lab mannual dcom

7/29/2019 Lab Mannual dcom

1/32

DEPARTMENT OF ELECTRONICS AND COMMUNICATIONENGINEERING, YCCE

DIGITAL COMMUNICATION LAB MANNUAL

1

Lab Manual

Faculty Name Designation Subject/

Subject code

Semester/

Branch

Lab/

Week

Dipika Sagne Assistant

Professor

DCOMM 6th

Sem(DCOM)

1 (2 Hours)

Purpose of the laboratory

The main goal of this laboratory is to give you a practical idea about different modulation and

coding schemes important for digital communication.


2/32



2

EXPERIMENT: - 1

OBJECTIVE:

To perform sampling and reconstruction of a signal and observe itswaveform.

EQUIPMENTS REQUIRED:

1. ST2101 with power supply cord

2. Oscilloscope with connecting probe

3. Connecting cords

THEORY:In analog communication systems like AM, FM the instantaneous value of the

Information signal is used to change certain parameter of the carrier signal. Pulse modulationsystems differ from these systems in a way that they transmit a limited no. of discrete states of a

signal at a predetermined time; sampling can be defined as measuring the value of aninformation signal at predetermined time intervals. The rate of which the signal is sampled is

known as the sampling rate or sampling frequency. It is the major parameter, which decides thequality of the reproduced signal. If the signal is sampled quite frequently (whose limit is

specified by Nyquist Criterion) then it can be reproduced exactly at the receiver with nodistortion.

Nyquist Criterion: The lowest sampling frequency that can be used without the side bands

overlapping is twice the highest frequency component present in the information signal. If wereduce this sampling frequency even further, the side bands and the information signal will

overlap and we cannot recover the information signal simply by low pass filtering. Thisphenomenon is known as fold-over distortion or aliasing.


3/32



3

Nyquist Criterion (Sampling Theorem):The Nyquist Criterion states that a continuous signal band limited to fm Hz can be completely

represented by and reconstructed from the samples taken at a rate greater than or equal to 2fmsamples/second. This minimum sampling frequency is known as NYQUIST RATE i.e. for

faithful reproduction of information signal fs > 2fm.

Effect of Sample and Sample Hold Output: If the pulse width of the carrier pulse train used in

natural sampling is made very short compared to the pulse period, the natural PAM is referred toas instantaneous PAM. As it has been discussed, shorter pulse is desirous for allowing manysignals to be included in TDM format but the pulse can be highly corrupted by noise due to

lesser signal power. One way to maintain reasonable pulse energy is to hold the sample valueuntil the next sample is taken. This

Technique is formed as sample value until the next sample-and-hold techniques. Now, the areaunder the curve (which is equivalent to the signal power) is greater and so the filter output

amplitude and quality ofreproduced signal is improved. The hold facility can be provided by acapacitor when the switch connects the capacitor to PAM output it charges to the instantaneous

value.

Aliasing: If the signal is sampled at a rate lower than stated by Nyquist criterion, then there is anoverlap between the information signal and the sidebands of the harmonics. Thus the higher and

the lower frequency components get mixed and cause unwanted signals to appear at thedemodulator output. This phenomenon is turned as aliasing or fold over distortion. The various

reasons for aliasing and its prevention are as described.

A) Aliasing due to Under-Sampling

If the signal is sampled at rate lower than 2fm then it causes aliasing. Let us assume a sinusoidalwaveform of frequency fin which is being sampled at rate fs


4/32



4

noise pick-up can be present. In this case band-limiting filters, generally known as anti-aliasingfilters are usually installed prior to sampling to prevent aliasing. As a principle, the system is

designed to sample at rate higher than the rate to take into account the equipment tolerances,ageing and filter response.

C) Aliasing due to Filter Roll-offRoll-off is a term applied to the cut-off gradient of a filter. No filter is ideal and therefore

frequencies above the nominal cut-off frequency may still have significant amplitudes at afilters o/p. If proper sampling rate and appropriate filter response is not chosen, aliasing will

occur.

D) Aliasing due to Noise

If very small duty cycle is used in sample-and-hold circuit aliasing may occur if the signal hasbeen affected by noise. High frequency noises generally mix with the high frequency

component of the signal and hence causes undesirable frequency components to be present at theoutput.

Low pass filters

The PAM system the message is recovered by a low pass filter. The type of filter used is veryimportant, as the signal above the cut-off frequency would affect the recovered signal if they are

not attenuated sufficiently.

