vision & Mission of Institute 2
5 Vision & Mission of Department 2
6 Programme Educational Objectives 2
7 Programme Outcomes 3
8 Experiment List 4
9 Experiment Beyond Curriculum 4
11 Mapping of Course Outcomes with Programme Outcomes 5
12 Mapping of Course Outcomes with Programme Specific Objectives 5
13 Prerequisite of Lab 5
14 Marking Scheme 5
15 Rotor Plan 6
16 Lab Plan 7
17 Lab Evaluation Sheet 8-9
18 Lab Rules (Do’s & Don’t) 10
19 Instructions (Safety Precautions) 11
20 Zero Lab 12
21 Experiment – 1 13-17
34 Conclusion
DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING Page | 1

COURSE OBJECTIVES
1. To understand the fundamental concept of analog and digital transmission of signals.
2. To understand and compare different analog and digital modulation techniques.
3. To understand basic concept of baseband data transmission.
4. To understand and analyze different multiplexing technique.
5. To understand the link between analog and digital signals.
COURSE OUTCOMES
After completion of this course students will be able to
1 Understand different analog modulation schemes and evaluate modulation index
2 Able to understand the principle of superhetrodyne Receiver
3 Develop time division multiplexing concepts in real time applications
4 Develop and able to comprehend different data formatting schemes
5 Comprehend and analyze the concepts of different digital modulation techniques in
communication.
VISION & MISSION OF THE INSTITUTION

VISION:-
To create knowledge based society with scientific temper, team spirit and dignity of labor
to face the global competitive challenges.
MISSION:-
To evolve and develop skill based systems for effective delivery of knowledge so as to
equip young professionals with dedication & commitment to excellence in all spheres of
life.
VISION & MISSION OF DEPARTMENT

VISION:
To establish an acknowledged department of academics in the field of Electronics and
Communication Engineering.
MISSION:
To provide comprehensive technical education with the exposure of latest technology
providing the learning environment for faculty members & students.
PROGRAMME EDUCATIONAL OBJECTIVES

The Department of Electronics & Communication Engineering has defined its PEOs as
follows:-
PEO1: The graduates will be competent enough to apply current knowledge and skill to
solve problems of the society.
PEO2: The graduates will be capable of identifying, formulating, analyzing and creating
engineering solutions using modern tools to develop novel products solutions.
PEO3: The graduates will be professional, lifelong learning skills, ethics, research skills
and leadership for independent or team working within a group.

PROGRAMME SPECIFIC OUTCOMES
PSO1: Graduates possesses the ability to understand and apply basic knowledge of core
Electronics & Communication Engineering for the benefit of society.
PSO2: Graduates will be proficient to apply electronic modern IT tools for the design and
analysis of complex electronic systems in furtherance to research activities.
PSO3: The ability to be adaptable to the multidisciplinary nature at workplace, develop
excellent Interpersonal Skills & Leadership qualities that benefits the individual &
organization.
PROGRAMME OUTCOMES
Programmed Outcome:
a.The graduates will be proficient through the knowledge application in the fields of
mathematics, science, and engineering.
b. The graduates will be proficient in designing of application, conduction of
experiments and skillful to analyze and interpreting the data.
c.The graduates will be able to meet the desired needs within realistic constraints such as
economic, environmental, social, political, ethical, health and safety, manufacturability and
sustainability through the designing of a component, a system or process.
d. The graduates will be able to function in multidisciplinary teams.
e.The graduates will be able to identify, formulate and solve engineering problems.
f. The graduates will be made robust & susceptible enough to handle all challenges in the
field of Electronics & Communication Engineering.
g. The graduates will be able to communicate their views effectively.
h. The graduates will be able to get the extensive education to understand the impact of
engineering solutions in a global, economic, environmental and societal context.
i. The graduates will be able to recognize the need of engineering covering its all spheres
engaging themselves in life-long learning.
j. The graduates will be able to get awareness about the contemporary issues and strive for
their remedies
k. The graduates will be able to use the techniques, skills and modern engineering tools
as per the need of engineering practices.
l. The graduates will be able to visualize the use of electronic design using latest products
and advanced technologies to meet the various industrial requirements.
m. The graduates will be able to design systems or processes in analyzing and interpreting
the system performance.

EXPERIMENT LIST
Sr. No. Contents

1 Observe the Amplitude modulated wave form & measure modulation index and
Demodulation of AM signal.
2 Harmonic analysis of Amplitude Modulated wave form.
3 Generation & Demodulation of DSB – SC signal.

4 Modulate a sinusoidal signal with high frequency carrier to obtain FM signal and
demodulation of the FM signal.
5 Verification of Sampling Theorem.
6 To study & observe the operation of a super heterodyne receiver.
7 PAM, PWM & PPM: Modulation and demodulation.
8 To observe the transmission of four signals over a single channel using TDM
PAM method.
9 To study the PCM modulation & demodulation and study the effect of channel
like attenuation, noise in between modulator & demodulator through the
experimental setup..
10 To study the 4 channel PCM multiplexing & de-multiplexing in telephony system.
11 To study the Delta & Adaptive delta modulation & demodulation and also study
the effect of channel like attenuation, noise in between modulator & demodulator
through the experimental setup.
12 To perform the experiment of generation and study the various data formatting
schemes (Unipolar, Bipolar, Manchester, AMI etc.)
13 To perform the experiment of generation and detection of ASK, FSK, BPSK,
DBPSK signals with variable length data pattern.
EXPERIMENT BEYOND CURRICULUM
Sr. No. Contents

1 Perform QPSK modulation technique in MATLAB
2 Perform the DPSK modulation technique in MATLAB

CO-PO MAPPING MATRIX OF COURSE
PO-9 (Individual and

(Design/Development
PO-5 (Modern Tool )
PO-6 (Engineer and
PO-7 (Environment
and Sustainability)
PO-1 (Engineering
PO-12 (Life-Long
(Communication)
Management and
Investigations of
PO-4 (Conduct
PO-2 (Problem
PO-11 (Project
PO-8 (Ethics)
Team Work)
Knowledge)
Learning)
Analysis)
Society)
PO-10
PO-3

CO-1 3 3 1 1 1 - - 1 2 2 - 2
CO-2 3 3 1 1 1 - - 1 2 2 - 2
CO-3 3 3 1 1 1 - - 1 2 2 - 2
CO-4 3 3 1 1 1 - - 1 2 2 - 2
CO-5 3 3 1 1 1 - - 1 2 2 - 2
CO-PSO Mapping Matrix of course

(Apply Basic
Leadership)
Knowledge)
(Modern IT
(Skills &
PSO-2
PSO-1
PSO-3
Tools)
COs and
LOs
CO-1 2 - -
CO-2 2 - -
CO-3 2 1 1
CO-4 2 - -
CO-5 2 1 -
PREREQUISITE OF LAB
Sr.No. Name of Lab/Subject Code
1 Signal Processing Lab 3EC4-23
MARKING SCHEME
I Mid term II Mid term Attendance Performance Common RTU Exam. Total
Pool
Pra. Viva Pra. Viva Pra. Viva

22 8 22 8 4 11 15 22 8 75
ROTOR PLAN
ROTOR 1
1. Observe the Amplitude modulated wave form & measure modulation index and
demodulation of AM signal.
2. Harmonic analysis of Amplitude Modulated wave form.
3. Generation & Demodulation of DSB – SC signal.
4. Modulate a sinusoidal signal with high frequency carrier to obtain FM signal and
demodulation of the FM signal.
5. Verification of Sampling Theorem.
6. To study & observe the operation of a super heterodyne receiver.
ROTOR 2
7. PAM, PWM & PPM: Modulation and demodulation.
8. To observe the transmission of four signals over a single channel using TDM-PAM
method.
9. To study the PCM modulation & demodulation and study the effect of channel like
attenuation, noise in between modulator & demodulator through the experimental
setup.
10. To study the 4 channel PCM multiplexing & de-multiplexing in telephony system.
11. To study the Delta & Adaptive delta modulation & demodulation and also study the
effect of channel like attenuation, noise in between modulator & demodulator
through the experimental setup.
12. To perform the experiment of generation and study the various data formatting
schemes (Unipolar, Bipolar, Manchester, AMI etc.)
13. To perform the experiment of generation and detection of ASK, FSK, BPSK,
DBPSK signals with variable length data pattern
14. Perform QPSK modulation technique in MATLAB
15. Perform the DPSK modulation technique in MATLAB

LAB PLAN
Total number of experiment 13
Total number of turns required 13
Number of turns required for

Experiment Number Turns Scheduled Lab
Experiment. 1 1
Experiment. 2 1
Experiment. 3 1
Experiment. 4 1
Experiment. 5 1
Experiment. 6 1
Experiment. 7 1
Experiment. 8 1
Experiment. 9 1
Experiment. 10 1
Experiment. 11 1
Experiment. 12 1
Experiment. 13 1

LAB EVALUATION SHEET
POORNIMA COLLEGE OF ENGINEERING, JAIPUR
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINNERING
EVALUTION SHEET (To Be Pasted on board)
Session: 2018-19 Lab Code: 4EC4-21
Year/Semester: Lab Name: Analog and Digital Communication Lab
Cycle:- II Name of the faculty:
EXP 1 EXP 2 EXP 3 EXP 4 EXP 5 EXP 6

S. Name of
Reg. No.
Marks
Marks
Marks
Marks
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Marks
No. Student
Date
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LAB RULES (DO’S & DON’T)
RESPONSIBILITIES OF USERS
Users are expected to follow some fairly obvious rules of conduct:
ALWAYS
 Enter the lab on time and leave at proper time.
 Wait for the previous class to leave before the next class enters.
 Keep the bag in the respective racks.
 Utilize lab hours in the corresponding.
 Turn off the instrumentation before leaving the lab unless a member of lab staff has
specifically told you not to do so.
 Leave the labs at least as nice as you found them.
 If you notice a problem with a piece of equipment (e.g. a CRO doesn't respond)
please report it to lab staff immediately. Do not attempt to fix the problem yourself.
NEVER
 Don't damage the equipment.
 If you're going to be away from your instrument for more than 10 or 15 minutes,
switch off before leaving. This is for the security of your experiment and to ensure
that others are able to use the lab resources while you are not.
 No food or drink is allowed in the lab or near any of the equipment. Aside from the
fact that it leaves a mess and attracts pests. If you need to eat or drink, take a break
and do so in the canteen.
 Don’t bring any external material in the lab, except your lab record, copy and books.
 Don’t bring the mobile phones in the lab. If necessary then keep them in silence
mode.
 Please be considerate of those around you, especially in terms of noise level. While
labs are a natural place for conversations of all types, kindly keep the volume turned
down.
If you are having problems or questions, please go to either the faculty, lab in-charge or the
lab supporting staff. They will help you. We need your full support and cooperation for
smooth functioning of the lab.

