Communication systems 1. DIGITAL COMMUNICATION SYSTEMS

P. Rama Krishna reddy

PAGE 1

INTRODUCTION

Need for digital communication:

− To reduce the noise interference since in the analog the interference of noise is very high.

− To increase the security. It is achieved by coding, since coding is not possible in analog

communication.

− To achieve construction easily and to achieve more processing techniques easily like

multiplexing.

Advantages:

− The main advantage of the digital communication over the analog communication is

noise immunity.

− Digital signals are better for processing and multiplexing techniques than analog signals.

− Digital signals are easy simpler to measure and evaluate.

− In digital communication the error in signals can detect and correct easily.

− The capacity of system is increased.

− Snr is very low

− High security and low cost.

− Easy to encrypt and decrypt the signals than the analog signals.

Disadvantages:

High band width requirement

Requires expensive synchronizing clock circuits at the receiver for reproducing the

original

Long distance communication

PAGE 2

Block diagram of digital communication:

INFORMATION SOURCE:

This block consists of two sub blocks.

1. Sampler

2. Quantization

In this two blocks the analog signals is converted into discrete signals.

SAMPLER:

In this the block the analog signal is converted into samples

QUANTIZATION:

It increases or decreases the sample point decimal values to its nearest integer value.

SOURCE ENCODER:

It converts the sampled integer value in digital bits. (I.e. it is an decimal to binary converter)

CHANNEL ENCODER:

In this block some redundant bits are added to digital data i.e. parity bits, stop or start bits.

MODULATOR:

it is the process in which some characteristics of carrier wave is varied in accordance with

instantaneous amplitude of the message signal.

There are three types of modulation. They are:

AMPLITUDE SHIFT KEYING

PHASE SHIFT KEYING

FREQUENCY SHIFT KEYING

DEMODULATOR:

It demodulates the modulated signal from channel and produce digital data.

PAGE 3

CHANNEL DECODER:

It detects output of the decoder and if necessary the data is corrected.

In this the added redundant bits are removed.

SOURCE ENCODER:

It converts the binary data to analog data through binary decoder and low pass filter.

TYPES OF CHANNEL:

There are two types of channels:

GUIDED CHANNEL:

Copper wire: In this the current signal is transmitted, and the band width is very

low.

Twisted pair: voice transmission is achieved. It have higher band width than

copper wire.

Co axial cable: Video and audio both are transmitted. Band width is very high

and noise interference is also very high.

Optical fiber: all types of data transfer is possible, high speed, low interference

of noise.

UNGUIDED CHANNEL:

Free space: low cost, high band width, high noise , long distance.

Digital modulation:

It is also knows as digital modulation. In this modulation the carrier signal is a pulse signal.

In this modulation the amplitude, duration and position of carrier is varied in accordance with the

instantaneous amplitude of message signal.

This method is mainly achieved by sampling process.

SAMPLING:

The process of converting an analog signal into discrete signal is known as sampling.

The time interval between two subsequent samples is known as the sampling interval.

PAGE 4

SAMPLING THEOREM:

The process of sampling is achieved by convoluting message signal with the train of impulses

with a fixed interval of time.

Depends on the sampling time the reconstruction of signal is achieved. Depending on the

sampling interval we have three cases. They are:

OVER SAMPLING:

If the sampling rate is more than twice of the message signal frequency then it is known as over

sampling.

Fs < 2fm.

UNDER SAMPLING:

If the sampling is less than twice of the message signal frequency is known as under sampling.

In this sampling during the reconstruction the signal is over lapped and the message is lost.

Such effect is known as “ALIASING EFFECT”

PAGE 5

Fs>2fm.

PERFECT SAMPLING:

If the sampling rate is exactly equal to twice of the message signal frequency then it is known as

perfect sampling.

Fs = 2fm.

In this sampling rate is also called Nyquist rate. The minimum rate at which a signal can be

reconstructed from its samples is known as Nyquist rate.

Types of sampling: they are,

1. Instantaneous sampling.

2. Natural sampling.

3. Flat top sampling

Instantaneous sampling:

Ideal Sampling is also known as instantaneous sampling or Impulse Sampling. Train of

impulse is used as a carrier signal for ideal sampling. In this sampling technique the sampling

function is a train of impulses and the principle used is known as multiplication principle

Natural sampling:

Natural Sampling is a practical method of sampling in which pulse have finite width equal to

τ. Sampling is done in accordance with the carrier signal which is digital in nature.

PAGE 6

− Natural Sampled Waveform

− Functional Diagram of Natural Sampler

With the help of functional diagram of a Natural sampler, a sampled signal g(t) is obtained by

multiplication of sampling function c(t) and the input signal x(t).

