LAB ELECTRONICS

LAB ELECTRONICS

NEW#5, OLD#27, II FLOOR, 10THAVENUE, ASHOK NAGAR, CHENNAI-83.

DIGITAL COMMUNICATION TRAINER

MODEL - X15B

FEATURES:

Lab Digital Communication Trainer is a versatile instrument which includes all

principles of modulation & demodulation techniques.

List of experiments that can be conducted using this trainer are as follows,

1) Time Division Multiplexer.

2) PPM/PWM modulation/demodulation

3) FSK transmitter.

4) FSK receiver.

5) PCM modulation/demodulation.

6) Transmission impairments

This unit consists of the signal source on the top panel of the trainer as mentioned

below.

AF OSCILLATOR:

OUTPUT : SINE /SQUARE

FREQUENCY :

X1 : 200 Hz TO 2 KHz

X10 : 2 KHz to 20 KHz

AMPLITUDE : 0 - 10V (P-P)

LAB ELECTRONICS

X-15B PAGE: 2

LAB ELECTRONICS

NEW#5, OLD#27, II FLOOR, 10THAVENUE, ASHOK NAGAR, CHENNAI-83

LAB ELECTRONICS

CLOCK GENERATOR:

OUTPUT:

1. CLOCK: Clock Frequency varies with frequency control Potentiometer.

2. PULSE: Clock Pulse width varies with Pulse width Control potentiometer.

1.5 MHz SYNC PULSE O/P: Varies from 0.9 MHz to 1.5 MHz (Frequency

control potentiometer provided for fine frequency adjustment)

NOISE INJECTOR: 15V AC varies with 1K trimmer provided on the top panel of

the trainer using as a noise source.

FREQUENCY

RANGE

FREQUENCY

VARIATION

X 0.1 0.7Hz - 10Hz

X 1 7Hz - 100Hz

X 10 70Hz - 1 KHz

X 100 700Hz - 10 KHz

X 1K 7 KHz - 100 KHz

X-15B PAGE: 3

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 1

TIME DIVISION MULTIPLEXER

OBJECTIVES:

1. To construct a pulse duration modulator.

2. To construct a 3-channel Time-Division multiplexed generator which uses pulse

duration modulation (PDM).

3. To study and observe the characteristics of Time Division Multiplexed generator

and verify its operation.

MATERIALS REQUIRED:

1. LAB digital communication trainer.

2. Dual Trace Oscilloscope.

3. Set of patching wires.

INTRODUCTION:

In modern measurement systems, the various components comprising the

system are usually located at a distance from each other. It therefore becomes

necessary to transmit data between them through some form of communication

channels. The term data transmission refers to the process by which the information

regarding the quantity being measured is transmitted to a location for applications

like data processing, recording or displaying.

MULTIPLEXING:

Multiplexing is the process of transmitting several separate information

channels over the same communication circuit simultaneously without interference.

There are two basic types of multiplexing: time division multiplexing (TDM) and

frequency division multiplexing (FDM).

THEORY:

X-15B PAGE: 4

LAB ELECTRONICS


LAB ELECTRONICS

TIME DIVISION MULTIPLEXER:

In TDM, several information channels are transmitted over the same

communication circuit simultaneously using a time-sharing technique. As an

example, PAM waveforms can be generated that have a very low duty cycle. This

means that if a single channel is transmitted, most of the transmission time would be

wasted. Instead, this time is fully utilized by transmitting pulses from other PAM

channels during the intervals. A PAM-TDM waveform for three channels is shown in

figure 1.

The first pulse is a synchronizing pulse, which is used at the receiver in

demultiplexing. The second pulse is amplitude modulated by channel 1, the third by

channel 2 and the fourth by channel 3. This set of pulses is called a frame. Four

complete frames as shown in figure 1.

FIGURE: 1 - THREE CHANNEL TIME DIVISION MULTIPLEX USING SINGLE POLARITY

PAM

The primary advantage of TDM is that several channels of information can be

transmitted simultaneously over a single cable, a single radio transmitter or any other

communications circuit. Also, any type of pulse modulation may be used in TDM. In

fact, many telephone systems use PCM -TDM.

X-15B PAGE: 5

LAB ELECTRONICS


LAB ELECTRONICS

STEP-BY-STEP PROCEDURE:

1. Study the Time Division Multiplexer circuit configuration given on the front panel

of the trainer.

2. Referring to the same circuit as given below, set R1, R2 & R3 to midrange.

FIGURE - 2: CIRCUIT DIAGRAM

3. Turn ON the trainer.

4. Connect your oscilloscope to pin 3 of the 555 IC. Adjust the triggering controls to

obtain a stable display. If you cannot stabilize the display, connect the

oscilloscope's external trigger input to pin 3 of the 4017 IC. Now switch your

oscilloscope to External Triggering. This should trigger the oscilloscope on the

TDM waveform's sync pulse.

+5V +5V +5V

+5V

X-15B PAGE: 6

LAB ELECTRONICS


LAB ELECTRONICS

5. The output waveform should appear as shown in figure 3. Note that the sync

pulse is a relatively short duration pulse while channels 1, 2 and 3 are

approximately equal. Turn trimmer R1, fully clockwise.

FIGURE: 3 - OUTPUT WAVEFORM

6. Return R1 to midrange. Now adjust R2 alternatetively clockwise and

counterclockwise. What happens to the output wave?

7. Return R2 to midrange. Adjust R3 fully clockwise and then fully counterclockwise.

What happens to the output wave when R3 is adjusted?

8. Turn off your trainer and read the following discussion.

DISCUSSION:

The circuit of figure 2 is a Time Division Multiplex Generator. The output wave

is as shown in figure 3 with pulse duration modulation used for channels 1, 2 and 3.

The 4017-decade counter alternately connects different timing resistors to the 555

IC. It first connects R4 which determines the sync pulse duration. When the 555 IC

output goes low, pin13 on the 4017 steps the counter to the next pulse. In this case,

R1 becomes the timing resistor. Therefore, when R1 is adjusted, the first pulse's

duration is changed.

CHANNELS 1,2,3 &TDM

OUTPUT

CRO ADJUSTMENT

TIME/DIV VOLT/DIV

1ms 2V

X-15B PAGE: 7

LAB ELECTRONICS


LAB ELECTRONICS

This is channel 1. Also, when R2 is adjusted, channel - 2's pulse duration changes.

The same is true for R3 and channel-3. Thus, this circuit is a time division multiplex

generator with the input signals being the position of R1, R2 and R3.

FIGURE: 4 - PIN DIAGRAM OF 4017

SIMULATED OUTPUTS

TIME DIVISION MULTIPLEXER

X-15B PAGE: 8

LAB ELECTRONICS


LAB ELECTRONICS

SIMULATED OUTPUT OF TIME DIVISION MULTIPLEXER

X-15B PAGE: 9

LAB ELECTRONICS


LAB ELECTRONICS

Channel -1

C1 C1

Channel-2

C2

Channel-3

C1 C2 C3 Sync C1 C2 C3

Note: Here C1, C2 and C3 are of same frequencies.

TDM OUTPUT

Sync Pulse

X-15B PAGE: 10

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM

X-15B PAGE: 11

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 2

PULSE CODE MODULATION & DEMODULATION

OBJECTIVE:

1. To construct and study the Pulse Code Modulator and observe its waveform

2. To observe the pulse decoded waveform.

MATERIALS REQUIRED:



3. Set of patching Cords.

THEORY OF PULSE CODE MODULATION:

Pulse code modulation or PCM is the major form of digital pulse modulation.

In PCM, the modulating signal is sampled, just as in other forms of pulse modulation.

The sample amplitude is then converted into a binary code and transmitted as a

stream of pulses.

In the other forms of pulse modulation, the sample amplitude is converted

directly into pulse amplitude, duration or position. However, in PCM, since the

amplitude must be transmitted as a specific number out of a limited range of

numbers, the sample amplitude must first be quantized. That is, each sample

amplitude must be converted to the nearest standard amplitude or quantum. For

example, suppose the PCM system has a total signal amplitude range of 7V and

each 1V level corresponds to a specific binary code. Therefore, for this system each

quantum or standard level is 1V. This is shown in figure 1 for a sine wave signal with

8 quantum steps from 0V to 7V. Note that the quantizing waveform is actually a form

of PAM, although it is limited to the quantum steps and is not continuously variable.

You'll notice that the first sampling point in figure 1 is approximately 3.3V.

Since there is no quantum level at this voltage; it is represented by the nearest level,

which is 3V. This occurs at many places on the waveform. This error or distortion is

called quantizing noise.

X-15B PAGE: 12

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE - 1: A QUANTIZED SINE WAVE

It is noise because the errors are random. This is because the difference between

the quantum level and the actual signal at any instant is completely unpredictable.

The obvious method of reducing quantizing noise is to increase the number of

quantum levels until the noise level is acceptable. However, increasing the number

of levels increases the transmission bandwidth, so a compromise must be made

between acceptable quantizing noise and bandwidth.

FIGURE - 2: CODING THE QUANTIZED WAVE

After quantization has occurred, each sample must be coded as a binary

number before it can be transmitted as PCM. Figure 2 shows the results of coding

the quantized waveform from figure 1. Since there are only 8 quantum levels, they

can be represented by a 3-bit binary word, with 000 representing 0V and 111

representing 7V.

QUANTIZATION

LEVELS

SAMPLING PULSES

SAMPLING POINT

QUANTIZING

WAVEFORM

X-15B PAGE: 13

LAB ELECTRONICS


LAB ELECTRONICS

Once the quantizing waveform is coded, each sequential sample is transmitted as a

pulse code. A table comparing the quantizing level, binary number and pulse code is

shown in figure 3.

FIGURE - 3: THREE BIT PCM

A PCM receiver is shown in figure 4. It is made up of a PCM-to-PAM

converter and a low-pass filter to convert that PAM back to the original modulating

signal. It is, in essence, a digital-to-analog converter.

As mentioned earlier, the primary advantage of PCM is its much better

immunity to noise and interference. For example, a typical PCM transmission can be

sent over a communication channel having a signal-to-noise ratio of 21 dB with

minimal error. In fact, the error would be just one pulse missed or decoded

improperly every 17 minutes. If the signal-to-noise ratio is improved to 23 dB, the

error rate drops to one error every four months. To achieve this; low error rate in an

AM system would require a signal-to-noise ratio of 60 to 70 dB.

