DIGITAL COMMUNICATION

Reference Book :

1. Digital and Analog communication system by B.P.Lathi

2. Principle of communication system by Taub & Schilling (TMH)

 Formatting A Base Band Modulation
(Sampling and pulse code modulation-ch.6)

1. Base Band System, The Sampling Theorem,

Aliasing
2. Pulse Code Modulation, Quantization, Quantization
Error In PCM, Non- Uniform Quantization,
Companding, DPCM, ADPCM
3. Delta Modulation, Adaptive Delta Modulation
 PURPOSE OF COMMUNICATION
Communication is all about transmitting and
receiving signal, analog or digital from source
to destination through channel.

Digital communication deals with transmission

and reception of signal using digital
techniques.
 Electronic
communication
system

Analog Digital
communication communication

Continuous Pulse
PCM DM ADM
wave systems modulation

AM FM PM PAM PWM PPM

 Digital Communication

The modulation system in which the

transmitted signal is in the form of constant
amplitude, phase and frequency is called
digital modulation system.
Example:PCM,DM
Block diagram of Digital Communication

Noise
+
Source:

Digital communication- data is transmitted in

digital form
Two types of sources:
Digital source
Binary source: sequence of bits e.g. binary file
Others: English text-can be converted to binary using ASCII values
Analog source
Examples: Audio signal captured from microphone, video signal
captured in a video camera.
Can be converted to digital form by A/D converter involves
sampling and quantization by source encoder.
 Advantages of Digital communication

Better noise immunity

It is possible to detect and correct errors
Repeaters can be used
Possible to use advanced data processing
techniques such as image processing and
digital signal processing
Simpler and cheaper as compared to analog
due to invention of high speed computers and
integrated circuits
 Drawbacks of Digital Communication

Bit rate is high so requires larger channel bandwidth

Needs synchronization
 Applications of Digital Communication

Long distance communication between earth

and space ships
Satellite communication
Telephone systems
Data and computer communications
Signal and system

In time domain signals are represented in the

form of waveform as a function of time,
whereas in frequency domain, signals are
represented by their amplitude or power
spectrum.
The two representation are related through
Fourier transform.
Fourier transform

Deals with nonperiodic signals.

E.g. speech, music, and video.

F.T. of rectangular pulse will be sinc function.
Frequency shifting property

e.g. X(t)cosw0t=1/2(X(W-W0)+X(W+W0))
 Fourier series
Deals with periodic signals.
All the periodic signals will be denoted by tilde(~)over the
variable.

An impulse is a or delta function, is an important signal used

in communication and is defined as (t).
(t)=1 for t=0
=0 otherwise
Impulse train is given by

An impulse train is a periodic sequence and also written in

form of fourier series as,
Discrete time Fourier transform(DTFT)

Deals with the sampled analog signals.

It is a periodic, with period of 2.
Baseband(Low pass) and Band pass signals
Base band signal is one which is not modulated.
All the voice, data and picture signals are called
as baseband signals.
It generally occupies frequency spectrum right
form 0 Hz.
Band pass signals are signal which has a non
zero lowest frequency in its spectrum.
This means that frequency spectrum of a band
pass signal extends from f1 to f2 Hz.
The modulated signal is called band pass signals.
 Applications of A-D converters
Music recording
People often produce music on computers using an analog recording and
therefore need analog-to-digital converters to create the pulse-code
modulation (PCM) data streams that go onto compact discs and digital music
files.
Radar systems
commonly use analog-to-digital converters to convert signal strength to digital
values for subsequent signal processing
Digital imaging systems
commonly use analog-to-digital converters in digitizing pixels.
Sensors
Sensors produce an analog signal; temperature, pressure, pH, light intensity etc.
All these signals can be amplified and fed to an ADC to produce a digital
number proportional to the input signal.
 Sampling
Introduction:
As discussed previously in applications of ADC, that
in practice although we have a large number of
analog signals, but we prefer processing of digital
signals. For this purpose we should be able to
convert continuous signal to discrete signals.
Also Due to some recent advance development in
digital technology, the inexpensive, light weight,
programmable and easily reproducible discrete time
systems are available.
Processing of discrete time signals is more flexible.
 Since analog signal has a infinite number of values that
means for every time instant there is a value, so it is
not possible to digitize the signal before discritising the
time and this problem is solved by fundamental
mathematical tool known as Sampling theorem,
which is extremely important and useful in signal
processing.