Spectrum of Natural Sampled Signal is given by:

G(f) = Aτ/ Ts .[ Σ sin c(n fs.τ) X(f-n fs)]

Flat Top Sampling:

Flat top sampling is like natural sampling i.e; practical in nature. In comparison to natural

sampling flat top sampling can be easily obtained. In this sampling techniques, the top of the

samples remains constant and is equal to the instantaneous value of the message signal x(t) at the

start of sampling process. Sample and hold circuit are used in this type of sampling.

Spectrum of Flat top Sampled Signal is given by: G(f) = fs .[ Σ X(f-n fs). H(f)]

PAGE 7

Comparison of types of sampling

Based on types of technique

used the amplitude modulation

is divided into different models. They are:

PAGE 8

PULSE AMPLITUDE MODULATION:

The pulse amplitude modulation is same as flat sampling.

Pulse amplitude modulation is the basic form of pulse modulation in which the signal is sampled

at regular and each sample is made proportional to the amplitude of the modulating signal at the

sampling instant.

The generation of PAM signal from the sampler which has two inputs i.e. modulating signal and

sampling signal or carrier pulse.

Generation of PAM signal:

Thus the amplitude of the signal is

proportional to the modulating signal

through which information is carried.

This is Pulse amplitude modulation

signal.

The figure shows the spectrum of pulse

amplitude modulated signal along with

the message signal and the sampling

signal which is the carrier train of pulses with the help of the waveform plotted in

time domain.

Pulse Modulation may be used to transmitting analog information, such as

continuous speech signal or data.

PAGE 9

DEMODULATION OF PAM SIGNAL:

VPAM(t)

M(t)

PAM detector

The hold circuit holds the signal peak amplitudes (i.e by using capacitor the peak values

are hold and is given to a low pass filter.

The low pass filter eliminates high frequency ripples and generates the demodulated

signal which has its amplitude proportional to PAM signal at all time instant.

The square pulse in frequency domain is represented as sync function as shown above

when the signal is passed through a L.P.F it allows only top peak (i.e it produces an

analog signal)

This signal is then applied to an inverting amplifier to amplify its signal level to have the

demodulated output with almost equal amplitude with the modulating signal.

Hold circuit L.P.F

PAGE 10

TIME DIVISION MULTIPLEXING:

The above figure shows the time division multiplexing. In time division multiplexing the

entire band of time is divided to equal slots for n number of channels the complete

channel bandwidth is allotted to one user for a fixed time slot.

As an example, if there are ten users, then every user can be given the time slot of one

second.

Thus, complete channel can be used by each user for one second time in every ten

seconds.

This technique is suitable for digital signals. Because digital signals are transmitted

intermittently and the time spacing between two successive digital code words can be

utilized for other signals.

These are extensively used in telephony.

Working:

In PAM every sample is converted into digital 8 bits (i.e. baud rate is 8 bits/sec) and from

every device it is connected to a multiplexer or rotatory switch.

The output of multiplexer is ‘n’ channels for entire band width (i.e. at one slot one device

is connected), so the output has n*fs number of pulses per sec.

TDM output consists of frames, each of which contains bit representing one sample from

‘n’ channels.one extra bit called frame bit is added for synchronization at the receiver.

Each frame consists of n*fs +1

PAGE 11

Band width = nfs/2

Fs=2fm

B.W = nfm

Pulse width demodulation:

Generation of PWM

PWM signal can be generated by using a comparator, where modulating signal and saw

tooth signal are given as the input of the comparator. It is the simplest method for PWM

generation.

The PWM generation is explained with the help of circuit given below.

PWM generation by a comparator

As shown in the figure, one input of the comparator is fed by the input message or

modulating signal and the other input by a sawtooth signal which operates at carrier

frequency

PAGE 12

o .

Considering both ±ve sides, the maximum of the input signal should be less than that of

sawtooth signal.

The comparator will compare the two signals together to generate the PWM signal at its

output as shown in the wave forms.

The rising edges of the PWM signal coincides with the falling edge of the sawtooth

signal.

When the sawtooth signal is at the minimum value which is less than the minimum of the

input signal, then the positive input of the comparator is at higher potential which gives

When the sawtooth signal rises and is at the maximum value, the negative input of the

comparator is at higher potential, which will produce the comparator output to be

negative.

Thus the input signal magnitude determines the comparator output and its potential,

which then decides the width of the pulse generated at the output.

In other words we can say that the width of the pulse generated signal is directly

proportional to the amplitude of the modulating signal.

PAGE 13

Demodulation of PWM:

For PWM demodulation, put a

ramp at the +ve edge which will

stop at the arrival of –ve egde.

The ramp will attain different

heights in each cycle since the

widths are different and the

heights attained are directly

proportional to the pulse width

and in turn the amplitude of the

message signal.