INSTRUCTIONS (SAFETY PRECAUTIONS)
BEFORE ENTERING IN THE LAB

 All the students are supposed to prepare the theory regarding the next experiment.
 Students are supposed to bring the practical file and the lab copy.
 Previous programs should be written in the practical file.
 All the students must follow the instructions, failing which he/she may not be
allowed in the lab.
WHILE WORKING IN THE LAB
 Adhere to experimental schedule as instructed by the lab in-charge.
 Get the previously executed program signed by the instructor.
 Get the output of the current program checked by the instructor in the lab copy.
 Each student should work on his/her assigned computer at each turn of the lab.
 Take responsibility of valuable accessories.
 Concentrate on the assigned practical and do not play games
 If anyone caught red handed carrying any equipment of the lab, then he/she will have
to face serious consequences.

POORNIMA COLLEE OF ENGINEERING, JAIPUR
DEPARTMENT OF ELECTRONICS & COMMUNICATION
ZERO LAB
1) NAME OF LAB WITH CODE: Analog and Digital Communication Lab & 4EC4-
21
2) INTRODUCTION TO LAB: Communication has been one of the deepest needs of the
human race throughout recorded history. It is essential to forming social unions, to educating
the young, and to expressing a myriad of emotions and needs. Good communication is
central to a civilized society.
Digital communication systems, by definition, are communication systems that use such a
digital1 sequence as an interface between the source and the channel input (and similarly
between the channel output and final destination placing a binary interface between source
and channel. The source encoder converts the source output to a binary sequence and the
channel encoder (often called a modulator) processes the binary sequence for transmission
over the channel. The channel decoder (demodulator) recreates the incoming binary
sequence (hopefully reliably), and the source decoder recreates the source output.
In modulation, the message signal changes the amplitude, frequency, or phase of the carrier
signal.
2) CONNECTION WITH PREVIOUS YEAR AND NEXT YEAR: This lab is related
to Signal & System subject and Signal Processing Lab in which basics and different
modulation technique are studied and in next year it is related to wireless communication
subject as well as wireless communication Lab in which advancement digital
communication is learnt.
3) LAB SCHEDULE PER WEEK: 2 Hours
4) ANY OTHER INFORMATION
PLACE: PCE, JAIPUR DATE:

SIGNATURE OF FACULTY

EXPERIMENT NO:-1
AIM:- . Observe the Amplitude modulated wave form & measure
modulation index and demodulation of AM signal.
APPARATUS:-
NAME OF THE
QUANTITY
COMPONENT/EQUIPMENT
ST-2201 FOR 1
TRANSMITTER
1
ST-2202 FOR RECEIVER
CONNECTING WIRES AND 6-7

PROBES
CRO 1
FUNCTIONGENERATOR 2
REGULATED 1
POWERSUPPLY
THEORY:-
Amplitude modulation (AM) is a technique used in electronic communication, most
commonly for transmitting information via a radio carrier wave. AM works by varying the
strength of the transmitted signal in relation to the information being sent.
For example, changes in signal strength may be used to specify the sounds to be
reproduced by a loudspeaker, or the light intensity of television pixels. Contrast this with

frequency modulation, in which the frequency is varied, and phase modulation, 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 amplitude modulated (AM) wave is defined by:
V(t) =A Cos(2πft+φ)
The demodulation circuit is used to recover the message signal from the incoming AM
wave at the receiver. An envelope detector is a simple and yet highly effective device that
is well suited for the demodulation of AM wave, for which the percentage modulation is
less then 100%. Ideally, an envelope detector produces an output signal that follows the
envelop of the input signal wave form exactly; hence, the name. Same version of this
circuit is used in almost all commercial AM radio receivers.
M= (Emax+ Emin)*100
(Emax -E min)
Where, Emax and E min are the maximum and minimum amplitudes of the modulated wave.
BLOCK DIAGRAM OF AM TRANSMITTER:

Fig 1: CIRCUIT FOR AMPLITUDE MODULATION
BLOCK DIAGRAM OF AM RECEIVER:

FIG 2: CIRCUIT FOR AM DEMODULATOR

PROCEDURE:-
1. Switch on the trainer and check the O/P of carrier generator on oscilloscope.
2. Connect 1 KHz with 2 Volts A.F signal at AF I/P to the modulator circuit. Connect the
carrier signal at carrier I/P of modulator circuit.
3. Observe the modulator output signal at AM O/P Spring by making necessary changes in
A.F. signal.
4. Vary the modulating frequency and amplitude and observe the effects on the modulated
waveform.
5. The depth of modulation can be varied using the variable knob (potentiometer) provided at
A.F. input.
6. The percentage of modulation or modulation factor can be calculated using the following
formulas.
7. Find the value of R from fm=1/(2*Pi*R*C) , C=0.1µF

8. Connect the circuit diagram as shown in Fig.2.
9. Feed the AM wave to the demodulator circuit and observe the output
10. Note down frequency and amplitude of the demodulated output waveform.
11. Draw the demodulated wave form.
OBSERVATIONS:-
Table 1:fm= 1KHz, fc=11KHz, Ac=15 V p-p.
S.No. Vm(Volts) Emax(volts) Emin(Volts) m %m (mx100)

Table 2: Am= 4 Vp-p, fc =11KHz, Ac=15 Vp-p.
S.No. fm(KHz) Emax(volts) Emin(Volts) m %m (mx100)
PRECAUTIONS:-
1. Connect the circuit as shown in the circuit diagram.
2. Apply the required voltages wherever needed.
3. Do not apply stress on the components.
MODEL WAVE FORMS
RESULT:-
We have observed results of amplitude modulation and demodulation also draw the
waveform modulation index (Ma) = 0.414
VIVA –QUESTION:-
1. What is Modulation? What happens in over modulation?
2. What is Amplitude Modulation?
3. Explain different modes used for communication.
4. What are different types of communication system?
5. What is the difference between DSB-SC and SSB-SC signal in terms of total power?
6. Draw the phase’s representation of an amplitude modulated wave?
7. Give the significance of modulation index?
8. What are the two ways of generating DSB_SC?
9. What are the applications of balanced modulator?
10. What are the advantages of suppressing the carrier?
11. What are the advantages of balanced modulator?
12. What are the advantages of Ring modulator?

EXPERIMENT NO:-2
AIM:- Harmonic analysis of Amplitude Modulated wave form.
APPARATUS:-
NAME OF THE
QUANTITY
COMPONENT/EQUIPMENT
FREQUENCY 1
SYNTHESIZER
CONNECTING WIRES AND 1

PROBES
CRO/DSO 1
REGULATED 1
POWERSUPPLY
THEORY:-
Amplitude modulation is defined as the process in which the amplitude of the carrier wave
c(t) is varied about a mean value, linearly with the baseband signal. An AM wave may thus
be dscribed, in the most general form, as a function of time as follows.
S(t)=Ac{1+Kam(t)}cos(2πfct)
Where Ka- Amplitude sensitivity of the modulator
S(t) –Modulated signal
Ac- carrier signal
m(t) –modulating signal
The amplitude of Ka m(t) is always less than unity, that is K a m(t) <1 for all t. It ensure that
the function 1+Ka m(t) is always positive. When the amplitude sensitivity Ka of the
modulator is large enough to make Ka m(t)>1 for any carrier wave becomes over
modulated ,resulting in carrier phase reversal whenever the factor 1+K am(t) crosses zero.
The modulate wave then exhibits envelope distortion.
The absolute maximum value of Ka m(t) multiplied by 100 is referred to as the percentage
modulation.
Vmax-Vmin
Or percentage modulation = -------------- ×100
Vmax +Vmin
The carrier frequency fc is much greater than the highest frequency component ω of the
message signal m(t),that is fc >>W. Where W is the message bandwidth.
If the condition is not satisfied, and envelope cannot be visualized satisfactorily. The trainer
kit has a carrier generator, which can generate the carrier wave of 100 KHz when the trainer
is switched on. The circuit’s carrier generator, modulator and demodulator are provided with
the built in supplies, no supply connections are to be given externally.

AM Fundamentals
Combining (“mixing”) the intelligence and carrier signals can be done:
1. Using linear device (e.g. resistor) –simple addition, but not suitable for
transmission (receiver cannot detect intelligence)
2. Non-linear device (e.g., BJT or OpAmp) –method used in practice
•Non-linear mixing results on:
–DC Component
–Components at original frequencies (intelligence & carrier)
–Components at sum & difference of original frequencies
–Harmonics of original frequencies
Only the following components resulting from non-linear mixing are used on an AM
waveform:
–Carrier frequency (fc)
–Lower-side frequency (fc-fi)
–Upper-side frequency (fc+ fi)
Fig. 2.1 Linear addition of two sine waves
Expected AM Waveforms
An AM modulated signal can be expressed as:

e(t) = (Ec+ Eisin wit) sin wct
where:
Ec= peak value of carrier signal
Ei=peak value of intelligence signal
wc= angular frequency of carrier signal
wi= angular frequency of intelligence signal

It can be demonstrated that:
e(t)= Ec sinwct+ (Ei/2)cos (wc-wi)t -(Ei/2)cos (wc+wi)t
Fig. 2.2: Carrier and side-frequency components result in AM waveform.
PRECAUTIONS:-
RESULT:-
We have observed results of superposition of different frequency components as an

AM waveform.
VIVA –QUESTION:-
1. What you understand from harmonics?
2. What is the role of harmonics in AM waveform?
3. What is Distortion?
4. What is Difference between fundamental and harmonic frequency?
5. Why first harmonic is not ther?
6. How to find the harmonics of any signal?
7. If the fundamental period is 0.001 sec than what are the different harmonics?
8. What is relation between frequency and time period?

