DSP - SGM

Sugata Munshi
Dept. of Electrical Engineering,

Jadavpur University, Kolkata.
DISCRETE-TIME SIGNALS AND

SYSTEMS
A PRESENTATION BY
SUGATA MUNSHI
DEPARTMENT OF ELECTRICAL
ENGINEERING
JADAVPUR
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UNIVERSITY
Sugata Munshi
Discrete-Time Signals
Digital Signal is particular type of discrete-time signal.
A discrete time signal is one which is defined only at
discrete instants of time.
s(t)
Its values are known as

Samples
samples.
Hence it is also known
t
as :
sampled-data signal.
If the samples are uniformly spaced in time, we have a

uniformly sampled signal.
One of the distinctive feature of digital signals is that, they

are quantized both in time and magnitude. That is, they are
discrete-time discrete-magnitude signals. That means not
only are they defined for discrete instants of time but they
can assume only discrete values.
Another distinctive feature of digital signals is that their

sample values are coded in binary numbers.
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magnitude quantization
levels
s(t)
5q
4q
3q
2q
q
Block diagram of a typical Digital Signal Processing System

Analog
signal
Digital
signal
ADC
Digital
Signal
Processor
Digital
signal
DAC
Analog
signal
Digital
Memory
The digitized analog signal is processed by a digital signal processor.

Processing can be of various types e.g. integrating or differentiating or
filtering or finding out the DFT (to determine the frequency spectra of the
signal). The processed signal is converted into an analog signal if necessary
or stored in memory for use in future.
BENEFITS
DIGITALLY:
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OF
PROCESSING
SIGNALS
Sugata Munshi
Guaranteed Accuracy determined only by

number of bits used.
Perfect
Reproducibility:
Identical
performance from unit to unit, since there is no

variations due to component tolerance.
No drift in performance with temperature or
age.
Greater Flexibility:
Can be programmed
and reprogrammed to perform a variety of

functions, without modifying the hardware.
Superior Performance: DSP can be used to
perform functions not possible with analog
systems. Example: Linear phase response can
be
achieved,
complex
adaptive
algorithms can be implemented.
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fitering
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Tremendous advancements in electronics

industry are fruitfully utilized: Compact
size, lower cost, low power nconsumption,
greater speed.
The Sampling Process (Uniform Sampling)
A sampling operation transforms a continuous time signal
into a discrete-time signal.
Ideal switch( closes periodically at intervals of ,

remains closed for a moment and opens immediately)
is the sampling period
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fs
is the sampling frequency.
A more realistic case
The duration of closure of the switch should be finite (<)
Sampling Process Modelled As Pulse

Amplitude Modulation (PAM):
The process of periodic sampling of a continuous time
signal x(t) can be considered as a pulse amplitude
modulation process in which a unit pulse train p(t) is
multiplied by the signal x(t) to produce the sampled version
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x*p t x t p t
p(t) (carrier)
Pulse
Amplitude
Modulator
x(t)
modulating
signal
x(t)
()
x*p(t) = x(t) p(t)
x*p(t)
= Time period of
unit pulse train
= Sampling period
= Pulsewidth =
Sampling duration.
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p t u t K u t K
K
x t x t u t K u t K
K
*
p
If the sampling duration is very small compared to the

smallest time constant of the continuous-time system that
*
has generated x(t), then x p t can be approximated by a
sequence of flat-topped pulses.

x*p t
That is,
for K t K 1
Then,
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K=0,1,2,....... ,
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x t x K u t K u t K
*
p
Taking the Laplace transform on both sides ,
1 es K s
X s x K
e
K
s
*
p
Now,
1 e
1 1 s
...........
2!
s
If is very small compared to , 1 e s
X *p ( s)
Ks
x
(
K
Taking inverse Laplace Transform of both sides,
x (t ) x (t )
*
p
x( K ) (t K )
for small each pulse becomes a vertical line.

R.H.S. represents a train of impulses, with the strength of
the impulse at t K equal to
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x( K )
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Hence the finite pulsewidth sampler can be approximated

by an impulse sampler followed by an attenuator with
gain .
x*p(t)
x(t)
()
x(t)
x*i(t)
= x*(t)
(very
small)
x*(t)
impulse sampler
x (t )
*
i
x( K ) (t K )
Here the role of is just scaling. Hence it is considered as unity.
x (t )
*
x( K ) (t K )
Thus, for the sake of mathematical maneuvering a sampleddata signal can be represented by a train of scaled impulses
and the strength of the impulse at any instant is equal to the
value of the sample at that instant. That is, actual sampling
process can be represented mathematically by
sampling or impulse modulation.
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impulse
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x(t)
x*(t)
Impulse
Modulator
t
T(t)
Output of the impulse sampler is
x (t ) x(t ) (t ) x(t ) (t K )
*
Reconstruction of Continuous Time Signal from

its Samples:
- 4
- 3 - 2
In the absence of any additional conditions or
information, a continuous time signal can not be

uniquely specified by a sequence of equally spaced
samples. In the above figure, three continuous time
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signals that can generate the given set of samples have

been shown.
In general, from a given sampled data signal an
infinite number of continuous time signals can be
reconstructed. Hence, reconstruction of a continuoustime signal from its samples becomes a problem, unless
some other information is available.
Reconstruction
of
Continuous-time
sinusoid from its samples:

x(t)
A
to
x*(t)
Let a continuous-time signal
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2
x t A sin t A sin 2 ft A sin
T
be sampled with a sampling period .

1
f
Sampling rate s
If the sampling starts at a time angle 2 fto from

the zero instant as shown, then the sampled-data signal is
where, n = 0,1,2,..
f
x n A sin 2
n
fs
2 n
A sin

or,
f
T
where, s
f
is the time period in number of samples.

Case I:
>> 1, i.e., fs >> f, << T

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x*(t)
Reconstructed analog
sinusoidal signal from
x*(t)
The analog sinusoid x(t) can be uniquely reconstructed from

x (n ).
Case II:
< 1, i.e., fs < f, > T
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f
fs f
2
n
x n A sin 2 n 2 2 n
1
A sin 2 n 1

f fs
A sin 2 n
f
s
f
f
fs
2
3
n
x n A sin 4 n 2 4 n
1

A sin 2 n 2

f 2 fs
A sin 2 n
f
s
f
f
fs
3
4
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x n A sin 6 n 2 6 n
1

A sin 2 n 3

f 3 fs
A sin 2 n
fs
for
f
f
fs
K 1
K
K 1, 2,3, etc
.x n A sin 2 K n 2 2 K n
f Kf s
A sin 2 n
f
s
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Also,
and
i.e,
2 n
x n A sin 2 Mn

n
x n A sin 2 Mn 2

M = 1,2,3, etc.
x n A sin 2 n M

1
x
(
n
ASin
[2
n
(
M
) ]
and
or,
M = 1,2,3, etc.
Mf s f
x n A sin 2
n
fs
Mf s f
x
n
A
sin
2
and
f
s
M = 1,2,3, etc.
Thus, since x n is also the sampled version

of a continuous-time sinusoid of frequency
f f Kf s f , the sinusoid that is actually
represented
by
f f Kf s
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x n , has a frequency
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Frequency
f Kf s is the Alias of
frequency f .
Phenomenon is called aliasing.
It is the
effect of undersampling.
Case III :
= 1, i.e., fs = f, = T
x n A sin 2 n A sin
for all values of n.
x(t)
x*(t)
A sin
Reconstructed
signal
Reconstruction gives a steady value.Samplings at zero

crossings yield no information.
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CASE IV:
1 2, i.e. , f f s 2 f ,
T
T
2
x(t)
x*(t)
t
x n A sin 2 A sin 2 n 2 1 n