Signal Reconstruction Connection Diagram:


5/32



5

Signal Reconstruction Connection Diagram:


6/32



6


7/32



7

PROCEDURE: -1. Assemble all the required components to perform practical.2. Connect a BNC to crocodile cable to CRO & m (t) (with in kit) to observe m (t). Note down

amplitude & its frequency.3. Now select a sampling frequency from sampling freq. selection circuit & observe it on CRO

for amplitude.4. Give both sampling clock & message signal as input to sampled output circuit, while keeping

trigger at internal position switch.5. Observe sampled o/p on CRO for amplitude & freq.

6. Connect a cable between sampled outputs to 4th filters input (LPF) for reconstructing signal.7. Observe reconstructed signal output.

8. Repeat above procedure for further sampling frequency.

OBSERVATION TABLES:

Entity Amplitude Frequency

Pulse

Sample O/P


Pulse

Sample O/P


PulseSample O/P


8/32



8

CONCLUSION:

QUESTIONS:

1. What is the use of sampling theorem?

2. What is the world wide standard sampling rate for speech signal?

3. What do you mean by over sampling?

4. What is aliasing effect in sampled signal?

5. What do you mean by natural sampling?


9/32



9

EXPERIMENT: - 2

OBJECTIVE:

To generate a PCM signal and demodulate it.EQUIPMENTS REQUIRED:

TDM Pulse Code Receiver Trainer (ST2104) TDM Pulse Code, Transmitter Trainer (ST2103),Patch cords, CRO etc.,

THEORY:Pulse Code Modulation (PCM): In PCM System the amplitude of the sampled waveform at

definite time intervals is represented as a binary code. The first three techniques of the above

described systems are not truly digital but in fact are analog in nature. The very fact that thevariation of a particular pulse parameter is continuous rather than being in the discrete stepsmakes the system analog in nature. As a result of this, the PAM signals are vulnerable to noise &

dispersion of the pulse. The channel introduces noise on the signal from various sources. Alsothe receiver is not noise free. The pulses also suffer attenuation & dispersion as they pass

through the channel. The primary line constants (L, C, G, & R) limit the velocity at which aparticular frequency can travel. The result is different frequency travel at different velocities in

the medium. Therefore some frequency component of the square wave arrives later as comparedto the other. This causes widening of the pulse width. The phenomenon is called 'dispersion.

The combined effect of attenuation, dispersion & noise is so large that the pulse is impaired &introduced at the receiver.

Steps in Pulse Code Modulation:

Sampling:The analog signal is sampled according to the Nyquist criteria. The Nyquist criteria states that for

faithful reproduction of the band limited signal, the sampling rate must be at least twice thehighest frequency component present in the signal. For audio signals the highest frequency

component is 3.4 KHz.

So,

Practically, the sampling frequency is kept slightly more than the required rate. In telephony the

standard sampling rate is 8 KHz. Sample quantifies the instantaneous value of the analog signalpoint at sampling point to obtain pulse amplitude output.

Allocation of Binary Codes :

Each binary word defines a particular narrow range of amplitude level. The sampled value isthen 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 as


10/32



10

Quantization & it is generally carried out by the A/D converter. There are two importantproblems associated with quantization.

a. Quantization noise :As we have seen the signal is approximated to the nearest level (step). Since the levels are

discrete where as the signal is continuous, the discrepancy creeps in. The difference between theanalog signal value & its approximated one (quantized one) is random & unpredictable. This is a

sort of unwanted, unpredictable, random signal which accompanies the information signal and istermed as 'Quantization noise'. Quantization noise can be reduced by increasing the number of

levels, hence reducing the approximation. But it can never be eliminated. Increasing the numberof levels to reduce quantization noise has the effect of increasing the number of bits. But nothing

comes without price. Increasing the number of bits to represent a sample increases the system'sbandwidth requirement.

b. Finite sampling time of A/D converter :Another problem associated with quantization is that the A/D Converter requires finite time to

convert the analog information to digital data. The A/D Converter requires that the value at itsinput, remain unchanged till the conversion is complete. But in practice, the duration of sampled

pulse is much smaller than the A/D converter's sampling time. This problem can be overcomeby using a sample & hold circuit prior to A/D converter output. The sample & hold circuitry

holds the sample value till the next sample. The encoding method described above is calledas uniform encodingi.e. the quantization levels are uniform for all the amplitude range. But this

method of encoding has disadvantages of its own. The quantization noise plays havoc with thelow level signals because the % approximation compared to the signal amplitude is very high.