This is then passed through a low

pass filter where it will follow

envelop i.e. the message signal,

which produces the demodulated

signal at the output.

Generation of PPM

PPM signal can be generated with the help of PWM as shown in Fig7 below.

PAGE 14

PPM

generation from PWM

The PWM signal generated above is sent to an inverter which reverses the polarity of the

pulses.

This is then followed by a differentiator which generates +ve spikes for PWM signal

going from High to Low and -ve spikes for Low to High transistion. The spikes generated

are shown in the fourth waveform of Fig8.

These spikes are then fed to the positive edge triggered pulse generator which generates

fixed width pulses when a +ve spike appears, coinciding with the falling edge of the

PWM signal.

Thus PPM signal is generated at the output which is shown in the waveforms, where

pulse position carry the message information.

Demodulation of PPM :

In PPM demodulation, ramp is used which starts at the +ve edge of the one pulse and

stops at the +ve edge of the next pulse.

Thus the height of the generated ramp is determined by the delay between the pulses

which indirectly follows the amplitude of the modulating signal.

This is then passed through a low pass filter which filters the envelop information as the

demodulated signal.

PAGE 15

PULSE CODE MODULATION:

The process of converting an analog signal into a PCM signal is called coding and the inverse

operation is known as decoding.

The function of each block is shown in below figure:

Low pass filter:

1. The main function of low pass circuit used in generation of PCM is to prevent aliasing.

Sample and Hold circuit:

2. The message signal is sampled using the sample and hold circuit (i.e. the analog signal is

converted into pam signal). For perfect reconstruction of the signal the sampling

frequency should be greater than or equal to the message signal frequency.

Quantizer:

3. The process of assigning voltage levels to the sample is called Quantization. In this

process the float values are rounded to the near integer values. N.O of levels for a sample

is depends on the no umber of bit per sample.

N =2m N = number of levels: m = no of bits per sample:

Due to this process noise occurs called Quantization noise.

PAGE 16

Encoder:

It is process in which the sample voltages are converted into the digital bits. If there are n

quantization levels then there are logN bits to represent voltage level. It is achieved by A/D

converter.

Parallel to serial converter:

The output of the encoder is in parallel form, so it is converted into serial format by using universal

register.

Demodulation of PCM signal:

− The received data is in serial format is converted into parallel by buffer register and is

given to a decoder.

− The decoder (D/A converter) converts the digital bits into the voltage levels which are

converted into an analog by passing it through L.P.F.

Advantages of PCM:

1. Ruggedness to transmit noise and interference.

2. Efficient a regeneration of coded signals along the transmission medium.

3. The possibility of uniform format for different types of message signals.

4. Easy of multiplexing

5. Easy of encryption

DISADVANTAGES:

1. Large band width

2. There should be perfect synchronization.

TYPES OF QUANTIZATION AND ITS NOISE:

The quantization are two types. They are:

1. Uniform quantization:

The quantization in which the step level of the signal is constant throughout signal is

known as uniform quantization

2. Non uniform quantization:

The quantization in which the step is varied with respect to the input of the message

signal is known as non-uniform quantization.

QUANTIZATION NOISE:

The difference between the quantized output and input signal is known as quantization nose.

Error = Xq(nts)-X(nts);

PAGE 17

The uniform quantization is again divide into three types.

They are:

1. Mid thread quantization

2. Mid riser quantization

3. Biased quantization

Mid quantization:

The quantization in which the output at the zero axis is zero is known as mid thread

quantization.

The quantization in which the output at the zero axis is non zero is known as mid riser

quantization.

In both cases the quantization levels at zero axis varies with /2 toward on either axis.

Biased

quantization

BIASED QUANTIZATION:

In biased quantization the quantization level is zero at the zero axis as shown in figure.

EXPRESSION FOR QUANTIZATION NOISE AND SNR IN PCM:-

Let Q = Random Variable denotes the Quantization error

q = Sampled value of Q

PAGE 18

Assuming that the random variable Q is uniformly distributed over the possible range

In decibels the

signal to noise ratio when the input is non sinusoidal:

Signal power = Vmax2

Noise power = 2/12 = 4Vmax2 / q/12 = Vmax

2 /3q { = 2Vmax/q}

S/N ratio = 3* q = q = 3*2v

PAGE 19

In db = 10(log3*2^v2)

= 4.667 + 6v db

When the signal is analog signal then the signal maximum value is considered as r.m.s

value

Signal power = Vmax2/2

Noise power = 2/12 = 4Vmax2 / q/12 = Vmax

2 /3q { = 2Vmax/q}

S/N ratio = 3* q = q = 3*2v /2

In db = 10(log3*2^v/2 2)

= 1.76+6v db