A sin 2
n
f fs
A sin 2
n
f
s
f f
A sin 2 s
n
fs
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Also
2 n
x n A sin 2 Mn

M = 1,2,3,etc
2 n
x n A sin 2 Mn

M = 2,3,4 etc.
Mf s f
x n A sin 2
n
fs
M=1,2,3,..
and
Mf s f
x n A sin 2
n
fs
M=2,3,4..
x n is also sampled version of sinusoid with freq.

f fs f f .
Thus analog sinusoid reconstructed by interpolation, from
samples
x n , will have a frequency f s f f .
So, frequency f of original signal is folded back into the

interval 0 f
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fs
.
2
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Frequency
fs f
is
alias of f for
f fs 2 f
Case V:
2, i.e., f s 2 f , T
x n A sin 2 A sin n
2
x(n ) ASin
x n A sin
for even n
&
for odd n .
x n can also be represented by

n
B sin n , i.e. B sin 2

2
such that
B sin A sin
So, x n can also be sampled version of sinusoid

B sin t .
Infinite no. of combinations of B &
B sin A sin .
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yield
Hence, if
0,
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it is possible to have infinite number of
reconstructions of analog sinusoids of same frequency f, but

differing in amplitude and phase.
x(t)
ion
Reconstruct
x*(t)
with f s 2 f , original sinusoid can not be uniquely

reconstructed from its samples, by interpolation.
Case VI:
2, i.e. f s 2 f , T
n
x n A sin 2

A sin 2 Mn 2

n
A sin 2 Mf s f
fs
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(M = 1,2,3......)
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Also,
n
x n A sin 2 Mf s f (M = 1,2,3,..)
fs
x(t)
Reconstruction
x*(t)
x n can be sampled version of x(t) as well as of analog

sinusoids with frequencies Mf s f f & Mf s f f .
Sinusoid x(t ) A sin 2 ft can be uniquely reconstructed

from x*(t) by interpolation.
Observations :
1
x t A sin 2 ft sampled at f s starting
from t =0, such that

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0 f
1
fs .
2
y t A sin 2 f Mf s t (where M is a +ve integer)

sampled at same rate fs.
Then the sampled versions are,
x n A sin 2 fn
(1)
&
y n A sin 2 Mf s f n
A sin 2 nM 2 fn
or
y n A sin 2 fn
Sequences x n and
(2)
y n are identical.
Not possible to distinguish between the

samples of sinusoids whose frequencies differ
by integral multiple of sampling frequency.
Frequency f Mf s (M=1,2,3,.) is folded back into

interval 0 f
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fs
.
2
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Frequency
1
fs
2
folding frequency
or Nyquist
frequency.
Frequency f is alias of Mf s f .
z t As i n 2 Ms f f tare sampled at
Again, if
f s , 0 f s , then
2
z n A sin 2 Mf s f n
A sin 2 nM 2 fn
or
z n
A sin 2 fn
Hence frequencies Mf s f
(3)
are folded back into
fs
0
interval
2.
Frequency
Spectra
of
Discrete-Time
Signals
x(t) is sampled at f s 1 to obtain a D.T. signal x*(t).
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Fourier Transform of x(t) is

X ( j ) F x(t )
x(t )e jt dt
F.T. of x*(t) is ,
X j F x n t n
n
F x t t n
n
*
Consider the periodic signal
t t n , with a period .
n
can be expanded into Fourier series as
( )
Fm
(t-n )=
n=-
(t ) e
jms t
dt
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e jmst
t 0
(t ) e
Fm e jmst
jms t
for all m,
dt
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Therefore,
( )
where,
(t-n )=
n=-
s 2 f s
e jmst
Hence,
X ( j )
*
X j m
s
m -
For simplicity, let x(t) be such that X(j) is a +ve real-valued

function of .
i.e. x(t) has only amplitude spectrum and no phase spectrum.
Then
X j X
and
X * j X *
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X ms f s X ms
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X ms has same waveform as X except that it is

centred at ms .
X * is sum of all repetitions of
ms .
f s X ( )
centred at
m 0, 1, 2,.....
Consider x(t) as bandlimited signal, with bandwidth W
r/s ; i.e. the frequency spectrum is 0 for > W.
f m BW
. . in Hz =
s 2W
i.e.,
W
2
f s 2 f m repetitions do not overlap
theoretically possible to exactly reconstruct x(t) from
x*(t) by
passing x*(t) through an ideal (brick-wall type) L.P. filter with cutoff
freq.
s
1
& pass band gain
(i.e. ).
fs
2
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If s 2W i.e.
f s 2 f m adjacent replications of
X*()
overlap & resulting X*() will no longer preserve

information of X .
If we try to reconstruct x(t) from x*(t) by L.P. filtering,

resulting signal will be x(t) contaminated by higher frequency
components. This phenomenon is known as aliasing.
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Sampling Theorem :If a Band-limited continuous time

signal x(t) containing no frequency component greater than
f m , is sampled at a frequency f s , then x(t) can be uniquely

reconstructed from its sampled version x*(t) if f s 2 f m .
Sampling rate f N 2 f m is known as Nyquist rate.
ANTI-ALIAS
FILTER
Practically no analog signal is naturally band limited. So, prior to

sampling (i.e. A/D conversion) the analog signal is made
bandlimited ( as far as practicable ) by passing it through an analog
filter.
Thus, to get rid of aliasing, the C.T.
x t is processed (prior to
sampling) by analog L.P. filter with cut-off frequency <
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fs
2 .
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Pass band of anti-alias filter depends on type of analog signal

to be processed.
Frequency spectra of human voices contain useful info up to

about 4 KHz. So for processing speech signals, cutoff freq. fc is
selected a little bit more than 4 KHz. The sampling is carried
out at f s 2 f c .
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THE Z-TRANSFORM
What is Z-Transform ?
x(t) sampled at a rate
fS 1
Sampled version of x(t) is
x * (t )
x(n ) (t n )
(1)
where n = 0, 1, 2,
or, x * (t )
where
x (t n )
(1a)
xn x(n) x[n] x n x t t n
L.T of both sides of equation (1a) yields
L[ x * (t )] L[ xn (t n )]
n
or,
X * ( s)
(2)
sn
x
e
n
(3)
X*(s) Discrete Laplace Transform or Starred Laplace

Transform.
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s
e
z . Then z is a complex variable.
Let
Substitution of
X * ( s)
z es
in (3) gives
n
x
z
n X ( z)
(4)
X(z) is known as the Z-transform of sequence xn

In operational form,
Z [ xn ] X ( z )
xn z n
(5)
where Z [ . ] stands for Z-transform operator.

