UNIT 1: BASICS OF DIGITAL COMMUNICATION

1.1. INTRODUCTION
1.1.1. BASIC DEFINATIONS
 COMMUNICATION SYSTEM:
The system which is used to transmit information bearing signal from one
point (source) to another (destination).
 SOURCE:
The system which generates information is called source.
 The information which is coming out of source may be in the form of
analog or digital in nature.
 Practically almost all sources generate analog signals.
 Examples for information are: voice, music, television picture,
teletype data, atmosphere temperature and pressure & soon.
 The above signals can be converted into electrical signals for
transmission through channel.
 The signals can be of three forms:
i. Analog: these are continuous in amplitude and time.
ii. Discrete: These are continuous in amplitude and discrete in
time.
iii. Digital: These are discrete in both amplitude and time.

 During last two decades Digital Communication techniques has

attained utmost predominance, because of the facts below.
I. Flexibility and compatibility with the digital formats
II. Improved reliability
III. Availability of solid state device technology

 SYMBOL RATE:
The rate at which the information source generates source alphabets.
It is expressed in symbols/sec.
 INFORMATION RATE:
The information rate represents the minimum average data rate required to
transmit information from source to destination.
Therefore Information rate = (Symbol rate ) X ( Source entropy).
This is expressed in bits/sec.
Where source entropy is the average information content per symbol
[bits/symbol]

1.1.2. MERITS AND DEMERITS OF DIGITAL COMMUNICATION:

MERITS:
I. Speed, size and cost:
Because of advances in digital IC technologies, the speed is increased
With reduced size and hence cost.
II. Security:
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 Data encryption techniques are very efficient to allow only a particular
receiver to detect the transmitted data.
III. Multiplexing:
Different type of signals can be transmitted over a single channel with good
synchronization.
Ex: speech, video and other data can be transmitted in a single channel.
IV. Noise
Accumulation of noise is less since the signal is discrete in both amplitude
and time; hence data can be transmitted for a long distance with less number
of repeaters.
V. Error detection:
Error detection and correction is easy at the receiver because efficient
encoding techniques can be used during transmission.

DEMERITS:

I. Requires more transmission bandwidth

II. Need good synchronization

NOTE:

Because of all the above advantages of digital communication, the analog

signal is converted into digital for effective transmission, to perform this the
below steps has to be done:

1) Sampling.
2) Quantization and
3) Encoding.

1.2. BASIC SIGNAL PROCEESING OPERATIONS IN DIGITAL

COMMUNICATION:

Fig1.1: block diagram of digital communication system.

Let us consider a digital communication system as shown fig1.1. the basic signal
operations performed here are source coding channel coding and modulation.

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 Source:

 Assume the signal generated by the source is a digital or an analog signal is converted
into digital.

Source encoder / decoder:

 The digital data from the source is converted into another form of digital data. This
mapping is one-to-one, therefore at the decoder reverse mapping is done to reconstruct
the original message.
 In the source encoding, the redundant bits are eliminated, therefore reducing the
bandwidth requirement.

Channel encoding/ Decoding:

 Channel encoder / decoder is used to map the source encoder output with the channel
input and channel output to source decoder input respectively.
 In this process some controlled redundancy is added at the encoder, which helps to
detect the errors at the receiver.
 This is necessary because error may occur due to addition of noise in the channel.
Therefore in source encoding redundancy is removed & in channel coding controlled
redundancy is added.
Modulator/ Detector:

 To provide an effective transmission over a noisy channel modulation is required.

 Digital modulators perform keying shifts in amplitude, frequency & phase of the
carrier signal, which results in following modulation methods:
1) Amplitude shift keying.
2) Frequency shift keying.
3) Phase shift keying.
 Detector performs the demodulation to produce channel encoder output.
 Combination of channel, modulation & detector are called as discrete channel,
therefore input and output at each stage are digital.

1.3. Sampling:

 To convert an analog signal into digital data, sampling, quantization and encoding has
to be performed.
 An analog signal is first sampled, which can be defined as “the process of converting
an analog signal into discrete time signal by measuring the signals at periodic instant
of time”.
 The sampling rate must be chosen properly, so that the discrete time signals obtained,
uniquely defines the continuous-time signals.