This causes a great amount of distortion at the receiver for low level signals. Also the quieter partof music or speech could become severely distorted & would make them unpleasant to listen. To

overcome this problem, a non-uniform encoding scheme is used. Here the quantization levels areclear together for low level than they are for the high levels. This has an effect of compression on

the extreme ends of the signal. The input/output characteristics for compression signal passedthrough a comparator network 'prior to compression (See figure 1). This process is called

compression.

The opposite effect is utilized at the receiver to undo the effect of compression, is termed asexpanding. The two processes are combined are known as compounding this feature is not


11/32



11

provided on trainer but you should be aware of its existence. Some error correcting codes &synchronization can also be transmitted along with the information signal. At receiver, the data is

decoded by the D/A converter; the recovered samples are filtered & reconstructed to provide theoriginal waveform. Various channels can be multiplexed in time domain i.e. the information data

from various sources are sequentially transmitted over the same transmission medium e.g Let usassume a 3 channel PCM system. The system samples 0-2 samples sequentially providing 3

samples to be converted to 3 "n" bit words. These three n bit words forms the basis of a frame.The frame contains these three n bit words also contains some synchronization & reference

positioning information.

Figure: A single channel PCM Transmission System


12/32



12

Figure:Message and Quantized Signal

PROCEDURE: -1. Make the connection according to the circuit diagram.2. Connect the audio frequency of 1 KHz, 2V signal to analog to digital converter.

3. Mode switches in fast position.4. Pseudo - random sync code generator switched 'Off'.

5. Error check code selector switches A & B in A = 0 & B= 0 position ('Off' Mode).6. Connect the PCM modulator output to CRO.

7. Observe output on CRO.


13/32



13


14/32



14

OBSERVATION TABLES:

M(t) Pulse Reconstructed M(t)

Amplitude:

Frequency:

Amplitude Amplitude

Frequency Frequency

Amplitude Amplitude

Frequency Frequency

Amplitude Amplitude

Frequency Frequency

Amplitude Amplitude

Frequency Frequency

CONCLUSION:

QUESTIONS:1. Which noise is occurs in PCM?

2. What is Quantization?3. What is the advantage of PCM?

4. At which factor bandwidth of PCM depends?


15/32



15

EXPERIMENT: - 3

OBJECTIVE:

Study and perform Delta Modulation and Demodulation.EQUIPMENTS REQUIRED:

Delta Adaptive and Delta Sigma Modulation-Demodulation Trainer (ST2105), CRO,patch cords

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 encodedinto just a single bit. I.e. at each sampling time we ask simple question. Has the signal amplitudeincreased 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 outputis at logic level 0. Thus, the output from the modulator is a series of zeros and ones to indicaterise and fall of the waveform since the previous value. One way in which delta modulator and

demodulator is assembled.

Delta Modulator: The analog signal which is to be encoded into digital data is applied to thepositive input of the voltage comparator which compares it with the signal applied to its negative

input from the integrator output (more about this signal in forth coming paragraph). Thecomparator's output is logic '0' or '1' depending on whether the input signal at positive terminal is

lower or greater then the negative terminals input signal. The comparator's output is then latchedinto a D-flip-flop which is clocked by the transmitter clock. Thus, the output of D-flip-Flop is a

latched 'l' or '0' synchronous with the transmitter clock edge. This binary data stream istransmitted to receiver and is also fed to the unipolar to bipolar converter. This block converts


16/32



16

logic '0' to voltage level of + 4V and logic 'l' to voltage level - 4V. The Bipolar output is appliedto 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 negative terminal of voltage comparator, thuscompleting the modulator circuit. Let us understand the working of modulator circuit with the

analog input waveform applied as below:

Figure: Technique of Delta Modulation

Block Diagram:


17/32



17

PROCEDURE: -

1. Connect the mains supply


18/32



18

2. Make connection on the board as shown in the figure3. Ensure that the clock frequency selector block switches A & B are in A = 0 and B = 0

position.4. Ensure that integrator 1 block's switches are in following position:

a) Gain control switch in left-hand position (towards switch A & B).b) Switches A & B in A=0 and B=0 positions.

5. Ensure that the switches in integrator 2 blocks are in following position:a) Gain control switch in left-hand position (towards switch A & B)b) Switches A & B are in A = 0 and B = 0 positions.