If xn represents a causal sequence, then,
Z [ xn ] X ( z ) xn z n
n 0
(6)
Eq. (5) gives 2-sided or bilateral Z-transform a series

in both +ve and -ve powers of z-1 .
Eq. (6) gives one sided or unilateral Z-transform a

series in +ve power of z-1.
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The Complex-Variable z
z e s
1
s ln z
z 1 1 z 1 3 1 z 1 5
ln z 2

..............
z
1
3
z
1
5
z
1 z 1
ln( z ) 2
1
1
Hence,
2 1 z 1
s
1 z 1
Bilinear transformation or Tustin transformation
s j
z es e e j
or,
z e
(in polar form)
e cos j sin
u jv
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(in cartesian form)
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z e
if 0,
z 1
if 0,
z 1
if 0,
z 1
Z-TRANSFORMS OF SOME
ELEMENTARY
SEQUENCES:
1. Unit Step Sequence
un 1 for n 0
0 for n0
Un
1
-2 -1
01 2 3 4 5 6
Z un un z
n
z n
n 0
1 z 1 z 2 ........
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1
z
1 z 1 z 1
; ROC : z 1
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L u t
1
s
jv
z-plane
s-plane
2 . Causal Exponential Sequence:
xn e an u n
Z xn e an z n
n 0
1 e a z 1 e 2 a z n e 3a z 3 .........
1
z
a
;
ROC
:
z
e
1 e a z 1 z e a
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Alternatively,
xn e anun ;
where
a = time const. in terms of
number of samples.
Then,
1
X ( z ) Z xn
1 e a z 1
;
ROC : z> e-a
If a is real:
3 . Unit Discrete Impulse Sequence or Kronecker Delta

Sequence or Unit Sample Sequence.
n = 1 for n=0
=0 for n 0
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n
1
n
0
-5 -4 -3 -2 -1
1 2 3 4 5
Z n n z n z o 1 ; ROC : Entire z-plane.

n
4 . Causal Sinusoidal Sequence:
xn 0 for n0
A sin o n for n 0
i.e. xn = A sin o n un
xn
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Z [ A Sin 0 n un ]
Az sin o
z e jo
z e jo
A causal sinusoidal sequence can be also expressed as :
x n = A Sin (0 n)un ; where 0 is in radian. ,

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then,
Az 1 Sin0
X ( z)
; ROC : z 1
-1
2
1-2z Cos0 z
Some Important Properties of Z-Transform

Linearity of z-transform :
If F z Z [ f n ] ,
xn f n n
z Z [n ] , & are constants and
, then
X z Z[ xn ] F z z
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X z Z xn Z f n n
f n n z n
n 0
n 0
n 0
f n z n n z n
Z f n Z n
Proof:
F z z
Multiplication by an (scaling in the z-domain)

If
X(z) = Z[ xn ] ; ROC : r2 z r1 ,
then ,
Z a n xn X a 1 z
ROC : a r2 z a r1
Z a n xn a n xn z n
n
xn a z
X a 1 z
1 n
1
Z un
Example :
2
1 1 z 1
2
Real translation property (shifting properly)

If X z Z xn & K is a +ve integer, then,
Z xnk z K X z
ROC:same as of X(z) except z = 0
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(1)
K
Z
x
z
X z
nk
and
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(2)
ROC: same as of X(z) except for z =

.
Thus, multiplication of a Z-transform by z
has the effect of
*
delaying the time function x t by K time units, i.e. xn by
K samples.
also known as right shift or backward time
shift.
Multiplication of a Z-transform by zk
has the effect of
*
advancing time function x t by K time units, i.e. by K
samples.
Also known as left
shift or forward time shift.
Real Convolution Property

D.T. linear convolution or convolution sum of sequences
g n is given by
yn xn g n xk g n k
k
for n 0, 1, 2, 3 etc.
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(1)
xn
&
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If Y z , X z and G z are Z-transforms of
yn
xn
& g n , then,
Y z X zG z ;
ROC: At least the intersection of those for
X z and G z .
That is, D.T. convolution of 2 sequences results in
multiplication in z-domain.
Proof:
Y z yn z
[ xK g n K ] z n
n K
or,
Y(z) = xK g n K z n
xK z
n K
gnK z
Let m=n-K
Y ( z )
Note:
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z m X ( z ).G ( z )
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If xn and g n are causal sequences, then

n
yn xn g n xk g nk
for n = 0, 1, 2,
k 0
etc.
Energy and Power Signals

Normalized energy:
N
E im xn xn
2
N n N
Normalized power:
N
1
xn
N 2 N 1 n N
P im
If E is finite, obviously P = 0
In this case,
xn
is an energy signal.
If E is infinite, P may be finite or infinite.

When P is finite and E = , then
xn
is a power signal.
Correlation Property
The cross-correlation of finite energy sequences x(n) is y(n) is
given by
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rxy (l )
x ( n) y ( n l ) x ( n l ) y ( n )
(1)
for
l 0, 1, 2, 3 etc.
and,
ryx (l )
y ( n) x ( n l ) y ( n l ) x ( n )
for
(2)
l 0, 1, 2, 3 etc.
Then,
S xy z Z rxy l X z Y z 1
; ROC is at least
1
the intersection of those for X z and Y z
Note:
rxy (l ) ryx (l )
Autocorrelation of a finite-energy sequence x(n) is
rx (l ) rxx (l )
x ( n) x ( n l ) x ( n l ) x ( n )
S x ( z ) Z rx (l ) X ( z ) X ( z 1 )
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Time Multiplication Property ( Differentiation in Z

Domain)
If
X ( z ) Z [ xn ]
Z [nxn ] -z
dX(z)
; both transforms have the same ROC
dz
Example :
Z [un ] U ( z )
z
;
z 1
Z[nu n ] -z
dU(z)
d z
z
z
dz
dz z 1 ( z 1) 2
Initial and Final Value Theorems
If Z xn X z , xn being a causal sequence,

Then,
x0 im X z
z
x im
z 1
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z 1 X
z
IVT
z 1) X ( z ) .FVT
z lim(
z 1
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Mapping Between s-plane and z-plane:

Importance of the study
Design & analysis of C.T. system often rely on pole-zero

configuration of system T.F. in the s-plane.
Similarly, poles & zeros of Z-transform of the system transfer

function determine response of a D.T. system at sampling
instants.
Hence the study of mapping between s-plane and z-plane is

important.
z e e cos j sin
j
jv
S-plane
Z-Plane
Z
Trnasform
For = 0, |z| = 1.
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Thus, the j axis in the s-plane corresponds to the circumference

of unit circle centred at origin in the z-plane.
For < 0 , |z| < 1
So entire left half of s-plane is mapped into interior of unit

circle centred at origin, in z-plane.
For > 0, |z| > 1.
Hence entire right half of s-plane is mapped into exterior of

unit circle centred at origin, in z-plane.
:
Mapping of Left half of the s-plane into the z-plane:
s-plane divided into infinite no. of horizontal periodic strips,

each of width
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in direction of j axis.
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1
1
Each strip extends from j ( N ) s to j(N )s in direction
2
2
of j axis, where N 0, 1, 2, 3,...........

j
j3
s
2
N=1
Complementary strip
Primary strip
Complementary strip
N=0
j3
N=-1
s
2
N = 0 primary strip,
Others are complementary strips.
Mapping of left half of Primary Strip
j
jv
j
C
-
Primary
strip A
D
s
2
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S-plane
s
2
Z-plane
u
A
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In s-plane, A B C D E A representing boundary of