EXPERIMENT NO:-3
AIM:- Generation & Demodulation of DSB – SC signal.
APPARATUS:-
NAME OF THE
QUANTITY
COMPONENT/EQUIPMENT
ST-2201 FOR TRANSMITTER 1
1
CONNECTING WIRES AND PROBES 6-7
CRO 1
FUNCTIONGENERATOR 2
REGULATED POWERSUPPLY 1
THEORY:-
In communication systems information is transmitted from one place to another using
electrical signals (telephone, TV and radio broadcast etc.). Usually the information bearing
signals (message signals) are not suitable for transmission due to its propagation qualities (a
large wavelength). Also, since these signals generally exist in the same frequency range it is
necessary to transmit them using different frequency allocations to avoid interference. One
of the methods used to solve these problems is linear modulation, which is merely the
frequency translation of the spectrum of the information (or message) signal to a usually
much higher frequency. The translated spectrum can be modified before transmission in
different forms resulting in different linear modulation schemes. Specifically, there are four
linear modulation methods: double-sideband (DSB) (also known as double-sideband with
suppressed carrier DSB-SC), amplitude modulation (AM) or DSB-LC (large carrier), single-
sideband (SSB) and vestigial-sideband (VSB). This experiment examines the characteristics
of the DSB modulation.
DSB-SC GENERATION:
Fig. 3.1: Block Diagram of DSB-SC generation

A DSB-SC signal can be obtained by simply multiplying modulating signal x(t) with carrier
signal cosωct . So we need to use a device called product modulator for the generation of
DSB-SC waves .
Consider two sinusoids, or cosinusoids, cosμt and cosωt. A double sideband suppressed
carrier signal, or DSBSC, is defined as their product, namely:
DSBSC = E.cosμt . cosωt (1)

Generally, and in the context of this experiment, it is understood that::
ω >> μ (2)
Equation (1) can be expanded to give:
cosμt . cosωt = (E/2) cos(ω - μ)t + (E/2) cos(ω + μ)t

(3)
Equation 3 shows that the product is represented by two new signals, one on the sum
frequency (ω + μ), and one on the difference frequency (ω - μ).

These two components were derived from a ‘carrier’ term on ω rad/s, and a message on μ
rad/s. Because there is no term at carrier frequency in the product signal it is described as a
double sideband suppressed carrier (DSBSC) signal.
The term ‘carrier’ comes from the context of ‘double sideband amplitude modulation'
(commonly abbreviated to just AM). AM is introduced in a later experiment (although,
historically, AM preceded DSBSC).

Fig.3.2: Time Domain representation of DSB-SC signal.
Demodulation method (coherent demodulation)
Fig. 3.3: DSB-SC Modulation and Demodulation block diagram and waveforms
PRECAUTIONS:
RESULT:-
We have observed results of DSB-SC modulation and demodulation also draw the
waveform.

VIVA –QUESTION:-
1. What is Modulation? What happens in over modulation?
2. What is Amplitude Modulation?
3. Explain different modes used for communication.
4. What are different types of communication system?
5. What is the difference between DSB-SC and SSB-SC signal in terms of total power?
6. What are the two ways of generating DSB_SC.
7. What are the applications of balanced modulator?
8. What are the advantages of suppressing the carrier?
9. What are the advantages of balanced modulator?
10. What are the advantages of Ring modulator?
11. Write the expression for the output voltage of a balanced modulator?
12. Give any two methods to avoid errors in synchrouns demodulator?
13. What is quadrature null effect in synchronous demodulator?
14. What is beats in synchronous detector?
15. Give the block diagram of synchronous detector?

EXPERIMENT NO:-4
AIM: -.Modulate a sinusoidal signal with high frequency carrier to obtain FM

signal and demodulation of the FM signal.
APPARATUS:-
NAME OF THE
QUANTITY
COMPONENT/EQUIPMENT
ST-2203 1

PROBES
CRO 1
FUNCTIONGENERATOR 2
REGULATED POWERSUPPLY 1
THEORY:-
Frequency modulation (FM) conveysinformationover a carrier wave by varying it
sinstantaneous frequency. This contrasts with amplitude modulation, in which the
amplitude of the carrier is varied while its frequency remains constant. In analog
applications, the difference between the instantaneous and the base frequency of the carrier
is directly proportional to the instantaneous value of the input-signal amplitude.
If we make the frequency of our carrier wave a function of time, we can get a generalized
function that looks like this:
SFM= ACos(2π[fc+ks(t)]+φ)
We still have a carrier wave, but now we have the value ks(t) that we add to that carrier
wave, to send our data.
Modulation:

FM signals can be generated using either direct or indirect frequency modulation:
 Direct FM modulation can be achieved by directly feeding the message into the input
of a VCO. For indirect FM modulation, the message signal is integrated to
generate a phase modulated signal. This is used to modulate a crystal-controlled
oscillator, and the result is passed through a frequency multiplier to give an FM signal.
Block Diagram of FM Transmitter:
Fig.4.1 FM TRANSMITTER
Demodulation:
Many FM detector circuits exist. A common method for recovering the information signal
is through a Foster-Seeley discriminator. A phase-locked loop can be used as an FM
demodulator. Slope detection demodulates an FM signal by using a tuned circuit which has
its resonantfrequency slightly offset from the carrier. As the frequency rises and falls the
tuned circuit provides a changing amplitude of response, converting FM to AM. AM
receivers may detect some FM transmissions by this means, although it does not provide
an efficient means of detection for FM broadcasts.
Block Diagram FM Receiver:
Observed Waveforms:-
ANALOG AND DIGITAL COMMUNICATION LAB MANUAL
PCE/ECE/IV SEM/ 4EC4-21
PRECAUTIONS:-
1. Check the connections before giving the power supply.
2. Observations should be done carefully.
RESULT:-We have studies about freq. modulation & demodulation & also drawn the waveform
VIVA QUESTIONS:-
1. What is Angle Modulation?
2. What are the advantages of Angle modulation over amplitude modulation?
3. What do you mean by FM and classify FM.
4. What is pre emphasis and de-emphasis?
5. What is the difference between wideband and narrowband FM?
6. What is the use of VCO in Phase lock loop?
7. What is the relationship between PM and FM?
8. With a neat block diagram explain how PM is generated using FM.
9. Define modulation index of FM.
10. Define Modulation index of PM.

EXPERIMENT NO:-5
AIM:- Verification of Sampling Theorem.
APPARATUS USED:-
1. Sampling and Reconstruction Trainer kit, ST-2151 ( Scientech Technologies)
2. A 20 MHz, dual channel Oscilloscope.
THEORY:-
NYQUIST CRITERION (Sampling Theorem):-
The Nyquist criterion states that a continuous signal band limited to f m hz, can be completely
represented by and reconstructed from, the samples taken at a rate greater than or equal to 2 f m
samples per second. This minimum frequency is called as "Nyquist Rate". Thus, for the faithful
reconstruction of the information signal from its samples, it is necessary that the sampling rate, f s
must be greater than 2fm.
Aliasing;-
If the information signal is sampled at a rate lower than that stated by Nyquist criterion, than there
is an overlap between the information signal and the side bands of the harmonics. Thus the lower
and the higher frequency components get mixed and cause unwanted signals to appear at the
demodulator output. This phenomenon is termed as Aliasingor Fold-over Distortion. The various
reasons for aliasing and the ways for its prevention may be summarized as under:-
A) Aliasing due to under-Sampling:-
If the signal is sampled at a rate lower than 2 f m, then it causes aliasing, as illustrated in the
following figure, where a sinusoidal signal of frequency fm is being sampled at a rate fs< 2fm, and
the dots represent the sample points

The LPF at demodulator effectively joins the sample causing an unwanted frequency component to
appear at the output. This unwanted component has a frequency = (fs – fm)
B) Aliasing due to wide band signal:-

The system is designed to take samples at a frequency slightly greater than that stated Nyquist rate.
If higher frequencies are ever present in the information signal, or it is affected by H. F. noise, then
the aliasing will occur. To prevent the aliasing, Anti-aliasing filters are usually installed prior to
sampling. In telephone networks, the speech signals are band-limited by filters before sampling to
avoid the effect of aliasing.
C) Aliasing due to noise:-
If very small duty cycle is used in sample-and-hold circuit, aliasing may occur if the signal has been
affected by the noise. High frequency noise generally mixes with the High frequency component of
the signal and hence causes undesirable frequency components to appear at the output. This type of
aliasing may, therefore, be prevented by slightly increasing the duty cycle of the sampling pulses.
D) Aliasing due to Filter Roll-off:-
Aliasing may also occur, if appropriate filter response is not chosen and the frequencies above the
nominal cut-off frequency of the filter have significant amplitudes at the filter's output. This is
called Aliasing due to Filter Roll-off

BLOCK DIAGRAM:-
Sampling circuit
PROCEDURE:-
1. Set the internal / external sampling selector switch in internal position.
2. Put the duty cycle selector switch in position 5 (to set 50 % duty cycle).
3. Connect 1 KHz internally generated sinusoidal signal available to signal input on the
sampling circuit board.
4. Now, turn the ON/OFF switch of the kit to ON.
5. Observe the information signal (1 KHz at t p12) on one channel and the sample output on the
channel of the CRO.
6. Connect the sample output to the input of 4th order LPF.
7. Trace the sampled output. Note that 32 samples are appearing in one cycle of the information
signal, as the default value of the sampling frequency is 32 KHz.
8. Now, keep on reducing the sampling frequency in steps and trace the sampled output at any
other two values of the sampling frequencies.
9. Observe the reconstructed signal at the output of the 4th order LPF at tp46 on setting the
Sampling frequency equal to 2 KHz. Note that it is distorted as fs = 2 fm
10. Now, connect the sample output to the input of 2 nd order LPF, and observe the reconstructed
signal at the output of the 2nd order LPF at different values of sampling frequencies. Compare

the outputs of both the LPFs for the same value of sampling frequency. Which one is better
and why? This completes the first part of the experiment.
11. For the second part, keep the sampling rate constant at some appropriate value (say 8 or 16
KHz), and vary the position of duty cycle selector switch, and observe the sampled signal
and the reconstructed signal at the output of the 4th order LPF (and also at the output of the
2nd order LPF (tp-42) at different values of the duty cycle varying from 40 % (position 4) to
90 % (position 9 of duty cycle selector switch). Record your observations. How does the
amplitude of the reconstructed signal vary with the variation of the duty cycle?
12. Now, disconnect the sample output (tp-37) from the input of the LPF and connect “sample
and hold output (tp-39) to the inputs of the LPFs, and observe the reconstructed signals at
their outputs one by one.
13. Vary the duty cycle again and observe that the reconstructed signal has now become
independent of duty cycle variation. Measure the amplitude of the reconstructed signal at
and) one by one.
14. Comment on the results obtained by using the 4th order LPF and the 2nd order LPF.

OBSERVATIONS:-
INPUT SIGNAL OUTPUT SIGNAL
SAMPLE AND HOLD CIRCUITII ORDER LPF SAMPLE SIGNAL
IV ORDER LPF SAMPLE SIGNAL
RESULT:-
We have performed& solve the aliasing problem and verify the Nyquist criteria through the
experimental setup.