1.3.1 Sampling theorem:

Statement:

“If a finite energy signal g(t) contains no frequencies higher than ‘ W ‘ hertz, it
can be completely recovered from its ordinates at a sequence points spaced ‘1/2W’
seconds apart”.

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Proof:

The sampling theorem for band limited signal can be proved in two parts:
1) Representing g (t) in terms of its samples.
2) Reconstructing it from its samples.

To prove this, let us consider an analog signal g (t), of finite energy and infinite
duration as shown in fig1.2 (a).

Representing g (t) in terms of its samples.

Let us consider the sample values of g(t) at t=0,±Ts,±2Ts, ±3Ts…….. Therefore

the entire series can be denoted as g (nTs), where n= 0, ±1, ±2, ±3 ……
Where, Ts = sampling Period.
1
fs  = sampling rate.
Ts
To obtain gδ(t) from g(t), we need to multiply g(t) with δ function


g  (t )   g (nTs) (t  nTs)
n  
--------------- >(1)

Where g(nTs) is the samples of g(t)

δ (t-nTs) indicates the samples placed at ±Ts, ±2Ts, ±3Ts, ±4Ts………….& so on.
Eq (1) can be graphically represented as shown in fig1.2 (b)
Since g(nTs) are samples of g(t)
g (t ) (t  nTs )  g (nTs ) (t  nTs ) ……………..(2)
Therefore, Eq (1) can be written as :

g  (t )  g (t )   (t  nTs )
n  

Let δ(t-nTs) is represented as δTs (t)

g (t )  g (t )   Ts (t) ----------------------------- >(3)

Where,  Ts (t) = ideal sampling function.

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 If we consider g (t) is message signal &  Ts (t) carrier signal then gδ (t) is the
modulated wave as shown below

We know that multiplication of two time function is equivalent to the convolution of their
respective Fourier transforms.
By definition g(t) ------------------> G(f)

gδ(t) ----------------- > Gδ(f)



δ(t − nTs) ------------------ > fs   ( f  nf )

n  
s

Therefore , G ( f )  F .T{g (t )}  F .T{ (t - nTs )}


G ( f )  G ( f )  f s   (f - nf
n  
s )

G ( f )  f s  G ( f )   (f - mfs)
m 
------------- > (4)

From the properties of the delta function convolution of G(f) and δ(t − nTs) is G(f-nfs).
Therefore eq(4) can be rewritten as


G ( f )  f s  G (f - nf
n  
s ) ---------------------- > (5)

Therefore from the above equation it is understood that G (f) is placed at ±fs, ±2fs, ±3fs,
±4fs………. & hence G (f) is periodic in fs.

The spectrum of the above expression assuming fs=2W is as shown in fig1.3

And we also know that, δ(t − nTs)------------- >exp (-j2πnfTs)
Taking F.T of eq(1) becomes

G ( f )   g(nTs) exp (-j2nfT
n  
s ) ---------- >(6)

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Reconstruction of signal from its samples.

Let us consider fs=2w, Ts= (1/2w).

Therefore equa (6) implies

n  jnf
G ( f )   g ( 2w ) exp(
n   w
) ------------->(7)
Substituting fs=2w in equ (5)
Gδ (f) =fs G (f)
Gδ (f) =2w G (f)
1
Therefore G( f )  G ( f ) ----------> (8) -w<f<w
2w
Substituting Eq (7) in Eq (8)
1  n  jnf
G( f )   g ( ) exp(
2w n 2w w
) ------------>(9) -w<f<w

Taking the inverse Fourier Transform of G (f)


g (t ) 

 G( f ) exp( j 2ft)df

1  n  jnf
g (t )  
2 w

n  
g (
2 w
) exp(
w
) exp( j 2ft)df


Interchanging the summation and integration.


n 1  n
g (t )   g(
n  
) 
2w 2w 
exp[ j 2f (t 
2w
)]


n 1 e j 2f (t nTs ) w
g (t )   g ( ) [ ] w
n   2w 2w j 2 (t  nTs)

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 n 1 e j 2w(t nTs )  e  j 2w(t nTs )

g (t )   g ( ) [ ]
n 2w 2w j 2 (t  nTs)

n 1 sin 2w(t  nTs)
g (t )   g ( 2w ) 2w [
n    (t  nTs)
]


n sin 2w(t  nTs)
g (t )   g ( 2w )[ (2wt  2wnTs) ]
n  


n sin  (2wt  n)
g (t )   g ( 2w )[
n    (2wt  n)
]


n
g (t )   g ( 2w ) sin c(2wt  n)
n  
---------->(10)

Equation 10 is called an interpolation formula for reconstructing the original signal g (t)
Fig.1.3(c) shows the response of the reconstruction filter.