6. Connect the DM modulator output to CRO.7. Connect the DM modulator output to receiver side & observe the output on CRO.

OBSERVATION TABLES:


Transmitter Clk

Data I/P

Integrator O/P

Data O/P

CONCLUSION:

QUESTIONS:1. How analog signal can be encoded in to bits?

2. What is the advantage of DM over PCM?

3. Which types of noise occur in delta modulation?

4. Define adaptive delta modulation.


19/32



19

EXPERIMENT: - 4

OBJECTIVE:

To study different types of digital data formats (RZ, NRZ and Manchester).EQUIPMENTS REQUIRED:

8 bit Variable Binary Data Generator (ST2111), Data

Formatting And Carrier Modulation Transmitter Trainer (ST2106), CRO, patch cords

THEORY:Line Coding Basics: Transmission of serial data over any distance, be it a twisted pair, fiber

optic link, coaxial cable, etc., requires maintenance of the data as it is transmitted throughrepeaters, echo chancellors and other electronically equipment. The data integrity must be

maintained through data reconstruction, with proper timing, and retransmitted. Line codes were

created to facilitate this maintenance. In selecting a particular line coding scheme someconsiderations must be made, as not all line codes adequately provide the all importantsynchronization between transmitter and receiver. Other considerations for line code selection

are noise and interference levels, error detection and error checking, implementationrequirements, and the available bandwidth.

Unipolar Coding: The most basic transmission code is unipolar or unbalanced coding. In thisscheme each discrete variable is transmitted with a different assigned level, 0V and for example

+2.5V. But this holds a number of disadvantages: The average power is two times other bipolar codes The coded signal contains DC and low frequency components. When long strings of zeros are present, a DC or baseline wander occurs.

This results in loss of timing and data because a receiver/repeater cannotoptimally discriminate ones and zeros.

Repeaters/receivers require a minimum pulse density for proper timing extraction.Long strings of ones or zeros contain no timing information and lead to timing

jitter (when a clock recovery is used) and possible loss of synchronization. There is no provision for line error rate monitoring.

Bipolar Coding: With bipolar, or also called balanced coding, the same data may be transmittedmore efficiently achieving the same error distance with half the power. This coding is often

referred to as Non-Return to Zero (NRZ) coding as the signal level is maintained for the durationof the signal interval. Although bipolar coding is more efficient than unipolar, it still lacks

provisions for line error monitoring and is susceptible to DC wander and timing jitter. Thiscoding scheme provides a number of features which:

Eliminate DC Wander Minimize Timing Jitter Provide for Line Error Monitoring. This is accomplished by introducing controlled

redundancy in the code through extra coding levels.

Data Formatting: -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 depends on the


20/32



20

system bandwidth, systems ability to pass DC level information, error checking facility, ease ofclock regeneration & synchronizations at receiver, system complexity & cost etc. The most

widely used formats of data representation are given below. These are also available on ST2106trainer. Every data format has specific

advantages & disadvantages associated with them.

Non - Return To Zero (Level) NRZ (L) :It is the simplest form of data representation. The NRZ (L) waveform simply goes

low 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.

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, exceptthat the information is contained in the first half of the bit, interval, while the level during thesecond half of each period is always 0 volts. The comparison of the two waveforms for a given

data is shown in figure.


21/32



21

Biphase (Manchester) Coding:The encoding rules for biphase (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 encodedas 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.Figure shows the biphase (Manchester) waveform for a given data stream.

BLOCK DIAGRAM:


22/32



22

PROCEDURE: -

1. Generate a clock signal having amplitude 5vp-p & freq. 240 kHz.2. Using a kit, generate data signal.3. Now pass the data signal & clock signal into another kit to generate NRZ L, RZ, and

Manchester respectively on CRO.

4. Signal can be matched by seeing periodic repetition.5. Unplugged the kits & CRO.

OBSERVATION TABLES:


Pulse


23/32



23

Data

NRZ

RZ

Manchester

CONCLUSION:

QUESTIONS:1. Why line coding is required in digital communication?

2. What is the advantages of manchaster coding?

3. Compare RZ with NRZ coding scheme.

4. Define jitter.


24/32



24

EXPERIMENT: - 5

OBJECTIVE: Study of Amplitude Shift Keying Modulation & Demodulation Technique


8 bit Variable Binary Data Generator, Data Formatting and Carrier Modulation TransmitterTrainer, Carrier Demodulation & Data Reformatting Receiver Trainer, CRO, patch cords.