With the help of sampling theorem, a continuous time

signal may be completely represented and recovered
from the knowledge of samples taken uniformly.
 Sufficient number of samples of the signal
must be taken so that the original signal is
represented in its samples completely.
Also it should be possible to recover or
reconstruct the original signal completely from
its samples.
The number of samples to be taken depends
on maximum signal frequency present in the
signal.
Different types of samples are also taken like
ideal samples, natural samples and flat top
samples.
Sampling theorem :
A continuous time signal may be represented
in its samples and recovered back if the
sampling frequency is fs 2fm
Where,
fs = Sampling frequency
fm = Maximum frequency present in the signal

To prove the sampling theorem, we will start

with a special class of signal called band
limited signal.
 Proof of sampling theorem:
We will see that a signal whose spectrum is
band limited to fm Hz, can be reconstructed
exactly without any error from its samples
taken uniformly at a rate fs > 2fm Hz.
Consider a continuous time signal x(t) whose
spectrum is band limited to fm Hz.
This means that signal x(t) has no frequency
components beyond fm Hz.
So X(w)=0 for |w|> wm where wm = 2 fm
Sampling theorem
 Few points about sampling
theorem
From the spectrum of G(w) it is clear that, as long as signal is sampled at a
rate fs > 2fm, specutrum will repeat periodically without overlapping.
The spectrum of sampled signal is extended up to infinity and the ideal
bandwidth of sampled signal is infinite. But our purpose is to extract our
original spectrum X(w) out of spectrum G(w).
The original desired spectrum X(w) is centered at w=0 and is having
bandwidth or maximum frequency equal to wm. The desired spectrum
may be recovered by passing sampled signal with spectrum G(w) through
low pass filter with a cut off frequency of wm. This means low pass filter
will pass only low frequencies up to cutoff frequency wm and rejects all
other higher frequencies, the original spectrum x(w) extended up to wm
will be selected and all other successive higher frequency cycles in the
sampled spectrum will be rejected.
So in this way original spectrum X(w) will be extracted out of spectrum
G(w). This original spectrum X(w) can now be converted to time domain
signal x(t).
 Nyquist rate and Nyquist interval
When the sampling rate becomes exactly
equal to 2fm samples per second, then it is
called Nyquist rate.
Nyquist rate is called minimum sampling rate,
given by,
fs = 2fm
Maximum sampling interval is called Nyquist
interval and given by
Ts = 1/2fm
 How to reconstruct the original signal?
Answer is :Interpolation.
The process of reconstructing a continuous time
signal from its samples is called an interpolation.

As discussed earlier, a signal x(t) band-limited to

fm Hz can be reconstructed completely from its
sample.
This is achieved by passing the sampled signal
through an ideal low-pass filter of cut-off frequency
fm Hz.
 h(t)=2fmTssinc (2fmt) .(1)

h(t) = 0 at all Nyquist instants t = n/2fm except at

t=0.
Assume sampling is done at Nyquist rate, then
Ts=1/2fm
So,2fmTs=1
Putting this value in equation 1,
h(t)=1*sinc (2fmt) = sinc (2fmt)

x[k](t-kTs) h(t) x[k]h(t-kTs)

X(t) = x[k] h(t-kTs)

= x[k] sinc (2fm(t - kTs))
X(t) = x[k] sinc (2fmt - k)

For the kth sample,

X(t) = x[kTs] sinc(2fmt - k)..Interpolation formula
 Practical difficulties in signal
Reconstruction
1. If a signal is a sampled at the Nyquist rate fs = 2fm Hz, the spectrum of
x(w) consists of repetitions of x(w) without any gap between successive
cycles. To recover g(t) from x(t), we need to pass sampled signal through
an ideal low pass filter. But such a filter is unrealizable.

2. A practical solution to this problem is to sample the signal at a rate

higher than Nyquist rate ( fs > 2fm, Oversampling). This yields x(w),
consisting of repetitions of x(w) with a finite band gap between
successive cycles. We can recover X(w) form G(w) using a low pass filter
with a gradual cutoff.
 3. Aliasing
If a signal is a sampled at the Nyquist rate fs < 2fm (Under
sampling), then the successive cycles of the spectrum of
sampled signal overlap with each other as shown below. And
this is known as aliasing.
From the figure it is clear that, because of overlap due to
aliasing, it is not possible to recover original signal from
sampled signal by low pass filtering since signal is distorted.
 Another practical difficulty in reconstructing a signal from its sample is
that sampling theorem was proved on the assumption that the signal is
band limited. But all practical signals are time limited. And it is proved that
a signal cannot be time limited and band limited simultaneously. Since any
information signal contains a large number of frequencies, so to decide a
sampling frequency is always a problem.

Therefore, a signal is first passed through a low pass filter. This low pass
filter blocks all the frequencies which are above fm Hz. This process is
known as band limiting of original signal.