PRECAUTION:-
1. Connections should be checked before switching ON the kit.
2. Observations should be taken properly.
VIVA QUESTION:-
1. Define Dirac comb or ideal sampling function.
2. State sampling theorem.
3. What is aliasing or fold over?
4. What do you mean by Nyquist Rate?
5. What is Guard Band?
6. What will happen when sampling rate is greater than Nyquist rate ?
7. What will happen when sampling rate is less than Nyquist rate ?
8. What is natural sampling?
9. What is flat top sampling?
10. What is first order hold sampling?

EXPERIMENT NO:-6
AIM:- To study & observe the operation of a super heterodyne receiver.
APPARATUS:-
NAME OF THE
QUANTITY
COMPONENT/EQUIPMENT
ST-2201 FOR 1
TRANSMITTER
1

PROBES
CRO 1
FUNCTIONGENERATOR 2
REGULATED 1
POWERSUPPLY
THEORY:-
The super heterodyne circuit is used in almost all the transistor radios (excepting a few used
for receiving local stations). In this system the frequency of incoming signals are changed to
a fixed frequency known as intermediate frequency IF. The main work of amplifying the
signal and of selecting the desired station, is done in IF section. This permits higher

amplification per stage and better selectivity. The block diagram of Radio receiver is shown
in below. It consists of the following parts / stages.
1. frequency converter (Mixer and Oscillator)

2. I.F. amplifier
3. Detector
4. Audio Preamplifier
5. Driver
6. Output
7. Power supply
FREQUENCY CONVERTER
It has two sections Mixer and Oscillator. Usually only one transistor is used for both
Sections. The oscillator produces oscillations at a frequency which is higher than the
frequency of the station being received. The difference equals the intermediate frequency
(I.F). Out of the signals interrupted by the aerial, the signals from the desired stations are
selected by a tuned circuit at the input of the mixer stage. These signals and the oscillations
from the oscillator are mixed by the mixer. The mixing of these signals results in the
production of their sum (f1+f2) and the difference (f1-f2) frequencies. The difference is IF
signal. It contains the same modulation as contained by the received signals. The IF signals
are given to the IF amplifier.
IF AMPLIFIER
It amplifies the IF signals. It provides a large part of the gain and selectivity. Two stages are
used in this section to obtain the required amplification. Tuned transformers known as IFT
are used as a collector load and for interstate coupling in the IF amplifier. The amplified IF
signals are given to detector.
DETECTOR
The detector separates the audio signal from the modulated carrier. A diode is generally used

for detection.
AUDIO POWER AMPLIFIER
The audio signals received from the detector are given to the audio preamplifier stage. This
stage amplifies these signals and gives them to the driver stage.
DRIVER
The signals obtained from the preamplifier are not strong enough to drive the output stage.
The driver stage further amplifies these signals and gives them to the output stage.
OUTPUT STAGE
The output stage amplifies the audio signals obtained from the driver and develops sufficient
power to drive the loud speaker. To obtain maximum economy in battery consumption, push
pull output stages using two transistors in class B circuit is normally used
POWER SUPPLY
Most of the transistor radios operate on a few cells/battery. The supply voltages from3 to 9V
are common. Some transistor radios can be operated from batteries as well as mains. In these
transistor the mains voltage is converted into D.C. by diode rectifier.
FIG 1: GENERAL BLOCK DIAGRAM OF SUPERHETRODYNE RADIO

RECEIVER

PROCEDURE-
1. Study the radio receiver kit.
2. Draw the block diagram of A.M. super heterodyne receiver.
3. Identify all stages according to the block diagram and note down the RF, IF transistors with
their specifications.
4. Measure the sensitivity (I/P and O/P voltages of frequency converter) and selectivity (I/P
and O/P voltages of IF stage)
5. Measure the test point voltages
6. Observe the test point waveform
PRECAUTIONS:-
1. Check the connections before giving the power supply.
2. Observations should be done carefully.
RESULT:-
We have studied about super heterodyne receiver principle and it is used as AM receiver
circuit.
VIVA QUESTION:-
1. State the advantages of super heterodyning?

2. What is intermediate frequency?
3. What are the different tuned receiver and super heterodyne receiver?
4. What are the objectives met by modulation?
5. What is the use of capacitor tuning?
6. What is image frequency?
7. What is the relation between image and intermediate frequency?
8. What is the range of RF carrier wave and If of AM radio receiver?
9. Which device provide the highest image rejection in AM radio receiver?

EXPERIMENT NO:-7
AIM:- PAM, PWM & PPM: Modulation and demodulation.
1. (A )PULSE AMPLITUDE MODULATION & DEMODULATION
AIM:-To generate a Pulse amplitude modulated signal and also demodulate the original
signal.
APPARATUS:-
NAME OF THE
QUAN
COMPONENT/EQUIPMENT
TITY
MARS ME-746 1

PROBES
CRO 1
FUNCTIONGENERATOR 2
REGULATED 1
POWERSUPPLY
THEORY:-
Pulse-amplitude modulation, acronym PAM, is a form of signal modulation where the
message information is encoded in the amplitude of a series of signal pulses. It is an analog
pulse modulation scheme in which the amplitude of train of carrier pulse are varied
according to the sample value of the message signal.
In pulse Amplitude Modulation the amplitude of the carrier pulses varies in

accordance with the instantaneous values of message signal and the pulse width is fixed, in
practice this is realized by a simple mechanical commentator or by electronic circuit.
Natural PAM signal sampling occurs when finite width is used in the modulators and tops of
the pulses are forced to follow the magnitude of modulating waveform. Flat tapped PAM
system is quite often used because of the generating the modulating waveform and

spectrum. Finite
width pulses are used but they are flat topped after modulation. Reconstruction of original
signal is possible by passing the modulated signal through a low pass filter.
CIRCUIT DIAGRAM:-
Pulse Amplitude ModulationCircuit:
DemodulationCircuit:

PROCEDURE:-
1. Connections are made as per the circuit diagram.

2. Modulating signal is given to a collector terminal of a BC 107 transistors and carrier signal
of High frequency is given to a base terminal of a transistor and switch ON the power
supply.
3. Then the modulated out put is observed across the transistor emitter terminal.
4. The output characteristics of a PAM signal is plotted on a graph.
OUTPUT WAVEFORM:
PRECAUTIONS:-
1. Connections must be tight.
2. The width of the signal must be noted carefully.
RESULT:-
We have observed the results of pulseAmplitude modulation & demodulation; also we have
traced waveforms of modulated and demodulated signal.
(B ) PULSE WIDTH MODULATION & DEMODULATION
AIM:-To construct pulse width modulation circuit and observe modulation and
demodulation.

APPARATUS:-
NAME OF THE
QUAN
COMPONENT/EQUIPMENT
TITY
MARS ME-746 1

PROBES
CRO 1
FUNCTIONGENERATOR 2
REGULATED 1
POWERSUPPLY
THEORY:-
(Pulse Width Modulation):-
A modulation technique that generates variable-width pulses to represent the amplitude of
an analog input signal. Like its fixed-width pulse density modulation (PDM) cousin, the
output switching transistor is on more of the time for a high-amplitude signal and off more
of the time for a low-amplitude signal. The digital nature (fully on or off) of the PWM
circuit is less costly to fabricate than an analog circuit that does not drift over time. See
PDM.
If the widths of the pulses are varying in accordance with the modulating signal it is called
pulse width modulation. In Pulse width modulation, the amplitude of the pulses is constant.
Generation of PWM the input modulating signal is given to non - inverting terminal of op-
amp .the op-amp now compares with both the input signals. The output of the comparator is
high only when instantaneous value of input modulating signal is grater then that of saw
tooth waveform. When saw tooth waveform voltage is grater then input modulating signal at
that instant the output of the comparator remains zero i.e. in negative saturation. Thus output
of comparator is PWM signal.
CIRCUIT DIAGRAM:-

Pulse Width Modulation Circuit
DemodulationCircuit:-

PROCEDURE:-
1. Connection is made as per the circuit diagram.
2. Vary the control voltage (0-5) v and observe the corresponding change in output square
Waveform using CRO.
3. Change in control voltage changes the width of the square wave.
4. Note down the T-ON and T-OFF.
5. Plot the observed waveform.
OUTPUT WAVEFORM:-

PCE/ECE/IVSEM/ 4EC4-21
RECAUTIONS:-
1. Connections must be tight.
2. The width of the signal must be noted carefully.
RESULT:-
We have observed the results of Pulse width modulation and demodulation; also we have
traced waveforms of modulated and demodulated signal
(C ) PULSE POSITION MODULATION & DEMODULATION
AIM:-To generate a Pulse position modulated/demodulated signal by using 555 timers.

APPARATUS:-
NAME OF THE
QUAN
COMPONENT/EQUIPMENT
TITY
MARS ME-746 1

PROBES
CRO 1
FUNCTIONGENERATOR 2
REGULATED 1
POWERSUPPLY
THEORY:-
PPM can be considered version of PDM in PDM, long pulses expend considerable
power during the pulse while bearing no additional information. If an arrangement is made
so that the unused power could be subtracted from the PDM, we get a more efficient pulse
modulation. In PPM the position of a pulse relative to its un-modulated time of occurrence
is varied in accordance with the message signal PPM may be obtained from PWM, in which
the position of PWM pulses are position modulated.