QUANTIZING LEVEL

BINARY NUMBER

PULSE CODE

X-15B PAGE: 14

LAB ELECTRONICS


LAB ELECTRONICS

INPUT PCM

TO PAM

LOW PASS

FILTER OUTPUT

DIGITAL TO ANALOG

CONVERSION

FIGURE - 4: PCM RECEIVER

INTRODUCTION:

The purpose of a communication system is to transmit information bearing

signals from a source, located at one point in space, to a user destination, located at

another point. As a rule, the message produced by the source is not electrical in

nature. Accordingly, an input transducer is used to convert the message generated

by the source into a time-varying electrical signal called the message signal. By

using another transducer at the receiver the original message is recreated at the

user destination. Figure-5 shows the block diagram of a communication system. The

system consists of three major parts: 1) transmitter, 2) communication channel, and

3) receiver.

FIGURE - 5: COMMUNICATION SYSTEM

The main purpose of the transmitter is to modify the message signal and to

make suitable for transmission over the channel. This modification is achieved by

means of a process known as modulation, which involves varying some parameter of

a carrier wave (e.g., the amplitude, frequency or phase of a sinusoidal wave) in

accordance with the message signal.

X-15B PAGE: 15

LAB ELECTRONICS


LAB ELECTRONICS

The communication channel may be a transmission line (as in telephony and

telegraphy), an optical fiber (as in optical communication), or merely free space in

which the signal is radiated as an electromagnetic wave (as in radio and television

broadcasting). In propagating through the channel, the transmitted signal is distorted

due to non-linearity and/or imperfections in the frequency response of the channel.

Other sources of degradation are noise and interference picked up by the signal

during the course of transmission through the channel. Noise and distortion

constitute two basic problems in the design of communication systems.

Usually, the transmitter and receiver are carefully designed so as to minimize

the effects of noise and distortion on the quality of reception. The main purpose of

the receiver is to recreate the original message signal from the degraded version of

the transmitted signal after propagation through the channel. This recreation is

accomplished by using a process used in the transmitter. An analog signal is a

continuous function of time, with the amplitude being continuous as well. Analog

signals arise when a physical waveform such as an acoustic or a light wave is

converted into an electrical signal. The conversion is effected by means of a

transducer; examples include the microphone, which converts sound pressure

variations into corresponding Voltage or current variations, and the photoelectric cell,

which does the same for light intensity variations. On the other hand, a discrete-time

signal is defined only at discrete times. Thus, in this case the independent variable

takes on only discrete values, which are usually uniformly spaced. Consequently,

discrete time signals are described as sequences of samples whose amplitudes may

take on continuous values.

When each samples of discrete time signal is quantized (i.e., its amplitude is

only allowed to take on a finite set of discrete values) and then coded, the resulting

signal is referred to as a digital signal. The output of a digital computer is an example

of a digital signal. Naturally, an analog signal may be converted into digital form by

sampling it in time, then quantizing and coding it.

In pulse-code modulation (PCM) the message signal is sampled and the

amplitude of each sample is rounded off to the nearest one of a finite set of allowable

values, so that both time and amplitude are in discrete form. This allows the

message to be transmitted by means of coded electrical signals thereby

distinguishing PCM from all other methods of modulation.

X-15B PAGE: 16

LAB ELECTRONICS


LAB ELECTRONICS

The use of digital representation of analog signals (e.g. voice, video) offers

the following advantages: 1) Ruggedness to transmission noise and interference, 2)

Efficient regeneration of the coded signal along the transmission path and 3) The

possibility of a uniform format for different kinds of base-band signals. These

requirement and increased system complexity are in increase. These advantages,

however, are attained at the cost of increased transmission bandwidth requirement

and increased system complexity.

With the increasing availability of wide-band communication channels,

coupled with the emergence of the requisite device technology, the use of PCM has

become a practical reality. In pulse-duration modulation (PDM) and pulse-position

modulation (PPM) only time is expressed in discrete form, whereas the respective

modulation parameters (namely, pulse amplitude, duration, and position) are varied

in a continuous manner in accordance with the message. Thus, in these modulation

systems, information transmission is accomplished in analog form at discrete times.

ELEMENTS OF PULSE-CODE MODULATION (PCM):

Pulse-code modulation systems are considerably more complex than PAM,

PDM, and PPM systems, in that the message signal is subjected to a greater

number of operations. The essential operations in the transmitter of a PCM system

are sampling, quantizing, and encoding, as shown in figure-6 (a). The quantizing and

encoding operations are usually performed in the same circuit, which is called an

analog-digital converter. The essential operations in the receiver are regeneration of

impaired signals, decoding, and demodulation of the train of quantized samples.

Regeneration usually occurs at intermediate points along the transmission route as

necessary. When time-division multiplexing is used, it becomes necessary to

synchronize the receiver to the transmitter for the overall system to operate

satisfactorily.

X-15B PAGE: 17

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE- 6 (B): BLOCK DIAGRAM OF A PULSE CODER & DECODER IC

SAMPLING:

The incoming message wave is sampled with a train of narrow rectangular

pulses so as to closely approximate the instantaneous sampling process. In order to

ensure perfect reconstruction of the message at the receiver, the sampling rate must

be greater than twice the highest frequency component of the message wave in

accordance with the sampling theorem. In practice, a low-pass filter is used at the

front end of the sampler in order to exclude frequencies greater than before

sampling. Thus, the application of sampling permits the reduction of the continuously

varying message wave to a limited number of discrete values per second.

AUTO

ZERO

5-V

REFERENCE

SUCCESSIVE APPROXIMATION

REGISTER

OUTPUT PCM

BUFFER

CONTROL

LOGIC

INPUT PCM

BUFFER

ANALOG

IN

ANALOG

OUT

INPUT SAMPLE &

HOLD

OUTPUT SAMPLE &

HOLD

PCM OUT

PCM IN

COMPARATOR

NON- LINEAR D/A

CONVERTER

FIGURE- 6 (A): BLOCK DIAGRAM OF PULSE CODER & DECODER

X-15B PAGE: 18

LAB ELECTRONICS


LAB ELECTRONICS

QUANTIZING:

A continuous signal, such as voice, has a continuous range of amplitudes and

therefore its samples have a continuous amplitude range. In other words, within the

finite amplitude range of the signal we find an infinite number of amplitude levels. It is

not necessary in fact to transmit the exact amplitudes of the samples. Any human

sense (the ear or the eye), as ultimate receiver, can only detect finite intensity

differences. This means that the original continuous signal may be approximated by

a signal constructed of discrete amplitude selected on a minimum error basis from

an available set. The existence of a finite number of discrete amplitude levels is a

basic condition of PCM. Clearly, if we assign the discrete amplitude levels with

sufficiently close spacing, we may make the approximated signal practically

indistinguishable from the original continuous signal.

FIGURE - 7: ILLUSTRATION OF THE QUANTIZING PRINCIPLE A) QUANTIZING

CHARACTERISTIC, B) CHARACTERISTIC OF ERRORS IN QUANTIZING,

C) A QUANTIZED SIGNAL WAVE.

The conversion of an analog (continuous) sample of the signal into a digital

(discrete) form is called the quantizing process. Graphically, the quantizing process

means that a straight line representing the relation between the input and output of

linear continuous system is replaced by a staircase characteristic, as in figure-7 (a).

INPUT WAVE

QUANTIZED OUPTUT

DIFFERENCE BETWEEN CURVES

1 & 2

MAGNITUDE

TIME

(a)

(b)

X-15B PAGE: 19

LAB ELECTRONICS


LAB ELECTRONICS

The difference between adjacent discrete values is called a quantum or step

size. Signals applied to a quantizer, with the input-output characteristic of figure-7

(a) are sorted into amplitude slices (the threads of the staircase), and all input

signals within plus or minus half a quantum step of the mid-value of a slice are

replaced in the output by the mid value in question.

The quantizing error consists of the difference between the input and output

signals of the quantizer. It is apparent that the maximum instantaneous value of this

error is half of one quantum step and the total range of variation is from minus half a

step to plus half a step. In part (b) figure-7 the error is shown plotted as a function of

the input signal, and in part (c) of the figure a typical variation of the error as a

function of time is indicated.

X-15B PAGE: 20

LAB ELECTRONICS


LAB ELECTRONICS

BASICS PCM WAVEFORMS WITH BINARY CODE:

FIGURE- 8: PARALLEL PCM WAVEFORMS WITH QUANTIZED LEVEL

(A)

(B)

X-15B PAGE: 21

LAB ELECTRONICS


LAB ELECTRONICS

ENCODING:

In combining the processes of sampling and quantizing, the specification of a

continuous base-band signal becomes limited to a discrete set of values, but not in

the form best suited to transmission over a line or radio path. To exploit the

advantages of sampling and quantizing, we require the use of an encoding process

to translate the discrete set of sample values to a more appropriate form of signal.

Any plan for representing each of this discrete set of values as particular

arrangement of discrete events is called a code. One of the discrete events in a code

is called a code element or symbol. For example, the presence or absence of a

pulse is a symbol. A particular arrangement of symbols used in a code to represent a

single value of the discrete set is called a code word or character.

REGENERATION:

The most important feature of PCM systems lies in the ability to control the

effects of distortion and noise produced by transmitting a PCM wave through a

channel.

This capability is accomplished by reconstructing the PCM wave by means of

a chain of regenerative repeaters located at sufficiently close spacing along the

transmission route.

As in figure-9 a regenerative repeater, namely, equalization, timing, and

decision-making, performs three basic functions. The equalizer shapes the received

pulses so as to compensate for the effects of amplitude and phase distortions

produced by the transmission characteristics of the channel.

FIGURE: 9 - BLOCK DIAGRAM OF A REGENERATIVE REPEATER

Decision

Making

Device

Amplifier

Equalizer

Timing

Circuit

Distorted

PCM Wave

Regenerated

PCM Wave

X-15B PAGE: 22

LAB ELECTRONICS


LAB ELECTRONICS

The timing circuitry provides a periodic pulse train, derived from the received

pulses, for sampling the equalized pulses at the instants of time where the signal-to-

noise ratio is a maximum. The decision device is enabled, when at the sampling time

determined by the timing circuitry, the amplitude of the equalized pulse plus noise

exceeds a predetermined voltage level. Thus, for example, in a PCM system with on-

off signaling, the repeater makes a decision in each bit interval as to whether or not a

pulse is present.

If the decision is "yes", a clean new pulse is transmitted to the next repeater.

If, on the other hand, the decision is "no", a clean base line is transmitted. In this

way, the accumulation of distortion and noise in a repeater span is completely

removed, provided, that the disturbance is not too large to cause an error in the

decision making process. Ideally, except for delay, the regenerated signal is exactly,

the same as the signal originally transmitted.