This low pass filtering is known as pre alias filter because it is used to
prevent aliasing effect. After band limiting it is easy to decide sampling
frequency since maximum frequency is now fixed at fm Hz.

In short to avoid aliasing:

1. Prealias filter must be used to limit band of frequencies of the signal
to fm Hz.
2. Sampling frequency must be fs > 2fm
 Sampling techniques
1. Instantaneous sampling (Ideal sampling)
We used this sampling in proof of sampling theorem
Practically not possible to implement
The circuit which produces ideal samples is known as switching
sampler.

2. Natural sampling:
This method is used practically
The circuit which produces ideal samples is known as Natural
sampler.

3. Flat top or rectangle sampling

This method is used practically
Widely used
 Applications of sampling theorem
PAM
PPM
PWM
PCM
DM
PCM(Pulse code modulation)
Digital pulse modulation technique..

Sampler Quantizer Encoder

Binary
m(t) output

- Band limited signal

- Impulse train sampling
- Nyquist criteria
- Interpolation formula
Quantization:
Rounding off the values/discritizing amplitude
The amplitude of the signal with amplitude lies in the
range(-mp,mp) is partitioned into L subintervals,
each of magnitude 2mp/L.
Now each sample value is approximated by the
midpoint value of the subinterval in which the
sample falls.
It means we are doing some kind of approximation.
And so process of quantization will introduce some
kind of error or noise.(Quantization noise).
 S/N ratio for PCM:
SNR=Signal power/Noise Power
We first calculate the noise power:
In PCM because of quantization, errors are introduce in the signal. This
error is called quantization error.
quantization error=Quantized value-Original signal sample value
We have seen that quantization levels are separated by =2mp/l.
Since the sample value is approximated by mid point of the subinterval
in which the sample falls, the maximum quantization error is v/2.
Since noise is a random phenomenon and so we can not define noise
but we can define its probability and for that there is one special
function called as probability density function(PDF)
After analysis we can write : Nq=(v)2/12
 Mean Square Error caused by
Channel noise in PCM
There are two sources of error in PCM scheme :
Quantization error and pulse detection error.
The pulse detection error is quite small
compared to the quantization error and ignored.
In present analysis, we shall assume that the
error in the received signal is caused exclusively
by quantization.
 Transmission bandwidth for PCM
Given by half of the signaling rate (number of
bits per second)
signaling rate = number of bits per second = (no. of
bits per sample) * (no. of samples per second)
And which yields, signaling rate is n*fs
So,
BW = 1/2 * (n*fs)
BW = n fm
Exponential increase of the output SNR
L = 2n L is quantization levels.
 Uniform Quantization
When the steps are uniform in size.
Problems with uniform quantization:
Only optimal for uniformly distributed
signal.
The quantization noise is the same for all
signal magnitudes.
Solution:
Using non-uniform quantization
 Non uniform quantization
Researchers have proved that:
Real audio signals (speech and music) are more
concentrated near zeros.
Human ear is more sensitive to the quantization errors
at small values.
Therefore, to handle these problems, researchers suggest
non-uniform quantization, where the quantization step size is
smaller near zero and it gets larger gradually towards the max
and min levels.
Since the quantization errors is directly proportional to the
step size, reducing near zero would reduce errors at this
region.
 Main approach for non uniform
quantization
Provide fine quantization of the weak signals
and coarse quantization of the strong signals.
How to get non-uniform quantization?
The non-uniform quantization is practically
achieved through a process called
Companding. ( = Compression + Expansion)
Transform the signal using any logarithmic
compression
-law
A-law
 -law:
In this companding, the compressor characteristics is
defined by equation. The normalized form of
compressor characteristics is shown in the figure.
The -law is used for PCM telephone systems in the
USA, Canada and Japan. A practical value for is 255.

A-law:
In A-law companding, the compressor characteristics
is defined by equation. The normalized form of A-law
compressor characteristics is shown in the figure.
The A-law is used for PCM telephone systems in
Europe. A practical value for A is 100.
A law and law:
The law (for positive amplitude) is

When law compandor is used the output SNR is given by,

 DPCM(Differential Pulse
code modulation)
 In addition to PCM some other forms of digital
modulation have been developed, which are:

DPCM
DM (Delta modulation)
ADM (Adaptive Delta Modulation)

It also involves the 3 basic steps of PCM. i.e.