Thus these pulses will have time displacement proportional to the instantaneous
value of the signal voltage. The simplest method of generation of PPM from PWM is to use
a mono-stable multi-vibrator. It is to be designed in such away that it triggers the trailing
edges of a PWM signal. If a PWM signal is applied at the inputs, the output will be
obviously a pulse position modulated signal whose duration will be determined by the
timing circuit of multi-vibrator. For demodulation of PPM, first it converted into PWM with
help of flip-flop, and then it is demodulated.
CIRCUIT DIAGRAM:-
Fig: 1 Pulse Position ModulationCircuit

Fig: 2 DemodulationCircuit
PROCEDURE:-
1. Connections are made as per circuit diagram.
2. The supply +5v are applied to pin4 and pin8.
3. The pins 2&6 are stored for self triggering.
4. The message signal of 500Hz, 2v pp amp- is applied to pin-5.
5. The output is observed on CRO at pin-3.
6. The input and output waveforms are rated from CRO and are plotted on graph sheet.
Timeperiod(ms)
Modulating signal Total Time
Pulse widthOFF
Amplitude(Vp-p) Pulse width ON(ms) period(ms)
(ms)
OUTPUT WAVEFORMS:-

RESULT:-
We have observed the results of Pulse position modulation & demodulation, also we have
traced waveforms of modulated and demodulated signal.
VIVA QUESTIONS:-
1. What are the classifications of pulse modulation techniques?
2. What is the transmission bandwidth of Pulse amplitude modulation?
3. What are the Draw backs in Pulse amplitude modulated signal?
4. Write the standard equation of a PAM in frequency domain?
5. What is meant by Aperture effect?
6. Explain with waveforms how PPM is derived from PWM.
7. What is the fundamental difference between pulse modulation, on the one hand, and
frequency and amplitude modulation on the other?
8. An audio signal consists of frequencies in the range of 100Hz to 5.5KHz.
9. What is the minimum frequency at which it should be sampled in order to transmit it
through
pulse modulation?
10. Draw a TDM signal which is handling three different signals using PWM?
11. What do you infer from the frequency spectrum of a PWM signal?
12. Clock frequency in a PWM system is 2.5 kHz and modulating signal frequency is
500Hzhowmany pulses per cycle of signal occur in PWM output?
13. Draw the PWM signal?
14. Why should the curve for pulse width Vs modulating voltage be linear?
15. What is the other name for PWM?
16. What is the disadvantage of PWM?
17. Will PWM work if the synchronization between Tx and Rx fails?
18. Why integrator is required in demodulation of PWM?
19. What kind of conversion is done in PWM generation?
20. Define PPM?
21. What are the differences between PPM & PWM?
22. Which type of sampling technique is used in PPM?
23. The Multi-vibrator used in PPM is?
24. Pulse duration of PPM is?
25. In PPM the position is proportionally varied in which parameter of the modulating signal?
26. What are the different types of PTM systems?
27. What is the other name of Pulse width modulation?
28. What do you mean by pulse time modulation?
29. What is the comparison between the PAM and PWM?
30. What is the definition of PWM?
31. What is the transmission bandwidth of PWM signal?
32. Which type sampling technique is used in PWM?
33. What are the applications of PWM modulation technique?

EXPERIMENT-8
AIM:- To observe the transmission of four signals over a single channel using
TDM-PAM method.
APPARATUS USED:
TDM Trainer kit, CRO
THEORY:-
Time Division Multiplexing – TDM

Several messages are sampled and their samples are interlaced to form a composite, or Time
Division Multiplexed-TDM signal.
By using quite short carrier pulses (small width), other sampled pulses can be placed in the
gaps between the pulses of PAM (Pulse Amplitude Modulation). This process is called as
Time Division Multiplexing (TDM). Therefore, more than one message signal can be
carried via a single communication channel. The operational fundamentals of TDM will be
examined by using the PAM; however TDM can be applied to other type of pulse
modulations as well.

If two messages are sampled at the same rate but at slightly different times, then two of
trains of samples can be added without mutual interaction. In Figure-1.a the signal x (t) and
the corresponding PAM signal are depicted.
(A): TDM of two signals
If the band-widths of both signals (x1(t) and x2(t)) is 3 kHz, according to the sampling
theorem each signal should be sampled with the frequency of 6 kHz. But in this case the
clock frequency should be 12 kHz. The distance between the pulses is Tn=Ts/n. Here, n
indicates the number of the input signals, Ts denotes the sampling period required for one
signal. The obtained TDM wave is shown below:

(B) :PAM (TDM) Wave
As considered before, if two different signals have the frequency of 3 kHz, the sampling
period of each signal will be Ts=1/6000 = 166.7 µsec. Since number of input signals n=2,
from Tn=Ts/n; the distance between samples becomes Tn=Ts/2 = 83.3 µsec. Therefore the
Minimum bandwidth to transmit these samples by TDM should be; B≥166.7.10−6 =6 kHz. A
TDM receiver block diagram is shown in Figure 4 below. The most significant issue in
recovery of input signals from TDM signals is the requirement for the proper
synchronization between TDM transmitter and receiver. Therefore, the clock signal in the
transmitter should be passed to the receiver correctly.
Block Diagram:-
x1(t) Sampling
+ PAM-TDM
x2 (t) Samplin
g
Pulse
Generator Clock
Pulse Generator
Block diagram of PAM-TDM
PROCEDURE
1. Take the signals from the function generator and give it to the channels (ch0 ... ch3)
present in the transmitter using patch chords. Note down the amplitude and time period
of each signal.
2. Measure the amplitude and time period at the transmitter output point.
3. Using a patch chord, connect transmitter output to receiver input.
4. For synchronization purpose, connect the transmitter clock and receiver clock and also
transmitter ch0 and receiver ch0.
5. See the output before the filter and after the filter for all the channels connected.
OBSERVATIONS
TRANSMITTER SECTION

TDM MULTIPLEXED SIGNAL
RECEIVER SECTION
RESULT:-

We have studied and observe the transmission of four signals over a single channel using
TDM-PAM method.
PRECUATION:-

VIVA QUESTIONS:-
1. Two signals g1 (t) and g2 (t) are to be transmitted over a common channel by means of
time division multiplexing. The highest frequency of g1 (t) is 1 KHz and that g2 (t) is 1.5
KHz. What is the minimum value of the permissible sampling rate? Justify your answer.
2. How is synchronization achieved in TDM?
3. Twenty four voice signals are sampled uniformly and then time division multiplexed.
The sampling operation uses flat top samples with 1μs duration. The multiplexing
operation includes provision for synchronization by adding an extra pulse of sufficient
amplitude and also 1μs duration. The highest frequency component of each voice signal
is 3.4 KHz.
a. Assuming a sampling rate of 8 KHz, Find the spacing between successive pulses
of the multiplexed signal.
b. Repeat your calculation assuming the use of Nyquist rate sampling.
4. What is the major drawback of digital communication?
5. Three signals m1, m2 and m3 are to be multiplexed.m1 and m2 have a 5 KHz bandwidth
and m3 has 10 KHz bandwidth. Design a commutator switching system so that each
signal is sampled at its Nyquist rate.
6. TDM is possible for sampled signals. What kind of multiplexing can be used in
continuous modulation systems?
7. What is the minimum rate at which a speech signal can be sampled for the purpose of
PAM?
8. What is cross talk in the context of time division multiplexing?
9. Which is better, natural sampling or flat topped sampling and why?
10. Why a dc offset has been added to the modulating signal in this board?
11. Was it essential for the working of the modulator? Explain.

12. Study about the frequency spectrum of PAM signal and derive mathematical expression
for it?
EXPERIMENT-9
AIM:- To study the PCM modulation & demodulation and study the effect of
channel like attenuation, noise in between modulator & demodulator through
the experimental setup.
APPARATUS USED:-
PCM Trainer kit, Power chords, 20 MHz Dual trace CRO,Power supply.
THEORY:-
Pulse Code Modulation (PCM) is different from Amplitude Modulation (AM) and
Frequency Modulation (FM) because those two are continuous forms of modulation. PCM
is used to convert analog signals into binary form. In the absence of noise and distortion it is
possible to completely recover a continuous analog modulated signal. But in real time they
suffer from transmission distortion and noise to an appreciable extent. In the PCM system,
groups of pulses or codes are transmitted which represent binary numbers corresponding to

modulating signal voltage levels. Recovery of the transmitter information does not depend
on the height, width, or energy content of the individual pulses, but only on their presence
or absence. Since it is relatively easy to recover pulses under these conditions, even in the
presence of large amounts of noise and distortion, PCM systems tend to be very immune to
interference and noise. Regeneration of the pulse reroute is also relatively easy, resulting in
system that produces excellent result for long distance communication.
PCM ENCODING:
The encoding process generates a binary code number corresponding to modulating signal
voltage level to be transmitted for each sampling interval. Any one of the codes like binary,
ASCII etc., may be used as it provides a sufficient number of different symbols to represent
all of the levels to be transmitted. Ordinary binary number will contain a train of’1’ and ‘0’
pulses with a total of log2N pulses in each number. (N is no of levels in the full range). This
system is very economical to realize because it corresponds exactly to the process of analog
– to – digital (A/D) conversion.
QUANTIZATION:
The 1st step in the PCM system is to quantize the modulating signal. The modulating signal
can assume an infinite no. of different level between the two limit values, which define the
range of the signal. In PCM a coded no is transmitted for each level sampled in the
modulating signal. If the exact no corresponding to the exact voltage were to be transmitted
for every sample, an infinitely large no of different code symbols would be needed.
Quantization has the effect of reducing this infinite no of levels to a relatively small
number, which can be coded without difficulty.
In the quantization process, the total range of the modulating signal is divided into a no of
small sub ranges. The number will depend on the nature of the modulating signal and will
form as few as 8 to as many as 128 levels. A number that is an integer power of two is
usually chosen because of the ease of generating binary codes. The result is stepped
waveform, which follows the counter of the original modulating signal with each step
synchronized to the sampling period.

The quantized staircase waveform is an approximation to the original waveform. The

difference between the two-wave form amounts to “noise” added to the signal by the
quantizing circuit. The mean square quantization noise voltage has a value of
E2np = S2/12 Where S is the voltage of each step. As a result the number of quantization
levels must be kept high in order to keep the quantization noise below some acceptable limit
given by the power signal-to-noise ratio, which is the ratio of average noise power.
DECODING:
The decoding process reshapes the incoming pulses and eliminates most of the transmission
noise. A serial to parallel circuit passes the bits in parallel groups to a digital to analog
converter (D/A) for decoding. Thus decoded signal passes through a sample and hold
amplifier, which maintains the pulse level for the duration of the sampling period,
recreating the staircase waveform approximation of the modulation signal. A low-pass filter
may be used to reduce the quantization noise.
BLOCK DIAGRAM:-
PCM block diagram
PROCEDURE:-

1) Connect the ac power supply to the PCM trainer kit and switch it ON.
2) Observe the analog output from AF signal generator by connecting it to ch-1 of CRO
and adjust its frequency to 1 KHz and amplitude to 0.5Vp-p with the help of
potentiometers p-1 and p-2.
3) Connect the analog output to analog input of sample and hold circuit.
4) Observe the conversion clock output by connecting it to ch-2 of CRO and adjust its
frequency with the help of potentiometer p-4.
5) Observe the sampling frequency by connecting it to ch-2 of CRO and adjust the
frequency to 8 KHz and connect it to the sample and hold circuit.
6) Observe the analog output and sample and hold output by connecting them ch-1 and
ch-2 of CRO.
7) Connect the sample and hold output to ADC sample and hold output and clk output to
ADC clock output.
8) Observe the PCM output and sample output simultaneously on CRO.
9) Observe the output of DAC amplifier by connecting them to ch-1 and ch-2 of CRO.
10) Observe the output of LPF with reference to input analog signal.
11) Calculate the frequency and time delay of output signal with reference to input analog
signal.
OBSERVATION:-
INPUT SIGNAL
PCM MODULATION OUTPUT SIGNAL

PCM DEMODULATION OUTPUT SIGNAL
RESULT:
The PCM modulation and demodulation have been studied and effects of channel have also
been observed.
PRECAUTIONS:
VIVA QUESTIONS
1. What do you mean by quantizing process?