DECODING:

The first operation in the receiver is to regenerate (i.e., reshape and clean up)

the received pulses. These clean pulses are then regrouped into code words and

decoded (i.e., mapped back) into a quantized PAM signal. The decoding process

involves generating a pulse amplitude of which is the linear sum of all the pulses in

the code word, with each pulse weighted by its place-value (20, 21, 22, 23...) in the

code.

FILTERING:

The final operation in the receiver is to recover the signal wave by passing the

decoder output through a low-pass reconstruction filter whose cutoff frequency is

equal to the message bandwidth . Assuming that the transmission path is error-

free, the recovered signal includes no noise with the exception of the initial distortion

introduced by the quantization process.

MULTIPLEXING:

In applications using PCM, it is natural to multiplex different message sources

by time-division, where by each source keeps its individuality throughout the journey

from the transmitter to the receiver. This individuality accounts for the comparative

ease with which message sources may be dropped or reinserted in a time-division

multiplex system.

X-15B PAGE: 23

LAB ELECTRONICS


LAB ELECTRONICS

As the number of independent message sources is increased, the time

interval that may be allotted to each source has to be reduced, since all of them must

be accommodated into a time interval equal to the reciprocal of the sampling rate.

This in turn means that the allowable duration of a code word representing a single

sample is reduced. However, pulses tend to become more difficult to generate and to

transmit as their duration is reduced. Furthermore, if the pulses become too short,

impairments in the transmission medium begin to interfere with the proper operation

of the system. Accordingly, in practice, it is necessary to restrict the number of

independent message sources that can be included within a time-division group.

SYNCHRONIZATION:

For a PCM system with time-division multiplexing to operate satisfactorily, it is

necessary that the timing operations at the receiver, except for the time lost in

transmission and regenerative repeating, follow closely the corresponding operations

at the transmitter. In a general way, this amounts to requiring a local clock at the

receiver keep the same time as a distant standard clock at the transmitter, except

that the local clock is somewhat slower by an amount corresponding to the time

required to transport the message signals from the transmitter to the receiver.

One possible procedure to synchronize the transmitter and receiver clocks is

to set aside a code element or pulse at the end of a frame (consisting of a code word

derived from each of the independent message sources in succession) and to

transmit this pulse every other frame only. In such a case, the receiver includes a

circuit that would search for the pattern of 1's and 0's alternating at half the frame

rate, and thereby establish synchronization between the transmitter and receiver.

When the transmission is interrupted, it is highly unlikely that the transmitter and

receiver clocks will continue to indicate the same time for long.

Accordingly, in carrying out a synchronization process, we must set up an

orderly procedure for detecting the synchronizing pulse. The procedure consists of

observing the code elements one by one until the synchronizing pulse is detected.

That is, after observing a particular code element long enough to establish the

absence of the synchronizing pulse, the receiver clock is set back by one code

element and the next code element is observed.

X-15B PAGE: 24

LAB ELECTRONICS


LAB ELECTRONICS

This searching process is repeated until the synchronizing pulse is detected.

Clearly, the time required for synchronization depends on the epoch at which proper

transmission is reestablished.

DEFINITION & APPLICATION OF PULSE CODE

MODULATION

Pulse code modulation (PCM) may be chosen as an example of technologies

for A/D conversion. PCM was invented as early as the 1930s, but did not start to

predominate until the 1960s when integrated transistor circuits became available.

PCM is a type of waveform coding and is standard for voice coding in the telephone

network. The bit rate generated per call - 64 K bits/s - has been a decisive factor in

switching and transmission design.

FIGURE 10: THE VOICE CURVE IS TIME DIVIDED INTO AMPLITUDE VALUES

Sound - movement of particles in an elastic medium, as the definition goes - is

intrinsically analog. This may be illustrated with a curve that shows how the

amplitude (sound level) varies over time. Figure-10 (considerably simplified) shows

such a voice curve. Let us imagine that we measure the amplitude at regular

intervals and jot down the values in a table. Since there are no values between the

readings, the table does not give us the whole truth. But it is quite clear that the

shorter the period of time between the readings, the better the description of the

voice curve.

SAMPLING:

Reading the amplitude at regular intervals is called sampling. It is important to

take the samples on the voice curve at suitable intervals, which means that the

quality obtained should allow us to clearly recognize each other's voices.

X-15B PAGE: 25

LAB ELECTRONICS


LAB ELECTRONICS

Taking too many samples is uneconomical; a suitable sampling frequency is

8,000 samples per second. The result will be a pulse amplitude-modulated (PAM)

signal where each pulse directly corresponds to the amplitude of the voice curve.

See figure-11.

FIGURE 11: PULSE AMPLITUDE-MODULATED SIGNAL

The sampling theorem;

Now, how have we reached the conclusion that a frequency of 8,000 samples

per second is sufficiently close between readings? The answer lies in the sampling

theorem, which states that:

"All the information in the original signal will be present in the signal described by the

samples”, if:

1. The original signal has a limited bandwidth, that is, it does not contain any

component with a frequency exceeding a given value, B

2. The sampling frequency is greater than twice the highest frequency in the original

signal; that is, >2 x B.

Since telephone connections operate in the 300 - 3,400 Hz band, 8,000 Hz is

a sampling frequency that meets the primary requirement for transmission quality: no

information should be lost. The sampling frequency is twice the maximum frequency,

which is significantly lower than 8 KHz.

QUANTIZATION:

Quantization means that we measure the amplitude of the pulses in the PAM

curve and assign a numerical value to each pulse. To avoid having to handle an

infinite number of numerical values, we divide the amplitude levels into intervals and

assign the same value to all samples within a given interval.

X-15B PAGE: 26

LAB ELECTRONICS


LAB ELECTRONICS

See figure-12, in principle, this is analogous with the way in which a person's age

may be viewed. A 25-year-old belongs to the "quantization interval" twenty-five years

for no less than 365 days.

FIGURE 12: SAMPLES WITH THE CORRESPONDING QUANTISED VALUES

Quantization also means that we forgo accuracy: the series of digits is not

really the whole truth about the voice curve. We call this deviation quantizing

distortion. See figure-12. But we will have a limited number of numerical values to

transmit, the equipment can be made less complex, and the risk of transmission

errors is reduced. In telephony, 256 quantizing intervals are used. Consequently,

there are 256 values to be transmitted.

Figure-12 shows that there are also other problems. In the figure-13, the

quantizing intervals are equally large and we will have the same quantizing distortion

regardless of the amplitude. But if we set distortion in relation to the amplitude, the

relationship will vary. A low distortion-to-volume ratio is crucial to audibility. This

means that a weak voice will be significantly disturbed if equally large quantizing

intervals are used.

One way of solving this problem is to make the quantizing intervals small

enough, so that even low amplitude deviations can be transferred with sufficiently

good audibility. Then again, we will have unnecessarily small intervals for the high

amplitudes, which also mean that there will be an unnecessary amount of numerical

values to be transferred.

X-15B PAGE: 27

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE 13: QUANTIZING DISTORTION

The ideal thing must be to allow the quantizing intervals to increase with amplitude.

The amplitude/distortion relationship should preferably be constant. In addition, it is

important to find the right relationship between the number of quantizing intervals

and the desired transmission quality. Here, too, we can refer to the common way of

saying or writing a person's age. At the beginning of our lives, we specify age in days

- then in weeks and months. Not until a child is two years of age do we start to use

"full-year quantizing intervals".

Two models are available. One of them, the A-law, is illustrated in figure-14

The other one, the µ-law, follows the same principle but has fifteen segments instead

of thirteen. The A-law is used in Europe, and the µ-law is used in the US.

X-15B PAGE: 28

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE 14: THE A-LAW

CODING:

Now it remains for us to give our 256 possible values a suitable layout for

transmission. Let us use binary pulses; that is, pulses with only two levels. Eight

such pulses, or bits, will suffice to form a unique code for each interval value (28 =

256). The equipment need only be capable of distinguishing between two pulse

levels, and of counting to eight. This technique-the elements of computer technology

is ideally suited for telephony applications.

FIGURE 15: BINARY PULSES

X-15B PAGE: 29

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE 16: QUANTISED VALUES WITH THE CORRESPONDING BINARY CODE

Three processing steps;

To sum up, there are three steps between the analog voice and the

digital transmission link.

1. Sampling, where we measure the amplitude 8,000 times per second.

2. Quantizing, where we assign one out of 256 values to each sample.

3. Coding, where each quantised value is expressed as a binary code of eight bits.

FIGURE 17: PCM WORD, EIGHT BITS

The result of this pulse code modulation process - the eight-bit binary code -

is called a PCM word. See figure-17. One PCM word corresponds to one sample.

8,000 PCM words are generated per second, and for each call we will have a bit

stream of 8 x 8,000 = 64,000 K bits/s in the digital link. The ITU-T calls this type of

voice coding "64 K bits/s PCM".

\

X-15B PAGE: 30

LAB ELECTRONICS


LAB ELECTRONICS

FROM DIGITAL TO ANALOG:

In the receiving equipment, the pulse train is received and converted into

analog form; that is, the sound curve is reproduced. This, too, is a process in three

steps.

FIGURE 18: A/D CONVERSION, TRANSMISSION AND D/A CONVERSION

Firstly, regeneration: the binary pulse train is received and the PCM words are

reproduced. Secondly, the PCM words are interpreted in a decoder and translated

into quantised amplitude values. Thirdly, the voice curve itself is reconstructed, and

we will again have an analog signal that can be made audible at the receiving end.

See figure-18.

Today, the PCM technique is also used to record music on compact discs

(CD) but in this application, the sampling frequency is higher: 44.1 KHz in contrast

with 8 KHz for telephony.

1. SYNCHRONOUS PULSE GENERATOR: (SYNC PULSE):

This part of the circuit generates a MHz SYNC Pulse signal using a transistor,

Schmitt trigger and a -flip flop. The frequency of this SYNC Pulse can be varied

using the 5K potentiometer (10K potentiometer).

2. PULSE CODER & DECODER:

a) ANALOG TO PCM (TRANSMIT SECTION):

The analog input signal is placed on the uncommitted op-amp’s terminals.

The op-amp allows for input gain adjustment if necessary to either 0 dB or the

system’s 0 levels.

X-15B PAGE: 31

LAB ELECTRONICS


LAB ELECTRONICS

The op-amp also acts as a 2nd order analog anti-aliasing filter by band-limiting

the input to less than half of the sampling frequency as per the Nyquist Rate

Theorem. The analog signal is filtered by a cosine filter, a 6th order low-pass filter,

and the high-pass filter before being sampled. The sampling is performed by a

capacitor array at a rate of 8 KHz and the value fed into the encoder.