1.3.2 QUADRATURE SAMPLING OF BANDPASS SIGNALS:

Practically signals are band pass in nature, in order to perform sampling for a band pass
signals, we use quadrature sampling.
Let the band pass signals are represented with its In phase and Quadrature components
separately which are represented as gI(t) and gQ(t) respectively, and each are sampled
separately.
Consider a band pass signal g(t) centered at frequency fc as shown in fig 1.4(a). The band
pass signal g(t) is expressed in terms of gI(t) and gQ(t) as follows:

W.K.T the complex envelope g(t) of band pass signal ğ(t) are related as

g(t)=Re[ğ(t)exp(j2πfct)] ----------->(1)
Since ğ(t) is complex, ğ(t) = gI(t)+jgQ(t) ------------->(2)
And exp (j2πfct) = cos (2πfct) +jsin (2πfct) -----------> (3)
Substituting equ (2) and equ (3) in equ (1)
g(t) = gI(t)cos(2πfct) – gQ(t)sin(2πfct) ------------>(4)
gI(t) and gQ(t) are obtained by multiplying cos and sin functions to band pass g(t) as shown
in fig.1.5(a) . To reconstruct g(t) from these components gI(t) and gQ(t) are multiplied by
cos and sin functions as shown in fig.1.5(b).

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Fig1.5(a): Generation of gI(t) and gQ(t)

Fig1.5 (b): Reconstruction of signal

1.4 PRACTICAL ASPECTS OF SAMPLING & SIGNAL RECOVERY:

In the previous sections, we studied ideal sampling in which the samples are impulses.
Practically, we use high speed switching circuits, which have finite duration pulses rather
than impulses.

1.4.1 ORDINARY SAMPLES OF FINITE DURATION:

This type of sampling is also called natural sampling or chopper sampling. Here the pulses
have finite duration ‘T’. It is called chopper sampling because the waveform of the
sampled signal appears to be chopped off from the original signal waveform.

Let us consider an analog signal g(t) to be applied to a switching circuit controlled by

sampling function c(t) that consists of an infinite succession of rectangular pulses of
amplitude A, duration T, with period Ts. When C(t) goes high, a switch is closed.
Therefore S(t)= g(t) and hen C(t) goes low output is zero.

Let g(t), c(t) and s(t) be the analog i/p, sampling function and o/p signals respectively as
shown in fig.1.6.

Fig1.6: Natural SAmpler

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If fs is the sampling frequency= (1/Ts), mathematically the sampled signal s(t) can be
written as
s (t) = c(t)g(t)………….(1)

Since c(t) is a periodic rectangular pulse train


C (t )  C
n  
n exp( j 2f s nt )................(2)

TA
Where, Cn  sin c( f nT )...................(3)
Ts
Where, T = pulse duration
A = amplitude of pulses
fn = nfs

Substituting equ (3) in equ (2)


C (t )   TAf
n  
s sin c(nf s T ) exp( j 2f s nt )................(4)
Substituting equ (4) in equ (1)

S (t )  f s TA  sin c(nf s T ) exp( j 2f s nt ) g (t )..................(5)
n  
Taking F.T of equ (5)

S ( f )  f s TA  sin c(nf s T )F .T {exp( j 2f s nt )  g (t )}
n  
W.K.T exp (j2πfsnt)g(t)……………..>G(f-nfs)
Therefore

S ( f )  f s TA  sin c(nf s T )G ( f  nf s )..................(6)
n  

Since G(f) is periodic in fs, the spectrum of S(f)is as shown in fig.1.7.