2. What will happen when sampling rate is greater than Nyquistrate?
3. What will happen when sampling rate is less than Nyquistrate?
4. Find the A/D Converter output for input DC voltage of 3.6V.
5. Mention some applications of PCM.
6. What do you mean by quantizing process?
7. Find the A/D Converter output for input DC voltage of 3.6V.
8. A PCM wave in which the amplitude levels of +1 volt and -1 volt are used to represent
binary symbols 1 and 0 respectively. The code word used consists of three bits. Find the
sampled version of an analog signal from which this PCM wave is derived.
9. Mention some applications of PCM.
10. What is the function of Sample and Hold circuit?

EXPERIMENT-10
AIM:- To study the 4 channel PCM multiplexing & de-multiplexing in

telephony system.
APPARATUS USED: -
4-channel TDM-PCM Transmitter &Receiver Trainer kit (ST-2804), CRO, CRO probes.
THEORY:-
Pulse Code Modulation technique involves following steps:
(a) Sampling:
The analog signal is sampled according to the nyquist criteria. Then nyquist criteria states
that for faithful reproduction of a band limited signal, the sampling rate must be at least
twice the highest frequency component present in the signal.So sampling frequency ≥ 2fm,
where fm is maximum frequency component present in the signal.
Practically the sampling frequency is kept slightly more than the required rate.

(b) Allocation of binary codes:
Each binary word defines a particular narrow range of amplitude level. The sampled value
is then approximated to the nearest amplitude level. The sample is then assigned a code
corresponding to the amplitude level, which is then transmitted. This process is called
quantization and it is generally carried out by the A/D Converter as shown below in
OBSERVATION:-
INPUT SIGNAL FOR CH. 1
DISCRETE SIGNAL
DIGITAL
SIGNAL OUTPUT FROM LPF
INPUT SIGNAL FOR CH.2 DISCRETE SIGNAL
OUTPUT FROM LPF DIGITAL SIGNAL

DIGITAL SIGNAL OUTPUT FROM LPF
DIGITAL SIGNAL OUTPUT

RESULT:-
Thus we performed the PCM Modulation and Demodulation.
PRECAUTIONS:-
VIVA QUESTION
1. What does PCM stands for?
2. What is Quantization noise?
3. Whether PCM signal is a digital signal or analog signal?
4. What are the types of PAM?
5. Why is the PAM signal contaminated by Noise?
6. Explain PCM modulation and demodulation using block diagram and waveforms.
7.Explain the input and output waveforms of PCM.
8.Explain analogue to digital conversion process in detail using appropriate sketches.
9. What are the advantages and disadvantages of PCM?
10. What are the practical applications of PCM?

EXPERIMENT NO:-11
AIM:- To study the Delta & Adaptive delta modulation & demodulation and
also study the effect Of channel like attenuation, noise in between modulator &
demodulator through the experimental setup.
APPRATUS USED:-
Delta Modulator Trainer Kit, Digital Storage Oscilloscope, Connecting Probes.
THEORY:-
Delta modulation is a system of digital modulation developed after pulse code modulation.
In this system, at each sampling time, say the Kth sampling time, the difference between the
sample value at sampling time K and the sample value at the previous sampling time (K-1)
is encoded into just a single bit. I.e. at each sampling X time we ask simple question. Has
the signal amplitude increased or decreased since the last sample was taken? If signal
amplitude has increased, then modulator's output is at logic level 1.If the signal amplitude
has decreased, the modulator output is at logic level 0.Thus, the output from the modulator
is a series of zeros and ones to indicate rise and fall of the waveform since the previous
value. One way in which delta modulator and demodulator is assembled is as shown:
Delta Modulation

Delta Demodulation
The Delta Modulator works as follows: The analog signal which is to be encoded into
digital data is applied to the +veinputof the voltage comparator which compares it with the
signal applied to its –veinputfrom the integrator output (more about this signal in forth
coming paragraph).The comparator's output is logic '0' or '1' depending on whether the input
signal at +veterminal is lower or greater than the -ve terminals input signal.The
comparator's output is then latched into a D-flip-flop which is clocked by the transmitter
clock. Thus, the output of D-flip-Flop is a latched 'l' or '0'synchronouswith the transmitter
clock edge.This binary data stream is transmitted to receiver and is also fed to the unipolar
to bipolar converter. This block converts logic '0' to voltage level of + 4V and logic 'l' to
voltage level - 4V.The bipolar output is applied to the integrator whose output is as follows:
a. Rising linear ramp signal when - 4V is applied to it, (corresponding to binary 1)
b. Falling linear ramp signal when + 4V is applied to it (corresponding to binary 0).
The integrator output is then connected to the -ve terminal of voltage comparator, thus
completing the modulator circuit.
Let us understand the working of modulator circuit with the analog input waveform
applied as below:

Technique of Delta Modulation

Suppose at some time-instance t = 0, the integrator output voltage is lower than the analog
input. This causes the voltage comparator voltage to go high i.e. logic '1'. Thisdata is latched
in the D- Flip- Flop at the rising edge of transmitter clock., The latched'1' output of D- flip
is translated to - 4V by the unipolar to bipolar converter block.The integrator then ramps up
to catch analog signal.
At the next clock cycle t = 1, the integrator output becomes more than the analoginput, so a
'0' is latched into D-Flip-Flop. The integrator now ramps downward as +4V voltage signal
from unipolar to bipolar converter appears at its input. Thus, theramp signal again tries to
catch the fallen analog signal.
As we can observe, after several clock cycles the integrator output is approximation ofthe
analog input which tries to catch up the analog input at each sample time. The datastream
from D-flip-flop is the delta modulators output.
The delta demodulator consists of a D-Flip-Flop a unipolar to bipolarconverter followed by
an integrator and a low pass filter. The delta demodulator receives thedata from D-Flip-Flop
of delta modulator. It latches this data at every rising edge ofreceiverclock, which is delayed
by half clock period with respect to transmitter clock.This has been done so that the data
from transmitter may settle down before beinglatched into the receiver Flip-Flop.The
unipolar to bipolar converter changes the output from D-Flip-Flop to either - 4Vor + 4V for
logic '1' and '0' respectively. As it has been seen in case of modulator when the output from
unipolar to bipolarconverter is applied to integrator, its output tries to follow the analog
signal in rampfashion and hence is a good approximation of the signal itself. The
integrator's output contains sharp edges, which are 'smoothened out' by the low - pass filter,
whose cut-offfrequency is just above the audio band.
Delta modulation offers many advantages as listed below:

1. Simple system / circuitry
2. Cheap
3. Single bit encoding allows us to increase the sampling rate or to transmit more
information at some sampling rate for given system bandwidth.
Unfortunately, the practical use of delta modulation is limited due to the following several
drawbacks:
a. Noise
Noise is defined as any unwanted unpredictable random waveform accompanying the
information signal. Whenever the signal is received at thereceived by any communication
medium, it is always accompanied by noise.
b. Distortion
Distortion means that the receiver's output is not the true copy of the analoginput signal at
the transmitter.Distortion in delta modulation occurs due to following causes:
As it has been seen, when the analog signal is greater than the integrator output,
theintegratorramps up to meet the analog signal. The ramping rate of integrator isconstant.
Therefore, if the rate of change of analog input is faster than the rampingrate, the modulator
is unable to catch up with the information signal. This causes a large disparity between the
information signal and its quantized approximation. This error / phenomenon are known as
slope - overloading and causes the loss of rapidlychanging the information.
Information Loss due to Rapid input Changes

At first it may look as though the problem of slope overloading can be solvedincreasing the
ramping rate of the integrator. But as it can be seen from the figure theeffect of the large
step-size is to add large sharp edges at the integrator output andhence it adds to noise
problem faced at receiver. This effect itself leads to distortedreceiver outputIncreasing

sampling rate cannot be the solution to the slope - overloading problem as it determines
how fast the samples are taken and not the ramping rate of theintegrator.
Effect of Sampling Rate on Quantization Noise & Step-Size

Another problem with delta modulation is that it unable to pass DC information. This is not
a serious limitation of speech communication but for the systems like video (picture)
transmission DC level does not provide information about brightness level of the picture for
more details see page no. 40.The above stated limitations of the delta modulation may
Be traded for acceptable price in speech application but is totally unsatisfactory for music or
video signals. Testing instruments needed for experimentation in this workbook
OBSERVATION:
INPUT SIGNAL DELTA MODULATED SIGNAL
BISTABLE
OUTPUT INTEGRATOR OUTPUT

ADAPTIVE DELTA MOD. OUTPUT BISTABLE OUTPUT
INTEGRATOR OUTPUT DEMODULATED OUTPUT
RESULT:-
We have performed of a Delta modulation system & to derive from it a delta sigma
modulation system.
PRECAUTIONS:-
1. Connect the circuit properly.
3. Do not apply stress on the components
VIVA QUESTION:-
1. Define delta modulation
2. Name the types of uniform quantizer.
3. What is the need for speech coding at low bit rates
4. Find the Fourier series expansion of an Impulse train.
5. Mention the applications of DM.
6. What are the advantages of Delta modulator?

7. What are the disadvantages of delta modulator?
8. How to overcome slope overload distortion?
9. How to overcome Granular or ideal noise?
10. What are the differences between PCM & DM?
11. Define about slope over load distortion?
12. What is the other name of Granular noise?
13. What is meant by staircase approximation?
14. What are the disadvantages of Delta modulator?
15. Write the equation for error at present sample?

EXPERIMENT NO:-12
AIM:- To perform the experiment of generation and study the various data
formatting schemes (Unipolar, Bipolar, Manchester, AMI etc.)
APPARATUS USED:-
Coding kit (ST-2156), CRO
THEORY:-
The symbols ‘0’ and ‘1’ in digital systems can be represented in various formats with
different levels & waveforms. The selection of particular format for communication the
system bandwidth, system’s ability to pass DC level information, error checking facility,
ease of clock regeneration & synchronizations at receiver, system complexity & cost etc.
The most widely used formats of data representation are given below. These are also
available on ST2156 trainer. Every data format has specific advantages &disadvantages
associated with them.