From the encoder the 8 bit PCM data is clocked out by the shift clock. Lastly,

an auto-zero loop (without any external capacitor provides cancellation of any DC

offset by integrating the single bit of the PCM data and feeding it back to the non-

inverting input of the comparator, and a sign bit - fixation circuitry reduces idle

channel noise during quiet periods).

b) PCM TO ANALOG (RECEIVE SECTION):

The PCM data is shifted into the decoder's input buffer register once every

sampling period. Once the PCM data has been shifted into the decoder register a

charge proportional to the received PCM data work value appears on the decoder's

capacitor array. A sample and hold circuit integrates to the charge value and holds

that value for the rest of the sampling period. Then low-pass switched capacitor filter

smoothes the signal and performs loss equalization to compensate for the (SIN X)/X

distortion due to sample and hold operation. The low pass filter's output is then

buffered and available for driving electronic hybrids directly.

c) TIMING REQUIREMENTS:

The 8 KHz transmit and receive sampling strobes need not be exactly 8 bit

periods wide. The codec has an internal bit counter that counts the number of data

bits shifted and forces the PCM output into a high-impedance state after the 8th bit-

has been shifted out. This allow the strobe signal to have any duty cycle as long as

its repetition rate is 8KHz and the shift clock is synchronized to it and the clock rate

is either 1.537MHz, 1.544MHz, or 2.048MHz. Note that all internal clocks for the

switched capacitor filters and timing conversions are automatically derived; no

external control signal for clock selection is required.

d) POWER DOWN CIRCUITRY:

The codec can be powered down in two ways. The most direct power down

command is to force the PD (Pin 14) mode select low. This will shut down the chip

regardless of the strobes. The second method is to stop strobing with the clock (Pin

11) input.

X-15B PAGE: 32

LAB ELECTRONICS


LAB ELECTRONICS

The clock can be held high, low or floating just so long as its state is not

changed. After the chip has been shut down the PCM is locked into a high

impedance state and the AOUT is connected to AGND for about 1ms to avoid output

noise to the system.

e) A-LAW CHARACTERISTICS:

Compression (refer to figure-19 (a)) allows more channels to be multiplexed

on a given transmission media by reducing the bandwidth of each individual channel.

Figure-19 (b) shows the A-LAW companding transfer functions used in telephony to

convert the speaker's analog voice signal into PCM. Figure-19(c) shown the

expansion transfer function used to convert the digital PCM signal back into an

analog signal for the end telephone user to hear.

FIGURE - 19A

QUANTIZING LEVEL

STRONG SIGNAL

WEAK SIGNAL

1 WITHOUT A COMPANDER 2 WITH A COMPANDER

X-15B PAGE: 33

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE - 19B: THE A-LAW A/D COMPANDING TRANSFER FUNCTION

FIGURE - 19C: THE A- LAW D/A COMPANDING TRANSFER FUNCTION

10100101

10110101

10000101

10010101

11100101

11110101

01110101

01100101

00010101

00000101

00110101

00100101

11000101

01000101

11010101

01111111 01010101

11111111

DIGITAL

OUTPUTS

ANALOG INPUT

-3 -2 -1 0 1 2 3

01010101

01111111

11111111

11010101

DIGITAL INPUTS BY DECODER

3 2 1 0 -1 -2

-3

ANALOG

INPUTS

X-15B PAGE: 34

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENTAL STEP-BY-STEP PROCEDURE:

1. Study the PCM & Demodulation Circuit configurations given on the front panel of

the trainer.

2. Connect the Patch Cords as shown in wiring diagram.

3. Switch ON the trainer.

a. Connect Oscilloscope across SYNC pulse output terminal (adjusted to 700KHz)

and ground. Observe the SYNC pulse output of frequency to be approx 0.05 MHz

to1.5 MHz & Amplitude will be approx 4V(P-P).

OSCILLOSCOPE SETTINGS: (For SYNC pulse)

Time / Div = 0.5μs to 1μs

V / Div = 2V.

SIMULATED OUTPUT OF SYNC PULSE GENERATOR O/P

b. Connect oscilloscope across clock output and Gnd. Observe the clock output.

Select frequency selector switch to x 100 and adjust the frequency to be 3.5 KHz

by adjusting frequency control knob amplitude will be approx 4V (p-p).

OSCILLOSCOPE SETTINGS: (For CLOCK)

Time / Div = 0.1ms to 50s

V / Div = 2V

X-15B PAGE: 35

LAB ELECTRONICS


LAB ELECTRONICS

C. Connect Oscilloscope across the AF output terminal and ground in the AF

oscillator section. Select function selector switch to SINE (on the front panel AF

Oscillator). You can observe sine output on CRO at this terminal. By varying the

frequency control and amplitude control potentiometers adjust frequency to be 1

KHz and amplitude to be 4V (P-P).

OSCILLOSCOPE SETTINGS: (For sine output)

Time / Div = 0.2ms

V / Div = 2V

4. Switch OFF the trainer and patch the circuit as shown in the wiring diagram.

You will observe that sync pulse output, clock and AF output are connected from

the top panel trainer to PCM and demodulation section.


6. Connect CRO across the PCM output and ground at pin 15 of IC44233. For

observing PCM output;

Verify that AF (SINE output) frequency to be approx 1KHZ and amplitude to be

4V (P-P).

Verify that clock generator output frequency range to be approx 7KHz and

amplitude to be 4V (P-P). Slightly adjust the frequency control Knob to observe

the PCM output.

Verify that SYNC pulse output frequency range to be approx 1.5MHz and

amplitude to be 4V (P-P). Keep pulse width control to maximum position

PCM OUTPUT:

7. Note: 60MHz dual trace oscilloscope can be used for better clarity while

observing PCM output

Time / Div = 0.1ms to 0.2ms

V / Div = 2V

X-15B PAGE: 36

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM

INDICATES PATCHING CONNECTIONS

X-15B PAGE: 37

LAB ELECTRONICS


LAB ELECTRONICS

SIMULATED OUTPUT OF PCM

AUDIO INPUT SETTING‘S

SIMULATED OUTPUT OF PCM

7. It is a very precise adjustment to obtain the PCM output. If you are not getting

the PCM output slightly adjust clock generator frequency control.

8. Ignore the Harmonic Distortion if any in the PCM output, as it is an educational

trainer only.

2

INPUT PCM O/P

X-15B PAGE: 38

LAB ELECTRONICS


LAB ELECTRONICS

9. Finally observe the audio output at pin 5 of IC44233, which is same as the audio

input. Vary the Gain potentiometer for amplitude variation. You can observe

some weak signals as shown in the figure -19(A). If required slightly adjust clock

generator frequency control and audio amplitude control.

IC 44233 TIMING DIAGRAM:

This is a LS-TTL compatible open drain output. It is active only during

transmission of digital PCs output for 8 bit periods of the transmit clock signal

following a positive edge on the transmit SYNC input data is clocked out by the

positive edge of 470 ohms, although only one 470 ohm resistor is required for eight

codes.

The timing chart of IC44233 is as given below.

FIGURE- 44233 TIMING CHART

X-15B PAGE: 39

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 3

PULSE POSITION AND PULSE WIDTH

MODULATION

OBJECTIVE:

1. To construct a pulse position carrier generator.

2. To show how this Pulse Position Modulation (PPM) and Pulse Width Modulation

(PWM) is modulated by any external AF modulating frequency.

MATERIALS REQUIRED:




INTRODUCTION:

In amplitude and angle modulation, some characteristics of the carrier

amplitude, frequency, or phase is continuously varied in accordance with the

modulating information. However, in pulse modulation, a small sample is made of the

modulating signal and then a pulse is transmitted. In this case, some characteristics

of the pulse is varied in accordance with the sample of the modulating signal. The

sample is actually a measure of the modulating signal at a specific time.

There are several types of pulse modulating systems. Three of the more

common types are pulse amplitude modulation (PAM). Pulse duration modulation

(PDM), and pulse position modulation (PPM). In each of these systems, a

characteristic of the pulse - such as amplitude, duration, or position is continuously

varied in accordance with the modulating signal. This type of pulse modulation,

where a pulse characteristic is continuously varied, is called analog pulse

modulation.

Another type of pulse modulation is pulse code modulation (PCM), which is

digital pulse modulation. With PCM, the modulating signal is sampled and then

quantized. In quantization, each sample is assigned a specific numerical value

according to its amplitude.

X-15B PAGE: 40

LAB ELECTRONICS


LAB ELECTRONICS

This numerical value is then represented by a group of pulses, which represent the

modulating signals value in the binary number system. This system has many

advantages and, therefore, has many applications in modern communications.

THEORY:

In pulse time modulation (PTM), the modulating signal is sampled, just as it is

in PAM. However, in PTM, the amplitude of the sample is indicated by a timing

variation of the modulated pulse rather than an amplitude variation. The variable

timing characteristics may be the duration, position, or frequency of the pulses.

Therefore, there are three basic types of PTM: pulse duration modulation, pulse

position modulation, and pulse frequency modulation.

This type of PTM is also called pulse width or pulse length modulation,

however, pulse duration modulation (PDM) is the preferred term. There are three

different classifications of PDM: symmetrical PDM, leading edge PDM and trailing

edge PDM. These are shown in figure 1 along with the sine wave modulating signal.

Figure 1 (a) shows a symmetrical PDM waveform. Here, the modulating signal is

sampled and both the leading and trailing edges of the pulse are varied in

accordance with the sample amplitude. When the sample is high, the negative

reference duration, the spacing between the centers of the pulses remains constant.

Leading edge PDM is shown in figure 1 (b). In this type of PDM, the sample

amplitude varies the leading edge of the pulse. The trailing edge of each pulse is

fixed and, therefore, the spacing or timing between each pulses trailing edge is

constant. Figure 1 (c) shows trailing edge PDM. Here, the sample amplitude varies

the trailing edge of the pulse, with the leading edge remaining fixed.

PULSE POSITION MODULATION:

The next form of PTM is known as pulse position modulation (PPM). With this

form of PTM, both pulse amplitude and duration remain constant while the position of

the pulse, relative to a reference pulse, is varied in accordance with the modulating

signal. Figure 2 (b) shows a typical PPM waveform. The modulating signal and the

reference pulses are shown in figure 2 (a). Note that when the modulating signal

goes negative, the output pulse now leads the reference pulse by a proportional

amount.