sampling, quantization and coding.
 Why DPCM?
The samples of a signal are highly correlated
with each other. This is because any signal
does not change very fast. i.e its value from
present sample to next sample does not differ
by large amount.
The adjacent samples of the signal carry same
info with very little difference.
When these samples are encoded by standard
PCM system, the resulting encoded signal
contain redundant information.
Differential Pulse Code Modulation
(DPCM)
A technique in which difference between the
actual sample value and its predicted value
(predicted value is based on previous sample or
samples) is quantized and then encoded
forming a digital value.
The DPCM works on the principle of prediction.
The prediction may not be exact but it is very
close to actual sample value.
 SNR improvement in DPCM
Let mp and dp be the peak amplitude of m(t)
and d(t) respectively.
If we use the same value of L in both cases, the
quantization step v in DPCM is reduced by the
factor dp/mp.
Because the quantization noise power is
(v)2/12 the quantization noise in DPCM is
reduces by the factor (mp/dp)2, and the SNR is
increases by the same factor.
Continue.
Moreover, the signal power is proportional to
its peak value squared.
Therefore the Gp(SNR improvement due to
prediction) is
Gp=Pm/Pd
Where Pm and Pd are the powers of m(t) and
d(t).
In terms of dB units, this means that the SNR
increases by 10log10(pm/pd) dB.
 Hence the equation,

applies to DPCM also with a value of that is

higher by 10log10(pm/pd) dB.
 Delta Modulation
Delta Modulation is a special case of DPCM.
In DPCM scheme if the base band signal is sampled
at a rate much higher than the Nyquist rate
purposely to increase the correlation between
adjacent samples of the signal, so as to permit the
use of a simple quantizing strategy for constructing
the encoded signal, Delta modulation (DM) is
precisely such as scheme.
Delta Modulation is the one-bit (or two-level)
versions of DPCM.
Practical realization of DM
 Analog signal m(t) is compared with feedback signal
m~q(t).
The error signal d(t) = m(t) - mq~(t) is applied to
comparator.
If d(t) is positive, the comparator output is constant
signal of amplitude +E and if d(t) is negative, the
comparator output is E.
Thus difference is a binary signal (L=2) that is
needed to generate 1 bit DPCM.
 The comparator output is sampled by sampler
at a rate of fs higher than the Nyquist rate.
The sampler thus produces a train of narrow
pulses of dq[k] with a positive pulse when
m(t)>mq~(t) and negative pulse when
m(t)<mq~(t).
 The modulated signal dq[k] is amplified and
integrated in the feedback path to generate
mq~(t) which is the approximation of m(t).
Delta modulation
Noise in Delta Modulation Systems

slope overload distortion

granular noise
 Slope overload distortion
Arises because of large
dynamic range of input signal

As observed from figure the

rate of rise of input signal is so
high that the staircase signal
cannot approximate it, the
step size becomes too small
for staircase signal mq(t) to
follow the step segment of
m(t).

Hence large error between

m(t) and mq (t), known as slope
overload.
 Granular Noise
In contrast to slope overload
Occurs when step size is too large
Usually relatively flat segment of the signal
Analogous to quantization noise in PCM systems
Conclusion:
1. Large step-size is necessary to accommodate
a wide dynamic range
2. Small step-size is required for accuracy with
low-level signals
* compromise between slope overload and
granular noise..
Adaptive delta modulation, where the step size is
made to vary with the input signal i.e. large step
size is required to accommodate wide dynamic
range of input signal and small step size are
required to reduce granular noise.
 Signaling rate for DM
signaling rate= number of bits per second
=(no. of bits per sample)*(no. of samples per
second)
=1*fs
signaling rate(DM)= fs
S/N ratio for DM
Comparison between DM and PCM
 Adaptive Delta Modulation
The performance of a delta modulator can be
improved significantly by making the step size of the
modulator assume a time-varying form.
In particular, during a steep segment of the input
signal the step size is increased.
Conversely, when the input signal is varying slowly,
the step size is reduced. In this way, the size is
adapted to the level of the input signal. The resulting
method is called adaptive delta modulation (ADM).
 ADM Transmitter
The logic for step size control is added in the
diagram. The step size increases or decreases
according to a specified rule depending on one bit
quantizer output.
If one bit quantizer output is high(i.e. 1), then step
size may be doubled for next sample.
If one bit quantizer output is low, then step size may
be reduced by next step.
ADM Transmitter
 ADM Receiver
In receiver of ADM, there are two portions.
The first portion produces the step size from each
incoming bit. Exactly the same process is followed as
that in transmitter.
The previous input and present input decides the
step size.
Then it is applied to an accumulator which builds up
staircase waveform.
The LPF then smoothens out the staircase waveform
to reconstruct the original signal.
ADM Receiver
 Advantages of ADM
The S/N ratio becomes better than ordinary DM
because of the reduction in slope overload distortion
and granular noise.
Because of the variable step size, the dynamic range
of ADM is wider than simple DM.
Utilization of bandwidth is better than DM.