Various formats of line coding
Non - Return to Zero (Level) NRZ (L): It is the simplest form of data representation. The
NRZ (L) waveform simply goeslow for one bit time to represent a data '0' & high for one bit
time to represent a data'1'. Thus the signal alternates only when there is a data change. See
figure 2
NRZ (L) Encoding
Clock Regeneration:
Since the level transition takes place at a predetermined moment (e.g. at rising/falling edge
of the data clock), it is possible to extract clock information at the receiver. However the
synchronization & clock information is sparse & sometimes even lost when a long stream of
zero or ones are encountered. The clock regeneration is very difficult in such cases. This
makes the clock regeneration design more complex.
Bandwidth:
The maximum rate at which NRZ (L) waveform can change is half the data clock.This
happens when the data stream consists of an alternate 0's and 1's. As it is known,it is the
maximum signal frequency which determines the bandwidth occupied by theNRZ (L) code.
As you will study other data formats you will appreciate that the NRZ
(L) Waveform requires comparatively narrow bandwidth.

DC Levels: Another problem with NRZ (L) code is that it contains DC Level hence cannot
be used for communication systems which cannot pass DC. E.g. transmission paths
involving transformers AC coupled amplifiers or series capacitors filters etc. This happens
particularly in telephone systems. Let us see now an NRZ (L) code is rendered useless in
such systems. Assume a sequence of repetitive data sent is 0110001 with data 1 level at +
5V & data '0' at 0V.
If the DC Level is lost, the waveform balances at the mean level Mean level = total value of
samples ÷ no of samples.= (0+5+5+0+0+0+5) ÷ 7 = 15 ÷ 7 = 2.14VThus if the DC Level
information is lost, the whole signal balances about 2.14V.Thus the peak value of + 5V will
shift to 5 - 2.14 = 2.86VIt may slip down to a level where the receiver cannot recognize as
level '1' & thus the data could be misread. In extreme case where the input is constant series
of logic 0'sthen the NRZ (L) output would be a constant level. Now if the input changed to a
stream of logic 0's, the output would still be a constant level. The only difference is the DC
Level. Therefore if the DC Level information is lost, we have no way of knowing whether
the original input will have all 0's or all 1's.Recommended testing instruments needed for
experiments in this work book

Clock Recovery:
The receiver can extract the timing information from the NRZ (M) waveform unless there
are long periods of no level changes corresponding to long streams of ‘0's. Here long stream
of 1's are not a problem as in NRZ (L) because now it causes a level change continuously &
the receiver can easily extract the clock information. This is a slight improvement over NRZ
(L) waveform.
Bandwidth A DC Level: The NRZ (M) is similar to NRZ (L) waveform in respect of the
bandwidth utilized &the passing of DC Levels. A considerable advantage of NRZ (M) is
that it is independent of the absolute level of the incoming data. The receiver simply has to
know the level changes. This is an advantage in phase shift keying as will be discusses later
on, where the receiver looks for a change in phase of the incoming signal.
Decoding: The NRZ (M) can be converted to NRZ (L) code by a bit decoder. The bit
decode r samples the incoming data bit, holds it for a moment takes a new sample
&compare the two, to see whether the changes has occurred. If it has occurred it gives
output logic '1' & if not it gives outputs logic '0'. This is the required NRZ (L).
Return to Zero (RZ) Format:

The RZ code provides a partial solution to overcome the receiver clock regeneration
problem with NRZ (L) code. It is similar to NRZ (L) code, except that the information is
contained in the first half of the bit, interval, while the level during the second half of each
period is always 0 volts. The comparison of the two waveforms for a given data is shown in
figure 7. The change may not see much, but it has two main effects as will be discussed in
following paragraphs.
RZ Format
RZ Format

Introduction to Bi phase Codes
The main disadvantage with all the previous formats is their inability to provider eliable
clock synchronized information to the receiver clock. Bi-phase code sover come this
problem by providing the transition in both 0's and ‘1’s.The two most common bi-phase
codes in practice are bi phase (Manchester) &bi-phase (Mark) codes. Also these codes are
independent of the DC Levels i.e. they have zero DC component.
Biphase (Manchester) Coding: The encoding rules for bi-phase (Manchester) code are as
follows. A data '0' is encoded as a low level during first half of the bit time and a high level
during the second half. A data'1' is encoded as a high level during first half of the bit time
and a low level during the second half. Thus string of l's or 0's as well as any mixture of
them will not pass any synchronization problem in receiver.
Biphase (Manchester) Format Bandwidth:

The Biphase (Manchester) code always contains at least one transition per bit time,
irrespective of the data being transmitted. Hence the maximum frequency of the Biphase
(Manchester) code is equal to the data clock rate when a stream of consecutive data '1' & '0'
is transmitted. Therefore the required bandwidth is same asthat of the RZ code & double as
that of the NRZ (L) code.
DC Level: Since the biphase (Manchester) code has a high level for half of each data bit
time &low level for second half irrespective of the data. The effective DC level of the
biphase coded waveform is zero. This allows it to be used in AC coupled communication
systems.

Problem in Decoding:
This form of coding certainly provides plenty of rising edges for clock synchronization but
they do not all occur at same time e.g. we have a rising transitionat the start of code for data
'I' where as for data '0' we have it at the midway of thedata bit time. This causes confusion
in the clock regeneration circuit.To overcome this, we employ a special biphase clock
recovery circuit which can besynchronized by the rising edge occurring at either time.Rest
of the decoding is same as for the RZ code. Since the valid data iscarried for infirst half of
each clock period, we ensure that the regenerated receiver clock's rising edge occurs at this
time.
Biphase (Mark) Format
Bandwidth:
The biphase (Mark) code being very similar to the biphase (Manchester) codingrequires
same amount of bandwidth i.e. double as that of the NRZ (L) code. This isthe required
bandwidth in worst case when a series of consecutive 'l's occur.
DC Level:

Similar, to the biphase (Manchester) code, this code too is independent of the DCLevel.
This is so because for every bit time the level is high for a half period &lowfor other half.
This feature makes it useful in applications & systems supporting onlythe AC coupling.
Receiver Clock Regeneration:

This coding provides plenty of rising edges for clock synchronization. But as with biphase
(Manchester) these do not all occur at the same time e.g. in one of the rising edge may occur
at the start of the bit interval where as in other bit it may occur midway between the bit
interval. Again to overcome this problem biphase clock recovery circuit is used which can
be synchronized by rising edge occurring at either time, as we have studied earlier. Rest of
decoding is same as that for biphase (Manchester). But since the output gives the
information other than the actual levels, it needs to be further processed in bit decoder as
has been studied in NRZ (M) decoding. In this circuit each sample is compared with the
previous one. If they are same, logic '0' is produced if different, alogic '1'.The advantage of
this system over biphase (Manchester) is that it is suitable for the systems where it is easier
to spot the changes in sequence.
Decoding: A simple comparator decodes the RB signal into RZ waveform. The comparator
output is logic 1(+5V), if the signal is greater than the bias voltage (threshold Voltage)
applied to the other input and logic 0 (0V) output .
Signal Decoding
The clock can be simply extracted by a circuit that gives a positive output when RBsignal is
not at 0volts (polarity doesn't matter) & provides a zero voltage whenever theRB waveform
is at bias level.

PROCEDURE:
1. Connect the data generator output to code generator kit. This gives the random binary
sequence o the kit.
2. Connect the clock signal to the trainer kit.
3. Connect the output to the CRO channel along with the clock signal.
4. Observe the waveforms with respect to clock on a dual channel CRO, and compare with the
model graph.
OBSERVATION:-
Waveforms for different codes

RESULT:-
We have performed of the various data formatting schemes (Unipolar, Bi-polar,
Manchester, AMI etc.).
PRECAUTIONS:-
VIVA QUESTION:-
1. Define modulation rate.
2. State NRZ unipolar format.
3. State Manchester format.

4. What is Inter symbol interference?
5. Give the advantages of Manchester encoding.
6. What is linear predictor? On what basis are predictor coefficients are determined
7. Identify the need of prediction filtering.
8. Summarize the need of Line
9. List the 2 properties of linear prediction.
10. Express the data 10011 using the Manchester code format.
11. Construct unipolar and RZ code for the binary data 01101001.
12. . Compare different source coding methods with respect to various parameters of a
modulations.
13. . For the following bit sequence 1101010011 draw the waveforms for RZ unipolar,
NRZ polar, AMI, Manchester, RZ polar and NRZ line coding techniques.

EXPERIMENT NO:-13(A)
AIM:- To perform the experiment of generation and detection of

ASK signal with variable length data pattern.
APPARATUS REQUIRED:-
ASK Trainer kit, Dual Trace oscilloscope
THEORY:-
Amplitude Shift Keying (ASK):
The simplest method of modulating a carrier with a data stream is to change the amplitude
of the carrier wave every time the data changes. This modulation technique is known
amplitude shift keying. The simplest way of achieving amplitude shift keying is by
switching ‘On’ the carrier whenever the data bit is '1' & switching off. Whenever the data
bit is '0' i.e. the transmitter outputs the carrier for a' 1 ' & totally suppresses the carrier for a
'0'. This technique is known as ‘On-Off’ keying figure 20 illustrates the amplitude shift
keying for the given data stream.
Thus, Data = 1 carrier transmitted, Data = 0 carrier suppressed
The ASK waveform is generated by a balanced modulator circuit, also known as linear
multiplier. As the name suggests, the device multiplies the instantaneous signal at its two
inputs. The output voltage being productof the two input voltages at any instance of time.
One of the inputs is AC coupled 'carrier' wave of high frequency. Generally; the carrier
wave is a sine wave since any other waveform would increase the bandwidth, without
providing any advantages. The other input which is the information signal to be transmitted,
is DC coupled. It is known as modulating signal.
The data stream applied is unipolar i.e. 0 volts at logic '0' & + 5 Volts at logic '1'. The
output of balanced modulator is a sine wave, unchanged in phase when a data bit ‘l' is
applied to it. In this case the carrier is multiplied with a positive constant voltage when the
data bit '0' is applied, the carrier is multiplied by 0 volts, giving rise to 0 volt signal at
modulator's output. The ASK modulation result in a great simplicity at the receiver. The
method to demodulate the ASK modulation results in a great simplicity at the receiver. The
method to demodulate the ASK waveform is to rectify it, pass it through the filter &'Square