X-15B PAGE: 41

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE - 1

A typical PWM waveform can be generated using a 555 timer. When the timer is

connected in the monostable mode and triggered with a continuous pulse train, the

output pulse width can be modulated by a signal applied to pin 5. Figure 3 shows the

circuit and the waveform for pulse width modulation. A typical PPM waveform can be

generated by connecting the timer in a stable mode with a modulating signal applied

to the control voltage terminal. The pulse position varies with modulating signal since

the threshold voltage and hence the time delay is varied. Figure 4 shows the circuit

and the waveform for a triangle wave modulating signal.

FIGURE - 2

X-15B PAGE: 42

LAB ELECTRONICS


LAB ELECTRONICS


1. Study the PP & PW Modulation circuit configuration given on the front panel of

the trainer.

2. Connect the Patch Chords as per the wiring diagram -1.

3. Connect the circuit for PWM as shown in figure 3 (Refer wiring diagram-1 in

experiment 3).

4. Switch ON the trainer and check the Power supply to be +5V.

5. Connect a sine wave modulating signal of 900Hz and 5V P-P to pin 5 from sine

output of AF oscillator section.

6. Connect a clock signal of 3.5 KHz frequency as trigger to pin 2.

7. Connect your oscilloscope across pin 3 and ground.

8. Vary the frequency and the amplitude of the modulating signal (sine wave) and

observe the corresponding change in the width of the output pulses.

9. Connect the circuit as shown in figure 4 for pulse position modulation (Refer

wiring diagram - 2 in Experiment 3).

10. Connect a sine wave modulating signal of frequency 1 KHz and amplitude of

5V p-p from an AF OSCILLATOR to pin 5.

11. Connect your oscilloscope to pin 3 and observe the PPM output vary the

frequency of the waveform and observe the corresponding changes in the

position of the output pulses.

FIGURE - 3: PULSE WIDTH MODULATION

X-15B PAGE: 43

LAB ELECTRONICS


LAB ELECTRONICS

CRO OBSERVATION PWM:

Time = 2ms/DIV

Top Trace Modulation I/P = 1V/DIV

Bottom Trace O/P = 2V/DIV

FIGURE - 4: PULSE POSITION MODULATION

CRO OBSERVATION (PPM):

Time = 50s/DIV

Top Trace Modulation Input = 1V/DIV

Bottom Trace Output = 2V/DIV

X-15B PAGE: 44

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM-1

PULSE WIDTH MODULATION

INDICATES THE PATCHING CONNECTIONS

FSK RECEIVER

PP/PW MODULATION

5

4 8

21

37

6

+5V

I/PTRIG

9K1

3K9

PP/PWMO/P

0.01F

1K

NE555

10K

X-15B PAGE: 45

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM-2

PULSE POSITION MODULATION


X-15B PAGE: 46

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 4

PULSE POSITION & PULSE WIDTH

DEMODULATION

OBJECTIVES:

1. To construct the pulse position and pulse width demodulation circuit.

2. To show that the pulse width and pulse position modulation signal is

demodulated.

3. To show that the P.W.M demodulation output is nearly the same as the

modulating frequency by using phase locked loop demodulator.

MATERIALS REQUIRED:




INTRODUCTION:

In pulse modulation, some parameter of a pulse train is varied in accordance

with the message. In pulse analog modulation systems, a periodic pulse train is used

as the carrier wave and some characteristic feature of each pulse (amplitude,

duration or position) is varied in a continuous manner in accordance with the

pertinent sample value of the message.

The phase locked loop is an excellent detector and gives an acceptable signal

to noise ratio from weak and noise-invested signals.

THEORY:

Referring to the circuit diagram is connected as an inverting amplifier with a

voltage gain of 20dB & this provides the necessary amplification of the pulse width

modulated signal. The clipped signal at the output of the op-amp is compatible with

the input of the PLL.

X-15B PAGE: 47

LAB ELECTRONICS


LAB ELECTRONICS


1. Study the circuit diagram for PW/PP demodulation given on the front panel of the

trainer.

2. Ensure that PW modulation circuit is connected as in Experiment - 3

3. Switch on the TRAINER.

4. Connect the pulse width modulated signal from the PWM OUTPUT to the input

terminal at the demodulated o/p section.

5. Connect the oscilloscope at DEMODULATED O/P and observe the modulating

signal waveform (SINE WAVE).

6. You will observe the amplified O/P on the CRO.

7. Compare this demodulated output to the original modulating signal of the pulse-

width modulation on a dual trace oscilloscope. You will find that the demodulated

output is in phase with the original signal.

8. Similarly connect the pulse position modulated (PPM) signal to I/P of

demodulator and repeat the steps from 2 to 7.

X-15B PAGE: 48

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM

PULSE WIDTH/POSITION DEMODULATION


X-15B PAGE: 49

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 5

FSK TRANSMITTER

OBJECTIVE:

1. To design a FSK transmitter, similar to those used in data communication

systems in simplex mode of operation.

2. To measure its "LOW" output frequency (800 Hz to 1000 Hz) when its data input

terminal is grounded and "HIGH" output frequency (1100 Hz to 1300 Hz) when its

data input terminal is connected to VCC.

3. To connect the data input terminal to the pulse train that creates frequency shift

at the output signal in response to the pulse input.

MATERIALS REQUIRED:



3. Set of patching Chords.

INTRODUCTION:

Generated waveforms can be modulated in a variety of ways in order to

convey information or to produce special sound effects. The three best known forms

of modulation are, of course, Amplitude Modulation (AM), Frequency Modulation

(FM), and Frequency-Shift Keying (FSK), but a variety of other forms of modulation,

such as Phase-Shift Keying (PSK), Sweep Modulation and Carrier Keying are also

used. When it is required to transmit digital data over a band pass channel, it is

necessary to modulate the incoming data on to a carrier wave (usually sinusoidal)

with fixed frequency limits imposed by the channel.

The data may represent digital computer outputs, or PCM waves generated

by digitizing voice or video signals, etc. The channel may be a microwave radio link,

or satellite channel, etc. In any event, the modulation process involves switching or

keying the amplitude, frequency, or phase of the carrier in accordance with the

incoming data.

X-15B PAGE: 50

LAB ELECTRONICS


LAB ELECTRONICS

AM INPUT

OR

OUTPUT

MULT. OUT

+VCC

TIMING

CAPACITOR

TIMING

RESISTORS

MULTIPLIER & SINE

SHAPER

CURRENT

SWITCHES

SYMETRY

ADJ

WAVEFORM

ADJ

GROUND

SYNC

OUTPUT

BYPASS

FSK

INPUT

XR - 2206

Thus there are three basic signaling techniques known as amplitude shift

keying (ASK) frequency shift keying (FSK) and phase shift keying (PSK) which may

be viewed as special cases of amplitude modulation, frequency modulation, and

phase modulation respectively.

Ideally, FSK & PSK signals have a constant envelope. The feature makes

them impervious to amplitude non-linearities as encountered in microwave radio

links and satellite channels. Accordingly, we find that, in practice, FSK & PSK signals

are much more widely used than ASK signals.

THEORY:

Frequency-shift keying is a form of frequency modulation in which the 'carrier'

switches abruptly from one frequency to another on receipt of a command or keying

signal. Most oscillator circuits can be subjected to FSK by simply designing them so

that an alternative frequency determining component or parameter is selected on

receipt of the 'key' signal. The 'key' signal may be delivered electro-mechanically via

a switch, or electronically via a transistor gate, etc.

The XR-2206 waveform generator has a terminal that is specifically allocated

for FSK use. Figure 1 shows the practical connections for making a split-supply

sine-wave generating FSK or ‘Warble-Tone’ XR-2206 oscillator.

FIGURE: 1 - XR-2206 SPLIT-SUPPLY F.S.K. SINE-WAVE GENERATOR

X-15B PAGE: 51

LAB ELECTRONICS


LAB ELECTRONICS

This IC has two alternative timing resistor pins (pins 7 and 8) and either pin

can be selected by applying a suitable bias signal to pin 9 of the IC. When the pin 9

FSK input terminal is open circuit or externally biased above 2V with respect to the

negative supply rail, the pin 7 timing resistor is automatically selected and the circuit

operates at a frequency determined by R1 and C1.

When pin 9 is shorted to the negative supply rail or biased below 1V with

reference to the negative supply rail, the pin 8 timing resistor is selected and the

circuit operates at a frequency determined by R2 and C1. The XR-2206 IC can thus

be frequency-shift keyed by simply applying a suitable keying or pulsing signal

between pin 9 and the negative supply rail.


1. Study the FSK transmitter circuit as shown in the front panel of the trainer.

2. Switch on the trainer.

3. In this circuit, the capacitor connected between pins 5 & 6 is C1. The value of C1

is .022F. Initially set R1 to 39 K, R2 to 47K and C1 to 0.022F (Connect digital

Multimeter across the two terminals of 50K trim pots and slowly adjust the value).

4. Connect data input to the GND through a patching wire so that data input is in

“LO” mode. Connect an oscilloscope across the FSK output, pin 2 and GND.

Switch ON the trainer and adjust the oscilloscope for a stable display. What is the

output frequency with the data input at “LO”__________Hz.

5. Now remove the data input from GND so that data input is in “HI” mode. What is

the output frequency now? ____________Hz. What is the frequency shift

_______Hz.

6. Connect the data input, pin 9, to the CLOCK generator output. What do you

observe on the oscilloscope? _______________________

7. Connect channel A to the input wave form and connect channel B to the FSK

output. Observe the frequency shifting in the waveform (Refer simulated

diagram).

X-15B PAGE: 52

LAB ELECTRONICS


LAB ELECTRONICS

DISCUSSION:

In this part of the experiment, you studied about the FSK transmitter. In step

5, you measured its ‘low' output frequency, which should have been between 800

and 1000 Hz. In step 6, you measured its "high" frequency, which should have been

between 1100 and 1300 Hz. The frequency shift should be between 200Hz to 350Hz

(approx). In Step 7, you have connected the data input to a CLOCK source which

simulates a data pulse train. There will be a shift in frequency of the output signal in

response to the data input.

STEP-BY-STEP PROCEDURE (CONT’D):

8. Vary R1 and R2 in accordance with the values given in the Tabular column and

note the high and low frequencies.

9. A high level signal selects the frequency fH = 1/R1C1 and a low level signal

selects the frequency fL = 1/R2C1.

TABULAR COLUMN

R1 R2 fL fH

28K 30K

26K 28K

26K 50K

DISCUSSION:

You have changed the resistances to exaggerate the frequency shift. In this

way you could easily see the FSK signal on your oscilloscope screen.