L.H. of
primary strip, is traversed in
counter clockwise
direction.
During
traversal,
varies
in
following
At A, z 10o
Interval [A,B]
z 1
180
At B, z 1180o
Interval B, C
z z 180o ,
1 z 0
0 l
At C , z 0 180o
Interval C , D ,
z z ,
1800 1800 ;
At D, z 0 180o .
Interval D , E :
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z 0
manner.
z z 180o ,
Sugata Munshi
1 z 0
At E , z 1 180o
Interval E , A :
z 1 , 1800 0o
Entire L.H. of primary strip is mapped into interior of unit

circle centred at origin in z-plane.
Mapping of left halves of complementary strips:

sc pt. in L.H. of strip with N = 1.
Sc
Sp
js j 3 s
2
p j s
2
0
Gc
j
s
2
j3
where
s p c j p is point in
L.H. of primary.
In general, any point sc in L.H. of any compl. strip can be
represented by
sc s p jNs
where
N 1, 2,........ & sp is a point in
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primary strip.
Sugata Munshi
Hence, in z-domain,
zc e
sc
s p jNs
s p
e e
j 2 N
s p
e e jNs
e
s p
zp
Inference:
Correspondence between
z-plane &
s-plane is not
unique.
A point in z-plane corresponds to infinite no. of points in
s-plane, although a point in s-plane corresponds to a
single point in z-plane.
So L.H. of every complementary strip is mapped into unit circle
in z-plane.
Aliasing of complementary strips into primary strip .
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Sugata Munshi
j
N=2
N=1
jv
Z
Trnasform
= 0, + s , + 2s------
Primary
Strip (N=0)
N=-1
N=-2
Z-Plane
S-Plane
Entire L.H. of s-plane is mapped into interior of unit circle in zplane, & entire R.H. of s-plane is mapped into exterior of unit
circle in z-plane.
j axis in s-plane maps into circumference of unit circle in zplane.
THE REGION OF CONVERGENCE

(ROC) OF Z-TRANSFORM
Consider bilateral Z-transform
X z Z xn xn z n
n
53 of 112
(1)
Sugata Munshi
where,
z re j and is in radians.
Now,
r z e
X z xn r n e jn
(2)
The region of convergence (ROC) of X(z) set of all values of z

for which X z attains a finite value.
Now,
X z xn r e
n j n
or,
X z xn r
or,
xn r n
n
xn r n
n0
xn
X z x n r n
n 1
n0 r
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n j n
xn r e
n
(3)
Sugata Munshi
If X(z) converges in some region of z-plane, both

summations in (3) must be finite in this region.
1st sum converges for values of r small enough such that

product sequence
x n r n , (1 n )
is absolutely
n
summable, i.e. x n r .
n 1
ROC of 1st sum consists of all points within a circle of

radius r
< r1.
2nd sum converges, for values of r large enough such that

xn
product sequence n , (0 n ) is absolutely summable, i.e.
r
n 0
xn
rn
ROC of 2nd sum consists of all points outside a circle of radius
r r2 .
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Sugata Munshi
Therefore, ROC of X(z) can be represented in general by
r2 r r1 ,
an annular region
i.e. r2 z r1 , which is a
common region where both of above mentioned sums are

< .
If r1 > r2 , there is no common region of convergence

of the 2 sums & hence X(z) does not exist.
Example:
xn a nun , an infinite duration causal sequence

X z a n z n az 1
n 0
n 0
az 1 1, i.e. z a ,
If
power
series
1 az .
1
X z
1
; ROC : z a
1 az 1
ROC is exterior of a circle of radius |a|.

If, a = 1,
X ( z ) [u n ]
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1
; ROC : z 1
1 z 1
converges
to
Sugata Munshi
jv
ROC
Example:
xn a nu n1 , anti-causal infinite duration sequence.

X z a
1
a 1 z
m 1
where, m = n .
2
3
Now, C C C ..........
C
1 C
where, |C| < 1.
a 1 z
1
X z
,
1
1
1 a z 1 az
if,
a 1 z 1, i.e.
z a
1
n
a
u
; ROC : z a .
n 1
That is,
1 az 1
ROC is interior of a circle of radius |a| & centred at origin in
z-plane.
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Sugata Munshi
Observation:
Closed form expressions of
Z-transforms of
sequences anun and -anu-n-1 are identical but

their ROCs are different.
Above examples reveal 2 important facts: 1st uniqueness of Z-transform.
2nd non-uniqueness of inverse Z-transform in absence of
citation of the ROC.
Thus a uniformly sampled signal xn is uniquely determined

by its Z-transform X(z) and the ROC of X(z).
Also seen ROC of Z-tr of a causal signal is exterior

of a circle centred at origin & that of
anti-causal
sequence is interior of a circle centred at origin in zplane.

Example:
xn a nun bnu n1
; two sided infinite duration sequence.
n 0
X z a n z n bn z n
az 1 b 1 z
n 0
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m 1
Sugata Munshi
The first sum converges if
z b
z a . The second sum converges if
Case I:
If
b a , the two ROC do not overlap. Hence X(z) does
not exist.
Example :
b 1 ,a 1
3
2
Case II:
If b a , there is a ring in z-plane with inner radius |a| and
outer radius |b| where both power series converge simultaneously.
Thus,
a b z 1
1
1
X z
; ROC: a z b
1
1
1
2
1 az
1 bz
1 a b z abz
This example shows that if there is a ROC for an infinite

duration two-sided time series, it is an annular region
centred at origin in the z-plane.
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Sugata Munshi
jv
jv
b
u
a
ROC
b < a
X(z) does not exist
Examples:
1.
xn 1, 2,5, 7 , a causal finite duration sequence.
X z 1 2 z 1 5z 2 7 z 3 ; ROC : entire z-plane

except z = 0,
2.
[r 2 = 0 ]
xn 2, 6,8,5,1 ; a two-sided finite duration sequence.
X z 2 z 2 6 z 8 5z 1 z 2 ; ROC : entire Z-plane
except z = 0 and z=.

[ r2 =0 ; r1 = ]
3.
xn 9,5,3, 2,1 , an anti-causal finite duration
sequence.
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Sugata Munshi
X z 9 z 4 5z 3 3z 2 2 z 1 ;ROC: entire z-plane except

z = .
[r1 =]
NOTE :
1. A rational Z-transform always has at least one pole
located on any boundary separating its regions of
convergence and divergence.
2. There is never a pole located inside the ROC.
INVERSE Z-TRANSFORM
If
Z xn X z ,
then,
xn Z 1 X z
xn is inverse Z-tr of X(z)

POWER SERIES EXPANSION:
LONG DIVISION (EXAMPLE- 1)
z2
X ( z)
; ROC : z 1
( z 1)( z 0.2)
Find Z -1 X ( z )
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Sugata Munshi
z2
1
X z 2
z 1.2 z 0.2 1 1.2 z 1 0.2 z 2

xn Z 1 X ( z )
ROC is z 1; x n is causal
We seek power series expansion in -ve powers of z.

1
xo Lim X z Lim
1
z
z 1 1.2 z 1 0.2 z 2
z 1 X z Lim 1 1 1.25
x Lim
z 1
z 1
z
1 0.2 z 1 0.8
The long division is illustrated below:
1 1.2 z 1 0.2 z 2
1 1.2 z 1 1.24 z 2 1.248 z 3 .......

1
1 1.2 z 1 0.2 z 2
1.2 z 1 0.2 z 2
1.2 z 1 1.44 z 2 0.24 z 3
1.24 z 2 0.24 z 3
1.24z -2 1.488 z 3 0.248 z 4
1.2483z 3 0.248 z 4
1.248 z ...................................
X z 1 1.2 z 1 1.24 z 2 1.248z 3 .............

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Sugata Munshi
But,
X z xo x1 z 1 x2 z 2 x3 z 3 ..............
xo 1
x1 1.2
x2 1.24
x3 1.248
div. can be contd. to give as many terms as desired.

Disadvantage not yielding xn in closed form.
Example 2.
z2
1
If X z 2
; ROC : |z| < 0.2,
1
z 1.2 z 0.2 1 1.2 z 0.2 z 2
It is clear that xn is anti-causal .