Up' the resulting waveform. The output is the original data stream. Figure 22shows the
functional blocks required in order to demodulate the ASK waveform atreceiver.
BLOCK DIAGRAM
ASK Modulator
ASK Demodulator
PROCEDURE:-
1. The ASK waveform is rectified by a diode rectifier, giving a positive going signal. This
signal is too rounded to be used as digital data. Also the carrier component is still present &
it is of unreliable amplitude due to the attenuation &noise in transmission path. In fact it is a
great drawback associated with ASK modulation. The data level may be misinterpreted by
the receiver if the amplitude change is too much.
2. After rectification, the signal is passed through the low pass filter to remove the carrier
component. This result in slightly rounded pulses of unreliable amplitude.
3. These rounded pulsed are then 'Squared Up' (i.e. shaped in a square wave fashion) by
passing it through voltage comparator set at a threshold level. If the input voltage exceeds
The threshold level, the comparator output is a +5V signal and into her case it is O\Z Thus
at the end we have the true copy of the original input
OBSERVATION:-

ASK Modulator output
ASK Demodulator output
RESULT: -
We have perform practical of a ASK Modulation & Demodulation
PRECAUTIONS:-

VIVA QUESTION:-
1. Application of ASK modulation & Demodulation
2. What is the Difference between ASK & PSK.
3. What is the meaning of grid on?
4. What is role of carrier signal in ASK modulation
5. Write down the code for generation for carrier signal.
6. Compare B-W, Power, Probability of error and S/N of ASK, FSK and PSK?
7. Which modulation technique is known as ON-OFF keying. Justify your answer?
8. Give the differences between bit rate and baud rate?

EXPERIMENT NO:-13(B)
AIM:- Generate the FSK wave forms & demodulate the FSK signals.
APPRATUS USED:-
ASK Trainer kit, Dual Trace oscilloscope
Theory:-
Frequency Shift Keying: In frequency shift keying, the carrier frequency is shifted in steps
(i.e. from one Frequency to another) corresponding to the digital modulation signal. If the
higher frequency is used to represent a data '1' & lower frequency a data '0', the resulting
Frequency shift keying waveform appears as shown in figure 25.Thus Data = 1 high
frequency Data = 0 low frequency
BLOCK DIAGRAM
FSK Modulator

FSK Demodulator
PROCEDURE:-
1. Connect the power supply to kit
2. Set the Data for modulation through DIP switch
3. Trace the modulation Output & Demodulation output
OBSERVATION:-
FSK Modulator Output
RESULT: -
We have perform of a FSK Modulation & Demodulation

PRECAUTIONS:-
VIVA QUESTION:-
1. What is FSK?
2. What are the Mark & Space Frequencies?
3. Differentiate ASK & FSK
4. What is the Application of FSK?
5. What is the Disadvantage of FSK?
6. Compare ASK and PSK.
7. List the advantages and disadvantages of ASK.
8. List the Applications of ASK
9. What are the advantages and disadvantages of PSK?
10.

EXPERIMENT NO.14
AIM:- To generate and demodulate quadrature phase shifted (QPSK) signal
using MATLAB
APPRATUS USED:-
MATLAB R2010a
THEORY:-
Generation of Quadrature phase shift keyed (QPSK) signal
QPSK is also known as quaternary PSK, quadrature phase PSK, 4-PSK, or 4-QAM. It
is a phase modulation technique that transmits two bits in four modulation states.
Phase of the carrier takes on one of four equally spaced values such as π/4, 3π/4, 5π/4
and7π/4.
Si (t) = √2E/T cos {2 πƒct + (2i – 1) π/4}, 0≤ t ≤T
0, elsewhere
Where i = 1,2,3,4, & E= Tx signal energy per symbol
T= symbol duration
Each of the possible value of phase corresponds to a pair of bits called dibits.
Thus the gray encoded set of dibits: 10,00,01,11
Si (t) = √2E/Tcos [(2i – 1) π/4] cos (2πfct) - √2E/Tsin [(2i –1) π/4)] sin (2πfct) ,0≤ t
≤Tb
0, else where
There are two orthonormal basis functions
c1 (t) = √2/T cos 2πƒct, 0≤ t ≤Tb
c2 (t) = √2/T sin 2πƒct, 0≤ t ≤Tb
Algorithm
Initialization commands
QPSK modulation
1. Generate quadrature carriers.
2. Start FOR loop
3. Generate binary data, message signal(bipolar form)

4. Multiply carrier 1 with odd bits of message signal and carrier 2 with even bits of
message signal
5. Perform addition of odd and even modulated signals to get the QPSK modulated signal
6. Plot QPSK modulated signal.
7. End FOR loop.
8. Plot the binary data and carriers.
QPSK demodulation
1. Start FOR loop
2. Perform correlation of QPSK modulated signal with quadrature carriers to get two
decision variables x1 and x2.
3. Make decision on x1 and x2 and multiplex to get demodulated binary data.
If x1>0and x2>0, choose ‘11’. If x1>0and x2<0, choose ‘10’. If x1<0and x2>0,
choose ‘01. If x1<0and x2<0, choose ‘00’.
End FOR loop
Plot demodulated data
Program
% QPSK Modulation
clc;
Clear all;
Close all;
%GENERATE QUADRATURE CARRIER SIGNAL
Tb=1; t=0 :( Tb/100):Tb;fc=1;
c1=sqrt(2/Tb)*cos(2*pi*fc*t);
c2=sqrt (2/Tb)*sin (2*pi*fc*t);
%generate message signal
N=8; m=rand (1, N);
t1=0; t2=Tb
For i=1:2 :( N-1)
t= [t1 :( Tb/100):t2]
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
%odd bits modulated signal
odd_sig(i,:)=c1.*m_s;
if m(i+1)>0.5
m(i+1)=1;
m_s=ones(1,length(t));
else
m(i+1)=0;
m_s=-1*ones(1,length(t));
end
%even bits modulated signal
even_sig(i,:)=c2.*m_s;
%qpsk signal
qpsk=odd_sig+even_sig;
%Plot the QPSK modulated signal
subplot(3,2,4);plot(t,qpsk(i,:));
title('QPSK signal');xlabel('t---->');ylabel('s(t)');grid on; hold on;
t1=t1+(Tb+.01); t2=t2+(Tb+.01);
end
hold off
%Plot the binary data bits and carrier signal
subplot(3,2,1);stem(m);
title('binary data bits');xlabel('n---->');ylabel('b(n)');grid on;
subplot(3,2,2);plot(t,c1);
title('carrier signal-1');xlabel('t---->');ylabel('c1(t)');grid on;
subplot(3,2,3);plot(t,c2);
title('carrier signal-2');xlabel('t---->');ylabel('c2(t)');grid on;

QPSK Demodulation
t1=0;t2=Tb
for i=1:N-1
t=[t1:(Tb/100):t2]
%correlator
x1=sum(c1.*qpsk(i,:));
x2=sum(c2.*qpsk(i,:));
%decision device
if (x1>0&&x2>0)
demod(i)=1;
demod(i+1)=1;
elseif (x1>0&&x2<0)
demod(i)=1;
demod(i+1)=0;
elseif (x1<0&&x2<0)
demod(i)=0;
demod(i+1)=0;
elseif (x1<0&&x2>0)
demod(i)=0;
demod(i+1)=1;
end
t1=t1+(Tb+.01); t2=t2+(Tb+.01);
end
subplot(3,2,5);stem(demod);
title('qpsk demodulated bits');xlabel('n---->');ylabel('b(n)');grid on;

Modal Graphs
PROCEDURE:-
1. Connect the power supply to kit.
2. Set the Data for modulation through DIP switch.
3. Trace the modulation Output & Demodulation output.
RESULT:-
The program for QPSK modulation and demodulation has been simulated in MATLAB and
necessary graphs are plotted.
PRECAUTIONS:-
VIVA QUESTION:-
1. What about the QPSK?
2. What are Advantages of QPSK?
3. What are Disadvantages of QPSK?
4. What is Application of QPSK?
5. What is the difference between QPSK & BPSK?

6. Write a matlab program to sample a message signal m(t) and reconstruct it.
7. Identify the error in the mat lab command Sin 3.
8. Draw the constellation diagram of QPSK.
9. Find the output of the following command. 5^ (2/3) – 25/(2*3)
10. What is the relationship between 4 QAM and QPSK?
11. Design a SIMULINK model for QPSK.

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TABLE OF CONTENTS

S.No. Contents Page No.

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING Page | 1

VISION & MISSION OF THE INSTITUTION

VISION & MISSION OF DEPARTMENT

PROGRAMME EDUCATIONAL OBJECTIVES

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING Page | 2

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING Page | 3

Sr. No. Contents

3 Generation & Demodulation of DSB – SC signal.

EXPERIMENT BEYOND CURRICULUM

Sr. No. Contents

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING Page | 4

PO-9 (Individual and

PO-5 (Modern Tool )

PO-6 (Engineer and

CO-PSO Mapping Matrix of course

Sr.No. Name of Lab/Subject Code

1 Signal Processing Lab 3EC4-23

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Number of turns required for

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EXP 1 EXP 2 EXP 3 EXP 4 EXP 5 EXP 6

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BEFORE ENTERING IN THE LAB

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DEPARTMENT OF ELECTRONICS & COMMUNICATION

3) LAB SCHEDULE PER WEEK: 2 Hours

4) ANY OTHER INFORMATION

PLACE: PCE, JAIPUR DATE:

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CONNECTING WIRES AND 6-7

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BLOCK DIAGRAM OF AM TRANSMITTER:

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7. Find the value of R from fm=1/(2*Pi*R*C) , C=0.1µF

Table 1:fm= 1KHz, fc=11KHz, Ac=15 V p-p.

S.No. Vm(Volts) Emax(volts) Emin(Volts) m %m (mx100)

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S.No. fm(KHz) Emax(volts) Emin(Volts) m %m (mx100)

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CONNECTING WIRES AND 1

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Combining (“mixing”) the intelligence and carrier signals can be done:

Fig. 2.1 Linear addition of two sine waves

An AM modulated signal can be expressed as:

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e(t)= Ec sinwct+ (Ei/2)cos (wc-wi)t -(Ei/2)cos (wc+wi)t

Fig. 2.2: Carrier and side-frequency components result in AM waveform.

We have observed results of superposition of different frequency components as an

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Fig. 3.1: Block Diagram of DSB-SC generation

7. Find the value of R from fm=1/(2PiR*C) , C=0.1µF