50K 50K 10f

2 SQUARE DATA

INPUT

150 0.02F

FSK OUTPUT

1F

4.7K

4.7K

XR - 2206

+15V

FIGURE: 2 - PIN OUT DIAGRAM OF XR - 2206 IC

X-15B PAGE: 53

LAB ELECTRONICS


LAB ELECTRONICS

SIMULATED OUTPUT SETTINGS FOR FSK TRANSMITTER

WIRING DIAGRAM


INPUT FSK O/P

X-15B PAGE: 54

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 6

FSK RECEIVER

OBJECTIVES:

1. To construct an FSK receiver using a phase locked loop and an operational

amplifier to demodulate the FSK signal, by adjusting the PLL to the center of the

FSK signal.

2. To prove the fact that the PLL output follows the input level exactly by comparing

the steady state input and output levels.

MATERIALS REQUIRED:




INTRODUCTION:

In computer peripheral and radio (wireless) communications, the binary data

or code is transmitted by means of a carrier frequency that is shifted between two

preset frequencies. Since a carrier frequency is shifted between two preset

frequencies, the data transmission is said to use a frequency shift keying (FSK)

technique.

A very useful application of the 565 PLL is a FSK demodulator. In the 565 PLL

the frequency shift is usually accomplished by driving a VCO with the binary data

signal so that the two resulting frequencies correspond to the logic 0 and logic 1

states of the binary data signal. The frequencies corresponding to logic 1 and logic 0

states are commonly called the mark and space frequencies. Several standards are

used to set the mark and space frequencies.

X-15B PAGE: 55

LAB ELECTRONICS


LAB ELECTRONICS

THEORY:

Figure 1 shows IC 565 configured for FSK. The input frequencies are applied

to pin 2, and the output is taken from pin 7. In addition to the low pass filter, a three-

stage RC filter is connected to pin 7 to remove the carrier from the output. The

output (pin 7) and the reference output (pin 6) are connected to a comparator, which

provides the output pulses. When the input is high, pin 7 goes to a lower DC level,

which produces a pulse at the output of the comparator. The free-running frequency

of the VCO is adjusted by R1 to give a slightly positive voltage at the output when the

input is low.

FIGURE: 1 - CIRCUIT DIAGRAM OF FSK RECEIVER

The IC 565 Phase Locked Loop is a general purpose circuit designed for

highly linear FM demodulation. During lock, the average DC level of the phase

comparator output signal is directly proportional to the frequency of the input signal.

As the input frequency shifts, it is this output signal which causes the VCO to shift its

frequency to match that of the input. Consequently, the linearity of the phase

comparator output with frequency is determined by the voltage-to-frequency transfer

function of the VCO.

+5V

-5V

F

S

K

I

N

P

U

T

X

R

-

2

2

0

6

DATA OUTPUT

5

6

5

565 FSK INPUT

X-15B PAGE: 56

LAB ELECTRONICS


LAB ELECTRONICS

Because of its unique and highly linear VCO, the 565 PLL can lock to and

track an input signal over a very wide range (typically 60%) with very high linearity

(typically, within 0.5%). A typical connection diagram is shown on the front panel of

the trainer.

The source can be direct coupled if the DC resistances seen from pins 2 and

3 are equal and there is no DC voltage difference between the pins. A short between

pins 4 and 5 connects the VCO to the phase comparator. Pin 6 provides a DC

reference voltage that is close to the DC potential of the demodulated output (pin 7).

Thus, if a resistance R2 is connected between pins 6 and 7, the gain of the output

stage can be reduced with little change in the DC voltage level at the output. This

allows the lock range to be decreased with little change in the free-running frequency

(f0). In this manner the lock range can be decreased from +60% of f0 to

approximately 20% of f0 (at +6V).

A small capacitor (typically 0.0011 F) should be connected between pins 7

and 8 to eliminate possible oscillation in the control current source. A single-pole

loop filter is formed by the capacitor C2, connected between pin 7 and positive

supply, and an internal resistance of approximately 3600. FSK refers to data

transmission by means of carrier which is shifted is usually accomplished by driving

a VCO with the binary data signal so that the two resulting frequencies correspond to

the "0" and “1” states (commonly called space and mark) of the binary data signal.

As the signal appears at the input, the loop locks to the input frequency and

tracks it between the two frequencies with a corresponding DC shift at the output.

The loop filter capacitor C2 is chosen smaller than usual to eliminate overshoot on

the output pulse, and a three-stage RC ladder filter is used to remove the carrier

component from the output. The band edge of the ladder filter is chosen to be

approximately half way between the maximum keying rate. The output signal can

now be made logic compatible by connecting a voltage comparator between the

output and pin 6 of the loop. The free-running frequency is adjusted with R1 so as to

result in a slightly positive voltage at the output at fin = 200Hz.

X-15B PAGE: 57

LAB ELECTRONICS


LAB ELECTRONICS


1. Study the circuit diagram in the front panel on the trainer.


3. Feed the FSK transmitter output to the input (pin 2) of the receiver section. Set

the oscilloscope to DC coupling, internal triggering. Now connect CRO across pin

6 of the IC 741 i.e SQUARE output (Refer wiring Diagram) and GND.

4. Set the 10K trimmer in FSK receiver section to mid range, now adjust the 10 K

TRIMMERS until you obtain a square wave output on the oscilloscope display.

This is a very precise adjustment; on one-side of the correct position, the output

will be negative, on the other side, it will be positive. Adjust it to the exact center

of these two indications, which should result in a symmetrical square wave

output.

5. Disconnect the data input pin 9 of the XR-2206 of FSK transmitter from the

clock generator and connect it to data input to GND. What is the data output

level? ___________________.

6. Remove the data from Gnd terminal i.e it will be in “HI” mode. What is the data

output level? _________

7. What mode of operation is the data communications system on your trainer

using? __________________________________

8. Switch OFF your trainer and disconnect the circuit. Read the following

discussion.

DISCUSSION:

In this part of the experiment, you have studied the FSK receiver using 565

PLL (Phase-Lock-Loop) and a 741 op-amp to demodulate the FSK signal. In step 4,

you adjusted the PLL to the center of the FSK signal. Therefore, PLL demodulates

the FSK signal. In steps 5 and 6, you have proved this by comparing steady input

and output levels; the PLL output followed by the input level is exact. The data

communication system on the experimenter is an example of the simplex mode of

operation.

X-15B PAGE: 58

LAB ELECTRONICS


LAB ELECTRONICS

SIMULATED OUTPUT OF FSK RECIVER

WIRING DIAGRAM


I

FSK INPUT

DE MOD O/P

X-15B PAGE: 59

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 7

DIGITAL CODING SYSTEM

OBJECTIVE:

To demonstrate how signals obtained from physical inputs such as switches

and keyboard contacts can be decoded into unique digital base band signals that

can be transmitted over parallel data lines.

MATERIALS REQUIRED:




INTRODUCTION:

In this trainer, you will build a digital coding circuit that will produce a

distinctive output for each possible input. You will first need a digital input, obtained

from the logic switches on the trainer. You will then use the LEDs on the trainer to

observe the operation of the circuit.

Using the circuit shown in trainer, you will generate a distinct fixed length code

for each possible combination of input signals. This type of coding is widely used in

communication systems.


1. In this experiment, you will use both the IC 74LS02 NOR gate and IC 74LS04 hex

inverter. Pin-out for the 74LS02 are shown in figure 1.

2. Examine figure 2. Analyze its operation carefully. (Remember that the IC 74LS02

is a NOR gate).

3. Study the digital coding system section in the front panel of the trainer. Switch

ON the TRAINER. Use the binary data switches provided on trainer the S1 and S2

as inputs. Observe the outputs of 74LS02 through the LEDs indicated.

X-15B PAGE: 60

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE 1: PIN OUT OF 7404 AND 7402

FIGURE 2: CIRCUIT TO ENCODE INPUTS

4. Set S1 and S2 to 0. Note the output LEDs. Does the LED turns ON, when the

output of the 74LS02 is HIGH or when it is LOW?

5. Turn OFF the trainer.

DISCUSSION:

Your analysis of the circuit in figure 2 revealed that it is wired so that each of

the NOR gates receives a different pair of signals from the input lines because of the

inverters. No NOR gate has the same inputs at the same time.

The truth table of a NOR gate shows that the output is 1 only when both

inputs are 0. When switch S2 and S1 is both neither set to 0, only the first NOR gate

(the one connected to LED 0 -"LO") detects 0 at both its inputs.

7404 7402

X-15B PAGE: 61

LAB ELECTRONICS


LAB ELECTRONICS

Because of the inverters, at least one of the inputs to each of the other gates

is 1, and so the output of each of the other gates is 0. You will find that the LEDS

glow when the signal to the LED is high, or 1. So you know that when an LED glows

it indicates 1 at the output from the NOR gate connected to it. You also know that the

output of the NOR gates associated with the LEDs that do not glow are 0.

STEP-BY-STEP PROCEDURE (CON'T):

6. Now set the logic switches to all possible combinations. Complete the following

chart by filling in which LED glows for each switch combination (1 = LED on, 0 =

LED off).

7.

S2 S1 L0 L1 L2 L3

0 0

0 1

1 0

1 1

8. Examine the chart that you have filled. What does this circuit accomplish?

______________________________________________________________ if

you wanted to transmit the outputs of this circuit to some other location, how

many wires would be needed? ___________________________.

9. This circuit will be used in the next experiment. Turn OFF the trainer.

X-15B PAGE: 62

LAB ELECTRONICS


LAB ELECTRONICS

DISCUSSION:

You should have obtained a truth table like that shown below for the LEDs

associated with this circuit:

S2 S1 L0 L1 L2 L3

0 0 1 0 0 0

0 1 0 1 0 0

1 0 0 0 1 0

1 1 0 0 0 1

This circuit is a 1-of-4 coder. For each possible input, a unique output is

produced in which one bit of the four in the code is a one. You don't see four bits?

Each LED represents a bit of the code; a lighted LED is a 1 bit, an unlit LED is a 0. In

no case, more than one LED is lit for any input condition. To send the electrical

signals that cause the LEDs to light from one place to another would require five

wires: one for each LED signal, and one to serve as a common ground. If there were

8 possible outputs, then nine wires would be needed: one per LED and a ground.

This is an example of parallel data transmission. You can see that the number of

physical wires needed increases dramatically as the number of bits in the code

increases.

SUMMARY:

In step 2, you analyzed the outputs from the circuit shown in figure 2. You

discovered that a separate HIGH (1) output was to be expected from the 74LS02 for

each possible input switch combination. A separate output of "1" occurred when

SW0 and SW1 were switched from 00 to 01 to 10 to 11.