So power series expansion in +ve powers of z reqd.
5 z 2 30 z 3 155 z 4 780 z 5 .......
0.2 z 2 1.2 z 1 1
1
1 6z 5z 2
6 z 5z 2
6 z 36 z 2 30 z 3
31z 2 30 z 3
31z 2 186 z 3 155 z 4
156 z 3 155 z 4
156 z 3
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Sugata Munshi
X z 5 z 2 30 z 3 155 z 4 780 z 5 ..............

x1 0, x2 5, x3 30, x4 155, x5 780
xn ..........780,155,30,5, 0, 0
INVERSE TRANSFORM OF NON-RATIONAL FUNCTION

Example
X z log 1 az 1 ;
z a
Using power series expansion for log (1+x) with |x|< 1,
1 x n
log 1 x
n 1
n
Here,
x az 1
1
X z
n 1
an z n
n 1
xn Z 1 X z
or, xn
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n 1
n 1
a n for n 1
n
0 for n 0
n 1
a nun 1
Sugata Munshi
Alternatively
X z log 1 az 1 ;
z a
dX z az 2
dz
1 az 1
xn Z 1 X z
Let,
From the differentiation property,

dX z
az 1
Z n xn z
dz
1 az 1
a
n
Z a a un
1
1 az
n 1
Z nxn Z a a un 1
Z [nxn ] a a
or, xn
n 1
un 1
n 1
a nun 1
Problem:
X z sin z, ROC includes z 1 .
Expanding X(z) in a Taylor series about z = 0,

X z X z z 0 z
z
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2
K
dX z
z2 d X z
zK d X z
......
.....
dz z 0 2! dz 2 z 0
K ! dz K z 0
z3 z5
z 2 n 1
n
..... 1
n 0
3! 5!
2n 1!
Sugata Munshi
X z xn z n
n
xn 1
1
,
2
n
1
!
n 1, 3, 5,.........
PARTIAL FRACTION METHOD
x t
u t
X s
1
s
1
e at u t
sa
1
at u t
s ln a
X z
z
z 1
z
e anun
z ea
z
a n un
za
xn
un
Examination of 1st -order functions in
table for Z-
transforms shows that a factor z is required in the numerator

of these terms.
X z
Hence we expand
into partial fractions & then
z
multiply by z to obtain expansion of X(z).
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Sugata Munshi
Problem:
X z
z
z2 z 1
is
Z-transform of
causal sequence xn.
Determine xn.
Solution:
X z
1
2
z
z z 1
1
3
1
3
z 2 z 1 z j
z
2
2
2
2
j
j
3
z e z e 3
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Sugata Munshi
X z
K1
K2
j
j
z
z e 3 z e 3
1
1
K1 j
e 3 e 3
j 2sin
3
1
1
K 2 j
e 3 e 3 j 2sin
3
X z
j
j
3
3
z
e
z
j 2sin
j 2sin
3
3
1
z
1
z
j
j
j 3 (z e 3 ) j 3 (z e 3 )
xn Z 1 X z
jn
1 jn 3
3
e
e
un
j 3
2
n
sin
3
3
un
Problem:
z3 z2
X ( z) 2
; ROC : z 3
z 4z 3
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Sugata Munshi
Solution:
X z
z2 z
z
z 1 z 3
Converting above improper rational function M N into sum

of a constant & a proper rational function,
X z
5z 3
1
z
z 1 z 3
K
K
5z 3
1 2
z 1 z 3 z 1 z 3
K1 z 1
K 2 z 3
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5 z 3
1
z 1 z 3 z 1
5 z 3
6
z 1 z 3 z 3
X z
1
6
1
z
z 1 z 3
1
6
or, X z z
1 z 1 1 3z 1
Sugata Munshi
xn n 1 un 6 3 un
n
CONTOUR INTEGRATION METHOD
X z Z xn
xn Z 1 X z
X z z
n 1
dz
2 j C
where C is a closed contour within ROC of X(z) that encircles

origin in z-plane in a counter clockwise direction.
According to Cauchys residue theorem,
xn
n 1
n 1
X z z dz Residues of X z z at the poles inside C
2 j C
For a simple pole z = zo ,residue at
z zo is
z zo X z z n1 z z
For a pole of multiplicity m > 1 at
z zo ,
1
d m1
m
z
X ( z ) z n 1
o
m
Res. m 1 ! dz
zz
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Sugata Munshi
Proof:
X z xn z n
n
Multiplying both sides by Zk-1 and integrating w.r.t. Z about

closed contour in the ROC of X(z),
n k 1
dz
X z z dz xn z
k 1
C n
xn z k n 1dz
n
dz 2 j kn
k n 1
Now,
By Cauchy
1 if k n
0 if k n
kn
Hence
X z z
C
dz 2 j xn kn
K 1
2 jxk
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Sugata Munshi
xK
X z z
K 1
dz
2 j C
1
n 1
xn
X z z dz
2 j C
C is a contour in the ROC of X(z) traversed about origin in the

counter clockwise direction/
DISCRETE-TIME LTI SYSTEMS
x*(t)
1.
input
Discrete-time
System
xn
input
Discrete-time
System
y*(t)
output
yn
output
Discrete-Time Differentiator
x t
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Differentiator
y t
dx t
dt
Sugata Munshi
Derivative of C.T. signal x(t) :
y t
dx t
dt
y (t )
(1)
dy (t )
slope of x(t) at t=t
dt t t
x(t)
x(n)
Discrete-time
Differentiator
y n
y(n)
dx t
dt t n
n=0,1,2,..
y n =slope of x(t) at t n ; n = 0,1,2,
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Sugata Munshi
x(t)
Actual slope
x n
Approximate x n x n

slope
x n
n 1
slope approx. by backward difference relation.
y(n )
dx(t)
dt t n
x(n )- x(n - )
( 2)
smaller, approximation better.
y n
or,
1
x n x n
yn ao xn a1 xn1
(3)
(3)
1
1
a
,
and
a
1
where o
D.T differentiation algorithm can be pictorially represented

as
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Sugata Munshi
ao
xn
yn
Unit
delay
xn-1
a1
Taking Z-transforms , transfer fn of D.T. differentiator is
G z
Y z
ao a1Z 1
X z
1 z
(4)
D.T. Integrator
Integral of C.T. fn x(t) over interval 0 to t is:
t
y t x t dt
o
(1)
Value of y(t) = area under x(t) over 0 t t.

x(t)
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Integrator
y(t)
Sugata Munshi
x(t)
x(n)
Discrete-time
Integrator
y(n)
Then,
n
y n x t dt
(3)
or,
y n x t dt x t dt
n
or
y n y n x t dt
n
(4)
2nd term of R.H.S. of relation (4) = area under x(t) over
n t n .
area shaded rectangle shown below.
Smaller better approx..
x(t)
x n
x n
2 n x n
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t'
Sugata Munshi
Hence,
n
x(t)
dt
x(n )
(5)
Then, equation (4) becomes,
y n y n x n
or,
yn yn1 xn
or,
yn ao xn yn1
where,
(6)
(6a)
ao
Equation (6a) gives the 1st order difference equation for D.T.
integrator using rectangular integration.
Algo can be pictorially represented as:
xn
ao
yn
unit delay
yn-1
Taking Z-transforms of both sides,
Y ( z ) ao X ( z ) z -1Y ( z )
(5)
Z-transfer-function of the D.T. integrator is