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 Fig1.7: Spectrum of s (f)

1.4.2 FLAT TOP SAMPLING:

In this type of sampling the top of the samples remain constant and equal to instantaneous
value of baseband signal g(t) at the start of sampling.
Let the duration of the sample is T and sampling rate is equal to fs = (1/Ts). Let h (t) be
the rectangular pulse of unit amplitude and duration T, as shown in fig.1.8

Therefore the shifted version of h(t) is to be multiplied with basedband signal to get the
samples

s(t )   g (nT )h(t  nT ) -------------> (1)
n  
s s

From fig 1.8

h (t) = { 1 0 < t <T
{ 0 t<0, t>T ------------------ > (2)

= rect ( T  2 ) ----------------- > (3)

t 1

We know that from instantaneous sampling


g (t )   g (nT ) (t  nT ) ------ > (4)
n  
s s

Convolving gδ(t) with h(t) we get


g  (t )  h(t ) 

 g (t )h(t   )d
 
g  (t )  h(t )    g (nT ) (t  nT )h(t   )d
n  
s s

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  
  g (nT )   (t  nT )h(t   )d --------- > (5)
n  
s s



g  (t )  h(t )   g (nT )h(t  nT ) ---------------- > (6)
n  
s s

From equ (1) and equ (6)

s(t )  g  (t )  h(t ) -------------- > (7)
Taking Fourier transform of equ (7)
S ( f )  G ( f ).G( f ) --------- > (8)


We know that, G ( f )  f s  G (f - nf
n  
s )


S( f )  fs  G(f - nf
n  
s )H(f)

Fig: 1.8 Flat top sampling

1.4.3 Aperture Effect:

The signal S (f) is obtained by G (f) passed through a filter having transfer function H(f).
From equ (3) h (t) is a rectangular pulse .Therefore signal has to be passed through this
filter whose spectrum is given by
H ( f )  Sinc ( f ) exp(  jf )
Therefore the spectrum of H (f) can be represented as shown in fig.1.8(b)
Thus by using flat top samples an amplitude distortion is introduced in reconstruction of
g(t) from S(t).the high frequency roll off of H (f) acts like a low pass filter & attenuates
upper portion of message signal. This effect is called aperture effect.

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 Fig.1.8: (a) one pulse of rectangular pulse train.
(b) spectrum of the pulse train of (a)

1.4.3.1 Compensation for aperture effect:

This distortion can be corrected by connecting an equalizer in cascade with the low pass
reconstruction filter as shown in fig.1.9. The equalizer has the effect of decreasing the in-
band loss of the reconstruction filter as frequency increases. The response of equalizer is
given by
k . exp(  jf )
H eq ( f )  H( f )
 Sinc1( fT )

Fig1.9: Compensation of Aperture effect

1.5. PRACTICAL SAMPLE & HOLD CIRCUIT:

Practical method of obtaining flat top samples from a continuous time signal g (t) is
known as sample & hold circuit. A sample and hold circuit is depicted in fig .1.10(a) & it
make use of 2 FET switches & a capacitor connected across second FET transistor. A
pulse applied at G1 closes the sampling switch & capacitor holds the sampled voltage
until discharged by a pulse applied at G2.

The time domain expression for flat top samples is


v(t )   g (nT )h(t  nT )
n  
s s
-------------- > (1)
Where h (t) = { 1 0< t < Ts

{0 otherwise

Flat top sampling is obtained by convoluting the output of ideal samples by a rectangular
pulse h(t) as shown in fig1.10 (b)
i.e . v(t )  g  (t )  h(t )
Taking FT on both side we get
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 V ( f )  G ( f ).H ( f )

V ( f )  fs  G(f - nf
m  
s )H(f)
------------ > (2)
Where
H ( f )  Sinc ( f ) exp(  jf )
H (f) acts like a LPF & attenuates the spectral components located at integer multiples of
fs
However, it also attenuates the HF components of the massage spectrum centered at f=0,
known as aperture effect. Fig.1.10(c) shows the reconstruction process. The
reconstruction filter [LPF] removes unwanted frequencies component present at integer
multiples of f s . An equalizer is cascaded with the reconstruction filter to compensate
for aperture effect. It may be notated that the amplitude frequency response of the
1
equalizer is H ( f ) .

Fig1.10 (a)

Fig.1.10 (b)

Fig.1.10(c)

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