THEORY:

Amplitude Shift Keying: The simplest method of modulating a carrier with a data stream is tochange 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 byswitching On the carrier wheneverthe 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'. Thistechnique 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 ASKwaveform is generated by a balanced modulator circuit, also known as a linear multiplier. As the

name suggests, the device multiplies the instantaneous signal at its two inputs. The outputvoltage being product of the two input voltages at any instance of time. One of the input 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 otherinput which is the information signal to be transmitted, is DC coupled. It is known as modulatingsignal.

Amplitude Shift Keying: The data stream applied is unipolar i.e. 0 volts at logic '0' & + 5 Voltsat logic '1'. The output of balanced modulator is a sine wave, unchanged in phase when a data bit


25/32



25

l' is applied to it. In this case the carrier is multiplied with a positive constant voltage when thedata 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. Themethod 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 shows the functional

blocks required in order to demodulate the ASK waveform at receiver.

ASK Demodulator

BLOCK DIAGRAM:-


26/32



26

PROCEDURE:

1. Make the connection according to the circuit diagram.2. Connect Binary Data Generator to the ASK modulator with desired data pattern output to

CRO.3. Connect ASK modulator output on CRO.

4. Now demodulate the ASK modulator output at receiver side.5. Find the transmitted data pattern on CRO

OBSERVATION TABLE:-

Signal Amplitude Frequency

Carrier

Data

ASK1s

0s

Data at Receiver


27/32



27

CONCLUSION:

QUESTIONS:1. Give the application of ASK.2. List out the disadvantages of ASK.3. Define symbol rate.


28/32



28

EXPERIMENT: - 6

OBJECTIVE:

Study of Frequency Shift Keying Modulation & Demodulation Technique


8 bit Variable Binary Data Generator, Data Formatting and Carrier Modulation TransmitterTrainer, Carrier Demodulation & Data Reformatting Receiver Trainer, CRO, patch cords.

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 higherfrequency is used to represent a data '1' & lower frequency a data '0', the resulting Frequency

shift keying waveform appears as shown in figure. Thus

Data = 1 high frequency Data = 0 low frequency

Frequency Shift Keying Modulator :On a closer look at the FSK waveform, it is apparent that it can be represented as the

sum of two ASK waveforms.


29/32



29

FSK Demodulator: The demodulation of FSK waveform can be carried out by a phase lockedloop. As known, the phase locked loop tries to 'lock' to the input frequency. It achieves this by

generating corresponding output voltage to be fed to the voltage controlled oscillator, if anyfrequency deviation at its input is encountered. Thus the PLL detector follows the frequency

changes & generates proportional output voltage. The output voltage from PLL contains the

carrier components. Therefore the signal is passed through the low pass filter to remove them.The resulting wave is rounded to be used for digital data processing. Also, the amplitude levelmay be very low due to channel attenuation. The signal is 'Shaped Up' by feeding it to the

voltage comparator. The functional block diagram of FSK demodulator is shown in the followingfigure.


30/32



30

Since the amplitude change in FSK waveform does not matter, this modulation technique is veryreliable even in noisy & fading channels. But there is always a price

to be paid to gain that advantage. The price in this case is widening of the required bandwidth.The bandwidth increase depends upon the two carrier frequencies used & the digital data rate.

Also, for a given data, the higher the frequencies & the more they differ from each other, thewider the required bandwidth. The bandwidth required is at least doubled than that in the ASK

modulation. This means that lesser number of communication channels for given band offrequencies.

BLOCK DIAGRAM:-

PROCEDURE:


31/32



31

1. Make the connection according to the circuit diagram.2. Connect Binary Data Generator to the FSK modulator with desired data pattern output to

CRO.3. Connect FSK modulator output on CRO.

4. Now demodulate the FSK modulator output at receiver side.5. Find the transmitted data pattern on CRO

OBSERVATION TABLE:-

Signal Amplitude Frequency

Data

Carrier 1

Carrier 2

FSK

Data at Receiver

CONCLUSION:

QUESTIONS:1. What is FSK?

2. Why FSK is preferred over ASK?

3. What is the difference between FM and FSK?


32/32



Ms. Dipika S. Sagne

Subject In-Charge,

ET Department

YCCE

Dr. P.L.Zade

Head of Department,

ET Department

YCCE

lab mannual dcom

Documents