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Sugata Munshi
ao
Y(z)
X(z)
1 z 1
GRI ( z )
1 z 1
z-1
(8)
X(Z)
G z
Y(Z)
z 1
jv
j
S-plane
Z-plane
u
True integrator
Approximate discrete-time
integrator
For more accurate integration,
x t dt
approx. by area of the shaded trapezium.
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Sugata Munshi
Hence,
(9)
From equation (4) and (9),
y n y n x n x n
2
yn
or,
yn ao xn a1 xn1 yn1
xn
or,
xn 1 yn 1
(10)
where, ao 2 , a1 2
Eq. (10) 1st order difference equation representing algo for
D.T. integrator based on trapezoidal integration.
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Sugata Munshi
xn
ao
yn
a1xn-1
yn-1
Taking Z-transforms :
Y ( z) a0 X ( z ) a1 z 1 X ( z ) z 1Y ( z )
The Z-transfer function is
Y z ao a1 z 1 1 z 1
GTI z

X z
1 b
2 1 z 1
z 1 (11)

2 z 1
z 1
X(Z)
Y(Z)
2 z 1
jv
j
S-plane
Z-plane
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u
-1
Sugata Munshi
1 z 1

s 2 z 1
2 z 1 2 1 z 1
s
Tustin Transformation relation

z 1 1 z 1
Observations:
S(t)
* If
S(n)
or
Sn
DTLTI
System
ds t
z n
dt t n
zn ao1sn a11sn 1
1
ao1 , a11
* If
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d 2s t
z n
dt 2 t n
Z(n)
or
Zn
Sugata Munshi
z n
ds t
ds t
dt t n
dt t n
x n x n x n x n 2
zn ao 2 sn a12 sn 1 a22 sn 2
ao 2 a22
, a12
2
* Similarly, if
d 3s t
z n
dt 3 t n
zn ao3 sn a13 sn1 a23 sn2 a33 sn3
* Hence, if
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d i s t
z n
dt i t n
Sugata Munshi
zn aoi sn a1i sn1 a2i sn2 ........ aii sni

Representing a discrete-time LTI system by a constantcoefficient difference equations:
x(t)
Continuous-time
LTI System
y(t)
The most basic mathematical model of a stable causal

continuous-time LTI system is the differential equation
representation, wherein i/p x(t) & o/p y(t) are related by the
linear differential equation.
For BIBO stability
d i y t M d i x t
bi
;
ai
i
i
i 0
i 0
dt
dt
N
M N
(1)
{ ai } & {bi} are constants.

N = order of the system.
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Sugata Munshi
x(t)
xn
Causal
Discrete-time
LTI System
yn
Signal
M
d i y t
d i x t
ai
i
i 0
dt t n i 0 dt i t n
N
Hence i/p & o/p of a causal DTLTI system can be related

by constant coeff. linear difference equation.
i 0
i 0
i yn 1 i xn 1
(2)
If N 0
Equation (2) can be expressed as
N
1 M
yn
i xni i yn 1
i 1
o i 0
Directly expresses o/p at instant n in terms of previous

values of i/p & o/p.
Taking Z-transform of both sides of eq. (2) ,
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Sugata Munshi
N
M
i z i Y z i z i X z
i 0
i 0
Z- T.F. of the system is :

M
i z
i
i z
o
i 0
M
Y z i 0
G z
N
X z
i z i
i i
z
i 1 o
N
i 0
Properties
of
Discrete-time
(3)
Convolution
and
Interconnection of LTI systems

X Z
xn
G Z
Y Z
yn
gn
g n z 1 G Z
Y Z G Z X Z
O/p yn of a DTLTI system excited by i/p sequence xn,

can be expressed as convolution of xn and gn, where gn is
the weighting sequence of the system.
yn xn g n xm gn m
(1)
Also,
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yn g n xn g m xn m
m
(2)
Sugata Munshi
xn n
If
yn g n n g m n m
m
Now
nm 1 for n m
0 for n m
and
yn gn
Hence gn is the impulse response of the system.
Following are the properties of convolution:1. Commutative Property:-
xn gn gn xn
(3)
xm g nm g m xn m
i.e. m
0
m0
xn
gn
(4)
yn
gn
Y(Z)
G(Z)
xn
yn
In the Z-domain,
X(Z)
G(Z)
Y(Z)
X(Z)
2. Associative Law:
xn g1n g 2n
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x n g1n g 2n
(5)
xn g1n y1n
Sugata Munshi
&
y1n g 2 n xn g1n g 2 n yn
From a physical point of view, x n may be interpreted as

the input signal to an LTI system with impulse sequence
g1n . The output of this system denoted as y1n , becomes
the input to a second LTI system with impulse sequence
g2n. Then the output is
yn y1n g2n xn g1n g2n

which is the left hand side of equation (5).
Thus, the LHS of equation (5) corresponds to having two
LTI systems in cascade. RHS of equation (5) indicates that
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Sugata Munshi
xn is applied to an equivalent system having an impulse

response
gn g1n g2n
and yn xn g n
From commutative law we also have
g n g 2n g1n
As
generalization of associative law, if there are L
DTLTI systems in cascade, with impulse response

sequence g1n , g2n ,.., gln, then equivalent system will
have an impulse-response
gn g1n g2n .......... g Ln

xn
g1n
g2n
gLn
yn
xn
g n g1n g 2 n ,......., g Ln
yn
In the Z-domain
X(Z)
Y(Z)
G1(Z)
G2(Z)
G3(Z)
3. Distributive to Property:
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GL(Z)
X(Z)
G(Z) = G 1(Z) G 2(Z) Y(Z)

GL(Z)
Sugata Munshi
xn g1n g2n xn g1n xn g2n
(6)
This law implies that if we have two LTI systems with

impulse responses g1n and g2n excited by the same input
sequence xn, the sum of the two responses is identical to
the response of an overall system with impulse response
gn g1n g2 n
As a Generalization the interconnection of L LTI systems
in parallel with impulse responses g1n, g2n, .., gLn
and excited by the same input xn is equivalent to one
overall system with impulse response
L
g n g jn
j 1
g1n
g2n
xn
g3n
gLn
In the Z-domain
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yn
xn
gn = g1n + g 2n
+ ...... + g Ln
yn
Sugata Munshi
G1(Z)
G2(Z)
X(Z)
Y(Z)
X(Z) G(Z) = G 1(Z) + G 2(Z) Y(Z)

+ + G L(Z)
GL(Z)
CONCEPT OF CAUSALITY OF DTLTI SYSTEMS
A system is said to be causal if the output yn at any
instant n depends on present and past inputs xn, xn-1, xn-2,

etc. and does not depend on future inputs xn+1 , xn+2 etc.
If a system does not satisfy this condition, it is non-
causal.
Examples of non-causal systems
# yn 2 xn 1.5 xn 3
# yn x2 n
# yn xn2
# yn x n y2 x2
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Sugata Munshi
From the definition of causality it is clear that for a causal system,

the impulse response sequence gn should be a causal sequence.
Constraint on Transfer Function
Example:
z3 2z 2 z 1 Y z
G z
2z2 4z 1
X z
z 2 z 1 z 2 Y z
G z
2 4 z 1 z 2
X z
Y z 0.5 zX z X z 0.5 z 1 X z 0.5 z 2 X z 2 z 1Y z 0.5 z 2Y z
yn 0.5 xn 1 xn 0.5 xn 1 0.5 xn 2 2 yn 1 0.5 yn 2
So system is non-causal.
So, order of numerator polynomial in z Order of denominator
polynomial in z for causality.
STABILITY OF DTLTI SYSTEMS

An LTI system is BIBO stable if and only if its output
sequence yn is bounded for every bounded input xn.
If xn is bounded there exists a constant Bx such that
xn Bx
for all n
Similarly if yn is bounded, there exists a constant By such that,
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yn By
for all n
Constraint on impulse response

The response of a DTLTI system (with weighting sequence gn ) to
a bounded input xn , is expressed as
yn g m xn m
m
; n 0, 1, 2 etc.
yn g m xn m g m xn m
xn m Bx
yn Bx g m
m
For yn to be bounded, we should have
gm
or, g n
n
Thus the DTLTI system is BIBO stable if its impulse
response is absolutely summable.
Constraint on transfer function
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G z Z gn gn z n
n
G z gn z
gn z n
n
when evaluated on the unit circle (i.e.
z 1)
G z gn
n
g n should be finite.
If the system is BIBO stable n
Hence, G z evaluated on the unit circle, should be finite.