In steps 3 through 6, you verified the accuracy of your analysis and examined

the operation of this circuit. You saw that it did truly function as you expected it to,

with only one LED lighting for each input switch setting combination. You also saw

how parallel data transmission could be applied to send the coded data in parallel.

One wire is needed for each possible code level, plus one wire to serve as a

common ground.

X-15B PAGE: 63

LAB ELECTRONICS


LAB ELECTRONICS

The circuit you built and analyzed is called a one-out-of-four decoder. It

produces a single distinct output for each of four possible unique input conditions.

By extension, you can build "m-out-of-n" decoders to provide unique combinations of

outputs for each input condition.

WIRING DIAGRAM

DIGITAL CODING SYSTEM

X-15B PAGE: 64

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 8

EFFECT OF NOISE & OTHER IMPAIRMENTS

ON DATA TRANSMISSION

OBJECTIVE:

To demonstrate how noise and line imperfections affect the received data

signal on a typical data transmission line, and how these imperfections can be

compensated.

MATERIALS REQUIRED:




INTRODUCTION:

In this trainer, you will study a simulated transmission line using Multiplexer,

Demultiplexer circuit to examine how noise and line distortion affect received data

signal. You will also build a noise generator, and then inject variable noise levels into

the multiplexed signal to observe the effect on outputs. In this way, you will study

how transmission line characteristics affect data transmissions.


1. Study the circuit shown in figure 1.

2. This section consists of the following features;

a. Time generator using IC 74112 J-K flip-flop.

b. Multiplexer using IC 7400 NAND Gates and 7404 Inverter.

c. Demultiplexer using IC 7400.

d. Simulated transmission line using R & C.

e. 74121 Monostable Multivibrator.

X-15B PAGE: 65

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE- 1: MULTIPLEXER WITH SIMULATED TRANSMISSION LINE

3. Patch the circuit as shown in wiring diagram for figure -1.

4. Turn ON the TRAINER and set the clock generator for either 2000 s or 2ms

pulses at a 300 Hz frequency. Adjust the sweep controls to show at least two

repetitions of the pulses on the upper trace. Sketch the upper and lower traces

on the scope graticule provided in figure 2.

How does the introduction of line resistance and capacitance influence the

demultiplexed signal?

X-15B PAGE: 66

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE: 2 - EMPTY GRATICULE FOR SKETCHING SIGNALS AT PIN 8 AND PIN 11

5. Now connect the lower trace to pin 6 of the IC 74LS04 and then to pin 8 of the

74LS00 in the space provided, sketch the signals you find at these two pins.

Pin 6, 74LS04: ____________________________

Pin 8, 74LS00: ____________________________

How does the introduction of line resistance and capacitance influence these

demultiplexed signals?

________________________________________________________________

___________________

6. Connect the lower trace to the top of the 1K emitter resistors of Q1 and Q2.

Observe the effect on the multiplexed signal of the line resistance and

capacitance. While observing the multiplexed signal, vary the amplitude of the

sine wave to observe what effect, changing pulse widths and shapes have. What

are these effects?

________________________________________________________________

________________________________________________________________

______________________________

X-15B PAGE: 67

LAB ELECTRONICS


LAB ELECTRONICS

Vary the frequency generated by the Clock by using the PULSE WIDTH, FREQ., and

FREQ. MULTIPLIER controls. What effect does this have on the distortion?

7. Turn off the TRAINER.

DISCUSSION:

The sketches that you made in figure 2 should approximately resemble to

those shown in figure 4. If your results do not look like these, go back and try again.

Figure 4 shows the signals that you should have found at pin 6 (74LS04) and pin 8

(74LS00). If your results do not match the ones shown in figure 4; recheck your

circuit and repeat that portion of the experiment.

FIGURE - 3: SIGNALS AT PIN 8 (741S04) AND PIN 11 (741S00)

FIGURE - 4: SIGNALS AT PIN 6 (74LS04) AND PIN 8 (74LS00)

X-15B PAGE: 68

LAB ELECTRONICS


LAB ELECTRONICS

As you examined the outputs of both multiplex channels in the presence of a

simulated coaxial cable transmission line, you observed that the received pulses

become rounded and more difficult to pick out as you increase the pulse frequency

and/or decrease the pulse width.

From these results, you probably guessed that there are optimum pulse

widths and speeds for signaling over a given cable. That is true in the general sense.

However, one rarely gets to design terminal equipment for only one configuration of

cabling, so it is generally the cable characteristics that must be changed to

accommodate distortion, and not the characteristics of the terminal equipment.

Once your circuit is functioning, you may want to vary the pulse - width, signal

amplitude, and generator frequency. As you vary these, watch your oscilloscope for

changes. You may also see some interesting results by changing the input signals

from sine wave to square wave. When you are through experimenting, return your

input signal and control switches to their original positions.

STEP-BY-STEP PROCEDURE (CON’T):

8. Modify your circuit as shown in figure 5. This circuit simulates the introduction of

noise at the power frequency onto a real transmission line.

FIGURE: 5 - MULTIPLEXER WITH “NOISE

INJECTOR”

X-15B PAGE: 69

LAB ELECTRONICS


LAB ELECTRONICS

9. Turn on the TRAINER and oscilloscope. Vary the setting of the 1K TRIMMER

and observe the effect of this noise on the pulses at the output (pins 8 and 11 of

the 74LS00) of the demultiplexer and on the multiplex line. What effect does the

noise have?

________________________________________________________________

What is the effect of increase in pulse frequency (vary using the FREQ. selector

and FREQ. MULTIPLIER CONTROLS in the CLOCK GENERATOR)

________________________________________________________________

_________________________of lower pulse repetition frequency?

_______________________________________________________________

10. Turn off the TRAINER.

DISCUSSION:

As you saw, adding noise to the transmission line makes the reception of

good signals more difficult. In fact, as you increased the amplitude of the noise

coupled in by adjusting the 1K-ohm trimmer you noticed the synchronization of the

multiplexer could be destroyed. When that synchronization is lost nothing meaningful

can be detected at the receiver.

As you noted, decreasing the signaling speed and increasing the pulse width

made it easier to extract the signal from the noise. Conversely, with narrow pulses

and fast signaling, detection became more and more difficult. Often the second pulse

in a series simply disappeared from the demultiplexed out.

Clearly, the mixture of noise and transmission line effects causes major

problems for data transmissions. Real lines do have noise and distortion, so this is

not merely a laboratory problem. Methods of dealing with this noise and distortion

effectively and inexpensively (naturally the ideal case) are still being developed.

Let's consider one possibility in the next step.

X-15B PAGE: 70

LAB ELECTRONICS


LAB ELECTRONICS


11. It might be possible to improve the reception of pulses in noise by passing the

received pulse through a one-shot multivibrator circuit. Let's see if this helps.

Look at the IC shown in figure 6. This IC includes not only a one-shot, but also a

Schmitt trigger. As you see, this chip can be used to clean up the effects of

distortion on a data transmission line.

FIGURE: 6 - PIN OUT FOR 74121 IC

12. Connect the 74121 IC into the circuit as shown in figure-7. You should have the

circuit as shown in figure-8. Connect pin 11 of the IC 74LS00 to the one-shot

input at pin 5 of the IC 74121. Set the oscilloscope controls. Connect the

oscilloscope upper trace to pin 11 of the 74LS00, and the lower trace to the

output at pin 6 of the IC 74121.

13. Turn ON the TRAINER and the oscilloscope. Turn the 1K trimmer fully counter

clockwise, so no AC noise is coupled onto the line. Adjust the 100K trimmer in

the pulse squarer section until the lower scope signal looks similar to the upper

trace drawing in figure 3. Compare the input to the one-shot with the "cleaned up"

output at pin 6 of the IC 74121. Does the one-shot help to detect the pulses

accurately? _________________ Is it a major improvement, or a minor one?

_________________. Why do you think it behaves as it does?

X-15B PAGE: 71

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE - 7: 74121 IC IN A CIRCUIT

14. Repeat step 12 by varying the 100K-ohm TRIMMER to add AC noise to the

multiplexed signal.

15. In steps 12 and 13, you observed signals like those shown in figure 9. If you did

not, recheck your circuit and repeat these steps

16. As you discovered in step 13, if you inject enough noise, you lose the

synchronization of the signal coming from the one-shot. The Schmitt trigger

allows the one-shot to cycle dependably - once a given input signal is reached.

The width of the one-shot output pulses depends on the 0.47μF capacitor and the

100K-ohm trimmer. Experiment with different settings of the 1K and 100K-ohm

trimmers to observe the effect of noise and one-shot pulse width on the

recovered signal.

17. Turn OFF the trainer.

X-15B PAGE: 72

LAB ELECTRONICS


LAB ELECTRONICS

FIGURE - 8

NOTES:

PIN 11 signals are badly distorted by line capacitance and resistance. The

width of pulses at pin 6 can be varied with the 100K trimmer.

FIGURE: 9

X-15B PAGE: 73

LAB ELECTRONICS


LAB ELECTRONICS

DISCUSSION:

In comparing the one-shot inputs to the outputs, you should have noticed an

improvement in the performance of the circuit in the presence of noise. This is due to

the fact that, once triggered, the one-shot will cycle and produce a pulse regardless

of what happens on the line while it is cycling. The pulse width is determined-by the

setting of the 100K ohm pot.

In the absence of any noise, you have observed that the one-shot allowed you

to recover clear pulses from a distorted input signal. Remember that the width of

these pulses is fixed by the one-shot circuitry. If pulse width modulation were used,

this approach would be limited.

A one-shot alone does not provide more noise immunity because it is

triggered by the same pulse that you are trying to detect. If the leading edge of that

pulse is distorted, the one-shot's triggering will also suffer. Further improvement

could be achieved in many applications by preceding the one-shot with a Schmitt

trigger, which produces an output pulse only when the input exceeds a specified

threshold value. You saw in step 13 that the Schmitt trigger/one-shot combination

provided a small amount of noise immunity.

Although distortion can make the job more difficult, the nature of digital signals

makes possible their recovery even when they are buried deep in noise. In steps 12

and 13, you saw how the received signal could be cleaned up by applying advanced

detection techniques. That is probably the single most important advantage of digital

signals over analog signals, and it is the reason why digital communication continues

to expand at an increasing rate.

SUMMARY:

In this experiment, you observed how line resistance and capacitance affect

the signals received over a data transmission link. As you varied the speed, width,

and shape of the transmitted pulses, you could see the effect on the fidelity of the

received signals.