G z can only be finite in the ROC of G(Z).
A DTLTI system is BIBO stable if and only if the ROC of the
system transfer function includes the unit circle centred at origin in
the Z-plane.
The above is valid for both causal and non-causal systems.
For a causal system, however, the condition for stability can be
narrowed to a certain extent.
For a causal system ROC of G(z) is |z| > r

For a stable system ROC of G(z) must include the
unit
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For a causal and stable system the ROC of the
system function is |z| > r < 1.
Since the ROC can not contain any pole of G(z), it can be seen that
a causal DTLTI system is BIBO stable if and only if all the poles
of G(z) are inside the unit circle.
Example: A DTLTI system has transfer function
G z
3
4
1
1
1 z 1 1 2 z
4
Determine the ROC of G(z) and find gn for

i)
a stable system.
ii)
a causal system.
iii)
Purely anti-causal system.
Solution: Poles of G(z) are at

i)
1
4
and z = 2
ROC must include unit circle and poles can not be within
ROC. So ROC is
1
z 2.
4
g n is non-causal
n
n
1
g n 3 un 4 2 un 1
4
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ii)
For causality of gn , ROC of G(z) is |z| > 2
gn 3 1
un 4 2 un
n
Since ROC does not include unit circle,

G(z) is unstable.
iii)
gn is purely anti-causal. Hence ROC of G(z) is
1
4
n
1
g n 3 u n 1 4 2 u n 1
4
Since ROC does not include unit circle,

G(z) is unstable.
Systems
with
Finite-Duration
and
Infinite-Duration Impulse Response:

It is convenient to subdivide the class of discrete-time LTI
systems into two types, those that have a finite-duration
impulse response (FIR) and those that have an infiniteduration impulse response (IIR).
An FIR system has an impulse response that is zero outside
some finite interval of time. For a causal FIR system,
gn 0 for n 0 and n K
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The convolution formula for such a system is
K 1
m0
m0
m K
yn g m xnm g m xn m g m xn m
K 1
or,
yn g m xn m
m0
It is worth noting that that output of the system at any instant n,

is simply a weighted linear combination of input signal samples
xn, xn-1 ,., xn-K+1. The system acts as a window that views
only the most recent K input signal samples and sums the K
products. It neglects all prior input samples i.e., xn-K , xn-K-1, ..
Thus it is said that an FIR system has a finite memory of length
of K samples.
A IIR linear time-invariant system has an infinite-duration
impulse response. Output of a causal IIR system, based on the
convolution formula, is
yn g m xn m
m0
Here, the system output is a weighted linear combination of

input signal samples xn , xn-1 , xn-2 , ., xo . Since this
weighted sum involves the present and all the past input
samples, it is said that the system has an infinite memory.
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The convolution summation expression suggests a means for

ready implementation of FIR systems involving additions,
multiplications, and a finite number of memory locations. If the
system is IIR, its practical realization using convolution is
clearly impossible, since it requires an infinite number of
memory locations, multiplications and additions. A question
that naturally arises, then, is whether or not it is possible to
realize IIR systems other than in the form suggested by the
convolution summation. Fortunately there is a practical and
computationally efficient means for implementing a family of
IIR systems.
This family of IIR systems are more conveniently described

by difference equations. This subclass of IIR systems is very
useful in a variety of practical applications, including the
implementation of digital filters.
RECURSIVE AND NON-RECURSIVE SYSTEMS:

Definitions: A system whose output yn at instant n depends on
any number of past output values yn-1 , yn-2 ,.. is called a
recursive system.
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A system whose output yn at any instant n depends only on the

present and past inputs xn , xn-1, ., but not on any past
output value, is called a non-recursive system.
Discussions:
Let us consider a discrete-time system
represented by a linear constant coefficient difference equation

N
i 0
i 0
i yni i xn i
(1)
If N 0,
On rearranging the terms, we have
N
1 M
yn
i xni i yni
i 0
o i 0
(2)
If N 0,
This equation directly expresses the output at time n in terms of
previous values of input and output. It can be immediately
seen that in order to calculate yn, we need to know yn-1, yn-2
,, yn-N. An equation of this form is called a recursive
equation, since it specifies a recursive procedure for
determining the output in terms of the input and previous
outputs. The system is hence, called a recursive system.
In the special case, when, in equation (1), N is zero, the
equation reduces to
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i
yn xn i
i 0
o
M
(3)
In this case, yn is a function of the present and previous values

of the input. For this reason, equation (3) is called a nonrecursive equation, since we do not recursively use previously
computed values of the output to compute the present value of
the output. The system represented by the equation is known as
non-recursive system. The impulse response of the system is
given by
i
g n n i
i 0
o
M
ni will be 0 for n i
which is obtained as
n
, for 0 n M
g n o
0 ,
otherwise
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for n M , i can never be equal to n
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Note that the impulse response has finite duration, i.e. it is nonzero only over a finite time interval. Because of this property
the system is an FIR system.
The basic differences between non-recursive and

recursive discrete-time systems are illustrated below.
xn
F xn , xn1,....., xnM
yn
Non-recursive System
xn
F yn1,...., ynN,
yn
xn , xn1,....., xnM
delay (s)
Recursive System
The fundamental difference between the two systems is the

presence of feedback loop in the recursive system, which feed
back the output sample(s) of the system through a delay
element. The presence of this delay element is crucial for the
realizability of the system, since its absence would force the
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system to compute yn in terms of yn, which is not possible for

discrete-time systems.
There is another important difference between recursive and
non-recursive systems. Suppose we wish to compute the output
yno of a system when it is excited by an input applied at time n =
0. if the system is recursive, to compute yno, all the previous
values yo , y1 ,., yno-1 should be computed first. If the
system is non-recursive, we can compute yno immediately
without computing yno-1 , yno-2, It implies that the output of
a recursive system should be computed in order [i.e. yo , y1 , y2,
.], whereas for a non-recursive system, the output can be
computed in any order [e.g. y100 , y25, y2 , y150, etc.]. This feature
is desirable in some practical applications.
Impulse Response of Recursive Systems:

The general form of difference equation for recursive systems is
repeated below
M
i 0
i 1
yn bi xn i ai yni
(4)
By substituting xn = n , the impulse response of the system is

obtained as
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g n bn ai g n i
i 1
(5)
No simple interpretation for this result can be given.

However it can be noted that the impulse response can remain
non-zero for even large indices.
The recursive portion
continues to generate an output long after the b ns

are zero. Therefore recursive discrete-time systems have
infinite impulse response. The point can be illustrated by
considering the following simple causal recursive system.
yn bo xn a1 yn1
(6)
The response of the filter to a unit sample sequence is obtained

by substituting xn = n and is given by
g o bo
g1 a1bo
g 2 a12bo
g3 a13bo
g n a1nbo a1nboun
The response has clearly infinite duration.
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gn
a1>1
a1 = 1
bo
0<a1<1
The sketches of the impulse response show that the nonrecursive coefficient (bo) merely controls the amplitude of the
response at n = 0. The recursive coefficient (a1) determines the
shape of the response. If a1 is negative, the response alternates
sign. If a1 is between 0 and 1, the response exponentially decays
to zero. If a1 is greater than 1, the response grows. If a1 is
unity, the response has a steady value.
IDEAL DIGITAL-TO-ANALOG CONVERSION

Let x(t) be a band-limited analog signal with highest
frequency component fm = W/(2) and is sampled at a rate fs =
2fm to obtain a discrete-time signal x*(t).
Let us now examine how to reconstruct x(t) from x*(t).
Now,
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x t F
1
F
fs
2 f s
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X * over the range W to W
W 2 f s
*
jt
X e d
jn jt

e d
x n e
2 f s W n
1
where f = sampling period.
s
Inter-changing the order of integration and summation and
considering that
W 2 f m f s
sin W t t
sin W t n sin W t t

t n dt
W t n
W t t t n W t t
2 f s
x t
x n e j t n d
W
x n e jwt n e jwt n

n 2W
j
t
x n
n
sin W t n
W n
(1)
This gives the ideal interpolation formula for reconstructing x(t)

from x*(t).
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Therefore, in equation (1),
sin W t t
x(t ) x n t n
dt
W t t
n
x* t gi t t dt
x* t g i t
sin Wt
sin Wt
g
t
where i
Wt
t
sin
2
t
t
is the impulse
response function of the ideal-interpolator or ideal discrete-time

to
continuous-time
converter
or
ideal
digital-to-analog
converter.
The frequency response function of the ideal-interpolator is
s
s
for
Gi j gi t
2
2
otherwise.
0
Hence the ideal interpolator is nothing but an ideal low pass filter
with pass band of
s
2
and pass band gain of .
Hence the ideal interpolation formula may be used to

reconstruct x(t) from x*(t) if fs > 2fm .
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There are however two important problems associated with the

ideal interpolation formula.
Firstly it requires infinite numbers of additions and

multiplications. This is not feasible in practice.
.
Secondly, the computation can not be carried out in real time.

By this we mean that the computation of x(to) requires not only
x(n) for n to but also x(n) for n > to.
Hence we can start to compute x(t) only after all x(n) (n = 0,
1, 2, .) are known.
Because of these two reasons, equation (1) is not very useful
in the reconstruction of x(t) from x*(t).
PRACTICAL DAC
In practice a D/A converter employs a zero order hold (ZOH) for
reconstructing the analog signal from its sampled version.
The function of the DAC is to decode the input digital words
(binary numbers) and then convert the digital signal into an analog
signal. Hence a DAC can be represented by a decoder followed by
a ZOH.
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x*(t)
xr(t)
Decoder
ZOH
DAC
If we consider a DAC as a system, the decoder does not affect the

system transfer function. So we can only consider the ZOH.
x*(t)
xr(t)
ZOH
A zero-order hold (ZOH) holds a sample value constant

until receiving the next sample value.
xr(t)
Samples
There are discontinuities in the analog signal constructed from the

digital signal by a ZOH. This implies the presence of high
frequency components.
The output staircase waveform of the ZOH can be expressed
as
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xr t x n u t n u t n
n
n 0, 1, 2, 3,..............
Taking the Laplace Transform of both sides,
e n s e n 1 s
X r s x n
n
s
1 e s
n s
x
n
s n
Now,
x(n )e
n s
X * ( s) L[ x* (t )]
G ho ( s)
1-e-s
is the transfer function of the zero-order hold.
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[A]
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Now,
[B]
Thus it is often said that the ZOH introduces a time delay of

approximately half the sampling period.
Frequency response function of Z.O.H
1 - e -s
G ho ( s)
s
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Gho ( j )
Sin
[C]
Gho ( j ) Sin
or,
Gho ( j ) m -
m = 0,1,2,..
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[D]
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G j
ideal filter Gi j
ZOH Gho j
0.637
3s
2s
s
2
s
2
2s
3s
The nature of Gho ( j ) reveals that the ZOH allows high

frequency components of x*(t) to pass through.
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Sugata Munshi
Hence the ZOH (i.e. DAC) should be followed by a low pass

analog filter (post-filter) with cut off frequency
2 to remove
these high frequency components.

x*(t)
ZOH
Post
filter
xo(t)
xo(t)
xr(t)
xr(t)
Hence a complete DSP system can be represented as
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x(t)
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PreFilter
xb(t)
ADC
x*(t)
Signal
Processing
Algorithm
y*(t)
DAC
yr(t)
Post
Filter
yo(t)

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Sugata Munshi

Dept. of Electrical Engineering,

DISCRETE-TIME SIGNALS AND

Its values are known as

If the samples are uniformly spaced in time, we have a

One of the distinctive feature of digital signals is that, they

Another distinctive feature of digital signals is that their

Block diagram of a typical Digital Signal Processing System

The digitized analog signal is processed by a digital signal processor.

Guaranteed Accuracy determined only by

performance from unit to unit, since there is no

and reprogrammed to perform a variety of

algorithms can be implemented.

Tremendous advancements in electronics

Ideal switch( closes periodically at intervals of ,

is the sampling frequency.

A more realistic case

The duration of closure of the switch should be finite (<)

Sampling Process Modelled As Pulse

x*p(t) = x(t) p(t)

If the sampling duration is very small compared to the

sequence of flat-topped pulses.

Taking the Laplace transform on both sides ,

Taking inverse Laplace Transform of both sides,

for small each pulse becomes a vertical line.

Hence the finite pulsewidth sampler can be approximated

Here the role of is just scaling. Hence it is considered as unity.

Output of the impulse sampler is

Reconstruction of Continuous Time Signal from

In the absence of any additional conditions or

information, a continuous time signal can not be

signals that can generate the given set of samples have

sinusoid from its samples:

Let a continuous-time signal

be sampled with a sampling period .

If the sampling starts at a time angle 2 fto from

is the time period in number of samples.

>> 1, i.e., fs >> f, << T

The analog sinusoid x(t) can be uniquely reconstructed from

< 1, i.e., fs < f, > T

Thus, since x n is also the sampled version

Reconstruction gives a steady value.Samplings at zero

x n is also sampled version of sinusoid with freq.

x n , will have a frequency f s f f .

So, frequency f of original signal is folded back into the

x n can also be represented by

B sin n , i.e. B sin 2

So, x n can also be sampled version of sinusoid

Infinite no. of combinations of B &

it is possible to have infinite number of

reconstructions of analog sinusoids of same frequency f, but

with f s 2 f , original sinusoid can not be uniquely

x n can be sampled version of x(t) as well as of analog

Sinusoid x(t ) A sin 2 ft can be uniquely reconstructed

from t =0, such that

y t A sin 2 f Mf s t (where M is a +ve integer)

Not possible to distinguish between the

Frequency f Mf s (M=1,2,3,.) is folded back into

are folded back into

Fourier Transform of x(t) is

Consider the periodic signal

can be expanded into Fourier series as