You also verified that wide pulses sent at a slow rate are much easier to

detect in the presence of distortion and noise than the narrow pulses sent at a high

rate.

X-15B PAGE: 74

LAB ELECTRONICS


LAB ELECTRONICS

While it appears obviously, this fact runs the counter to what we want to do with data

circuits, namely make them operate fast. For that reason, this "feature" of data

communications is often overlooked.

In step 9, you injected line noise into the transmission path. As you have seen

in previous experiments, this is a common type of noise found on transmission lines.

By changing the signal parameters and the magnitude of the noise, you gained an

understanding of just how noise affects the ability of a receiver to detect a data

signal. You found that these effects can range from the merely bothersome to the

truly destructive.

In steps 11 through 13, you evaluated the usefulness of a one-shot, which

compensates for noise coming from the detected pulses.

As a result of this experiment, you should have gained a better understanding

of how line imperfections and noise affect data signals. You should have become

sufficiently familiar with this subject that you can make basic measurements to

observe qualitatively the fidelity of received base band data signals. You should also

be able to discuss the performance of circuits and equipment in the presence of

noise and distortion, and form preliminary judgments about the suitability of various

equipment interconnections for data communication use.

X-15B PAGE: 75

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM

FOR FIGURE -1

EFFECT OF NOISE & OTHER IMPAIRMENTS ON

DATA TRANSMISSION


TO CRO

X-15B PAGE: 76

LAB ELECTRONICS


LAB ELECTRONICS

EXPERIMENT - 9

ACTIVE BAND PASS FILTER

OBJECTIVES:

To observe the frequency response of Band pass filter that permits a range of

medium frequencies to pass through while rejecting frequencies above & below this

medium. To Calculate "Q" of this band pass filter.

MATERIALS REQUIRED:




INTRODUCTION:

An electric filter is often a frequency-selective circuit that passes a specified

band of frequencies and blocks or attenuates signals of frequencies outside these

band-Filters may be classified in a number of ways:

1. Analogue or digital.

2. Passive or active.

3. Audio (AF) or Radio frequency (RF).

Analogue filters are designed to process Analogue signals. While digital

filters, process analog signals using digital techniques. Depending on the type of

elements used in their construction, filters may be classified as passive or active.

Elements used in passive filters are resistors, capacitors, and inductors. Active

filters, on the other hand, employ transistors or op-amps in addition to the resistors

and capacitors. The type of element used dictates the operating frequency range of

the filter. For example, RC filters are commonly used for audio or low frequency

operation, whereas LC or crystal filters are employed at RF or high frequencies.

Especially because of their high Q value (figure of merit), the crystals provide more

stable operation at higher frequencies.

X-15B PAGE: 77

LAB ELECTRONICS


LAB ELECTRONICS

THEORY:

First, this presents the analysis and design of analog active-RC (audio

frequency) filters using op-amps. In the audio frequencies, inductors are often not

used because they are very large, costly and may dissipate more poor. Inductors

also eliminate magnetic fields. An active filter offers the following advantages over a

passive filter.

1. Gain and frequency adjustment flexibility. Since the op-amp is capable of

providing a gain, the input signal is not attenuated as it is in a passive filter.

Besides that, the active filter is easier to tune to adjust.

2. No loading problem, because of the high input resistance and low output

resistance of the op-amp, the active filter does not cause loading of the source or

load.

3. Cost effectively, active filters are more economical than passive filters. This is

because of the variety of cheaper op-amps and the absence of inductors.

Although active filters are most extensively used in the field of

communications and signal processing, they are employed in one form or another in

almost all sophisticated electronic systems. Radio, television, telephone, radar,

space satellites, and biomedical equipment are but a few systems that employ active

filters.

The most commonly used filters are these:

1. Low-pass filter

2. High-pass filter

3. Band-pass filter

4. Band-reject filter

5. All-pass filter

Each of these filters uses an op-amp as the active element and resistors and

capacitors as the passive elements. Although the IC 741-type op amp works

satisfactorily in these filter circuits, high-speed op-amps such as the IC LM318 or the

ICL8017 improve the filter's performance through their increased slew rate and

higher unity gain bandwidth.

X-15B PAGE: 78

LAB ELECTRONICS


LAB ELECTRONICS

Figure 1(a) shows the frequency response characteristics of five types of

filters. The ideal response is shown by dashed curves, while the solid lines indicate

the practical filter response. A low pass filter has a constant gain from 0 Hz to a high

cutoff frequency fH.

Therefore, the bandwidth is also fH. At fH the gain is down by 3 dB; after that (f

> fH). It decreases with the increase in input frequency. The frequencies between 0

Hz and fH are known as the pass band frequencies, whereas the range of

frequencies, those beyond fH that are attenuated includes the stop band frequencies.

FIGURE -1: FREQUENCY RESPONSE OF THE MAJOR ACTIVE FILTERS.

(A) LOW PASS; (B) HIGH PASS; (C) BAND PASS; (D) BAND REJECT; (E) PHASE

SHIFT BETWEEN INPUT AND OUTPUT VOLTAGE OF AN ALL-PASS FILTER

X-15B PAGE: 79

LAB ELECTRONICS


LAB ELECTRONICS

Figure 1(a) shows the frequency response of the low-pass filter. As indicated

by the dashed line, an ideal filter has a zero loss in its pass band and infinite loss in

its stop band. Unfortunately, ideal filter response is not practical because linear

networks cannot produce the discontinuities. However, it is possible to obtain a

practical response that approximates the ideal response by using special design

techniques as well as precision component values and high-speed op-amps.

Butter worth, Chebyshev, and Cauer filters are some of the most commonly

used practical filters that approximate the ideal response. The key characteristic of

the Butterworth filter is that it has a flat pass band as well as stop band. For this

reason, it is sometimes called a flat-flat filter. The Chebyshev filter has a ripple pass

band and flat stop band, while the Cauer filter has a ripple pass band and a ripple

stop band. Generally, the Cauer filter gives the best stop band response among the

three. Because of their simplicity of design, the low-pass and high-pass Butter worth

filters are discussed.

Figure 1 (b) shows a high-pass filter with a stop band 0 < f < fL and a pass

band f > fL. fL is the low cutoff frequency and f is the operating frequency. A band-

pass filter has a pass band between two cutoff frequencies fH and fL, where fH > fL,

and two stop bands: 0 < f < fL and f > fH. The bandwidth of the band-pass filter,

therefore, is equal to (fH - fL). The band-reject filter performs exactly opposite to the

band-pass- that is it has a band stop between two cutoff frequencies fH and fL and

two pass bands: 0 < f < fL and f > fH. The band-reject is also called a band-stop or

band-elimination filter. The frequency response of band-pass and band-reject filters

is shown in figure 1(c) and (d). In these figures, fC is called the center frequency

since it is approximately at the center of the pass band or stop band.

Figure 1(e) shows the phase shift between input and output voltages of an all-

pass filter. This filter passes all frequencies equally well: that is, output and input

voltages are equal in amplitude for all frequencies; with the phase shift between the

two as a function of frequency.

The highest frequency up to which the input and output amplitudes remain

equal is dependent on the unit gain bandwidth of the op-amp. At this frequency,

however, the phase shift between the input and output is maximum.

X-15B PAGE: 80

LAB ELECTRONICS


LAB ELECTRONICS

THE MULTIPLE-FEEDBACK BAND PASS FILTER:

FIGURE: 2 - BASIC MULTIPLE-FEEDBACK BAND PASS FILTER

As shown in figure 2, the basic multiple-feedback band pass filter is useful for Q's up

to approximately 15-20 with "moderate" gains. For this circuit, the center frequency is

determined from the relation

As a function of the filter's pass band gain G, and Q, the five components are found

from the equations.

and

The filter pass band gain, i.e. the gain at the filter's center frequency is the selection

of the five component values eased by making C2 and C4 equal.

X-15B PAGE: 81

LAB ELECTRONICS


LAB ELECTRONICS

So that

Where,

One nice feature of this circuit is that the center frequency can be shifted to a

new frequency ωo, while the pass band gain and bandwidth remain constant, simply

by changing resistor R3 to a new value R3΄, so that

On the other hand, because of the denominator of equation, we are restricted to the

condition

Normally we select a convenient value for C2 and C4 and then calculate the values

for the three resistors based on the required values for Q, G, and ωo.

Without any derivations, the amplitude response of a band pass filter using a single

op amp is given by

This is graphed in figure-3. The response is a maximum of 0 dB at the center

frequency (normalized at 1.0) and then drops off on both sides. How fast the

response drops is dependent on Q. However, all the curves eventually start to

straighten out so that all the curves appear to be parallel. At the extremes, the rolloff

of all the curves approaches 6 dB/octave, or 20 dB/decade, regardless of the

X-15B PAGE: 82

LAB ELECTRONICS


LAB ELECTRONICS

filter's Q. When the response of band pass filter approaches a rolled of 6 dB/octave,

it is termed a second order 1 pole band pass filter.

The sharpness of the band pass filter's response in the vicinity of its center

frequency depends on its Q, which in turn depends on its 3-dB bandwidth.

The 3-dB bandwidth is the difference between the upper and lower frequencies

where the amplitude response is 3 dB less than the response at the center

frequency, equation becomes

FIGURE - 3: AMPLITUDE RESPONSE OF BAND PASS FILTER USING A SINGLE OP

AMP

X-15B PAGE: 83

LAB ELECTRONICS


LAB ELECTRONICS


1. Study the circuit diagram given for band pass filter on the front panel of the

trainer.

2. Switch on the trainer.

3. Connect a sine wave oscillator from the AF sine output at the input and adjust the

amplitude to 2V p-p and frequency to 200 Hz.

4. Measure the output voltage (v0).

5. Vary the frequency of the generator in steps of 100Hz and note the output

voltage at each step.

6. Measure the frequency (fC) at which the amplitude is maximum. Now determine

the upper and lower 3dB frequencies (fH & fL) by measuring the frequencies at

which the amplitude response drops by 0.707 times the maximum voltage gain

which is the 3dB decrease.

7. Calculate the following factors - Frequency Gain, Upper 3dB Frequency, Lower

3dB Frequency, Bandwidth, Center Frequency

fO

Quality Factor (Q) = ----------

fH - fL

8. Plot the graph between output voltage versus frequency.

X-15B PAGE: 84

LAB ELECTRONICS


LAB ELECTRONICS

WIRING DIAGRAM

ACTIVE BAND PASS FILTER


LAB ELECTRONICS

Documents

Transcript of LAB ELECTRONICS