INSTRUMENTACIN

INDUSTRIAL Y CONTROL
AUTOMTICO

Ing. Hctor Marcial

PROCESAMIENTO DIGITAL
DE SEALES
Tomado de:
Digital Signal Processing
Engr. Abdul Rauf Khan Rajput

Engr. A. R.K. Rajput NFC IET Multan

Parte 1
Definiciones
Aplicaciones

 Signal:
A signal is defined as a function of one or more variables
which conveys information on the nature of a physical
phenomenon. The value of the function can be a real
valued scalar quantity, a complex valued quantity, or
perhaps a vector.

System:
A system is defined as an entity that manipulates one or
more signals to accomplish a function, thereby yielding
new signals.

 Continuos-Time Signal:
A signal x(t) is said to be a continuous time signal if it is
defined for all time t.

Discrete-Time Signal:
A discrete time signal x[nT] has values specified only at
discrete points in time.

Signal Processing:
A system characterized by the type of operation that it
performs on the signal. For example, if the operation is
linear, the system is called linear. If the operation is nonlinear, the system is said to be non-linear, and so forth.
Such operations are usually referred to as Signal
Processing.
5

Basic Elements of a Signal Processing

System
Analog input
signal

Analog output
signal

Analog
Signal Processor
Analog Signal Processing

Analog
input
signal

A/D
converter

Digital
Signal Processor
Digital Signal Processing

D/A
converter

Analog
output
signal

 Advantages of Digital over Analogue Signal

Processing:
A digital programmable system allows flexibility in
reconfiguring the DSP operations simply by changing the
program. Reconfiguration of an analogue system usually
implies a redesign of hardware, testing and verification
that it operates properly.
DSP provides better control of accuracy requirements.
Digital signals are easily stored on magnetic media (tape
or disk).
The DSP allows for the implementation of more
sophisticated signal processing algorithms.
In some cases a digital implementation of the signal
processing system is cheaper than its analogue
7
counterpart.

DSP Applications
Space

Medical

Space photograph enhancement

Data compression
Intelligent sensory analysis
Diagnostic imaging (MRI, CT,
ultrasound, etc.)
Electrocardiogram analysis
Medical image storage and retrieval

Image and sound compression for

Commercial multimedia presentation.
Movie special effects
Video conference calling

Telephone

Video and data compression

echo reduction
signal multiplexing
filtering
8

DSP Applications (cont.)

Military

Radar
Sonar
Ordnance Guidance
Secure communication

Oil and mineral prospecting

Industrial Process monitoring and control
Non-destructive testing

Scientific

Earth quick recording and analysis

Data acquisition
Spectral Analysis
Simulation and Modeling

Classification of Signals
Deterministic Signals
A deterministic signal behaves in a fixed known way with
respect to time. Thus, it can be modeled by a known
function of time t for continuous time signals, or a known
function of a sampler number n, and sampling spacing T
for discrete time signals.

Random or Stochastic Signals:

In many practical situations, there are signals that either
cannot be described to any reasonable degree of accuracy
by explicit mathematical formulas, or such a description
is too complicated to be of any practical use. The lack of
such a relationship implies that such signals evolve in time
in an unpredictable manner. We refer to these signals as
random.
10

Even and Odd Signals

A continuous time signal x(t) is said to an even signal if it
satisfies the condition
x(-t) = x(t) for all t
The signal x(t) is said to be an odd signal if it satisfies the
condition
x(-t) = -x(t)
In other words, even signals are symmetric about the
vertical axis or time origin, whereas odd signals are
antisymmetric about the time origin. Similar remarks
apply to discrete-time signals.
Example:

even
odd

odd

11

Periodic Signals
A continuous signal x(t) is periodic if and only if there
exists a T > 0 such that
x(t + T) = x(t)
where T is the period of the signal in units of time.
f = 1/T is the frequency of the signal in Hz. W = 2/T is the
angular frequency in radians per second.
The discrete time signal x[nT] is periodic if and only if
there exists an N > 0 such that
x[nT + N] = x[nT]
where N is the period of the signal in number of sample
spacings.
Example:
Frequency = 5 Hz or 10 rad/s
0

0.2

0.4

12

Continuous Time Sinusoidal Signals

A simple harmonic oscillation is mathematically
described as
x(t) = Acos(wt + )
This signal is completely characterized by three
parameters:
A = amplitude, w = 2f = frequency in rad/s, and =
phase in radians.
A

T=1/f

13

Discrete Time Sinusoidal Signals

A discrete time sinusoidal signal may be expressed as
x[n] = Acos(wn + )
- < n <
Properties:
A discrete time sinusoid is periodic only if its frequency is a
rational number.
Discrete time sinusoids whose frequencies are separated by
an integer multiple of 2 are identical.

1
0
-1

10

14

Parte 2
Operaciones entre Seales

15

Basic Operations on Signals

(a) Operations performed on dependent
variables
1. Amplitude Scaling:
let x(t) denote a continuous time signal. The signal y(t)
resulting from amplitude scaling applied to x(t) is
defined by
y(t) = cx(t)
where c is the scale factor.
In a similar manner to the above equation, for discrete
time signals we write
2x(t)
y[nT] = cx[nT]
x(t)
16

(b) Operations performed on independent

variable
Time Scaling:
Let y(t) is a compressed version of x(t). The signal y(t)
obtained by scaling the independent variable, time t, by
a factor k is defined by
y(t) = x(kt)
if k > 1, the signal y(t) is a compressed version of
x(t).
If, on the other hand, 0 < k < 1, the signal y(t) is an
expanded (stretched) version of x(t).

17

Example of time scaling

1
0.9
0.8
0.7

Expansion and compression of the signal e-t.

exp(-t)

0.6
0.5
0.4

exp(-2t)
exp(-0.5t)

0.3
0.2
0.1
0
0

10

15 18

Time scaling of discrete time systems

x[n]

10
5

0
-3
10

-1

0
-1.5
5

-1

-0.5

0.5

1.5

0
-6

-4

-2

0
n

x[2n]

x[0.5n]

-2

19

Time Reversal

This operation reflects the signal about t = 0

and thus reverses the signal on the time scale.
x[n]

-5
0

x[-n]

0
0
0

2
n

20

Time Shift
A signal may be shifted in time by replacing the independent variable n by n-k,
where k is an integer. If k is a positive integer, the time shift results in a delay of
the signal by k units of time. If k is a negative integer, the time shift results in an
advance of the signal by |k| units in time.

x[n+3]

10

x[n-3]

x[n]

1
0.5
0 -2
1
0.5
0 -2
1
0.5
0 -2

10

n4

10

21

2. Addition:
Let x1 [n] and x2[n] denote a pair of discrete time signals. The signal y[n] obtained by the addition of
x1[n] + x2[n] is defined as
y[n] = x1[n] + x2[n]

Example: audio mixer

3. Multiplication:
Let x1[n] and x2[n] denote a pair of discrete-time signals.
The signal y[n] resulting from the multiplication of the
x1[n] and x2[n] is defined by
y[n] = x1[n].x2[n]

Example: AM Radio Signal

22

Parte 3
Conversin
DSP

23

Analog to Digital and Digital to Analog

Conversion
A/D conversion can be viewed as a three step
process
1. Sampling: This is the conversion of a continuous time signal into a discrete time signal
obtained by taking samples of the continuous time signal at discrete time instants. Thus, if
x(t) is the input to the sampler, the output is x(nT), where T is called the Sampling interval.

2. Quantization: This is the conversion of discrete time

continuous valued signal into a discrete-time discrete-value
(digital) signal. The value of each signal sample is
represented by a value selected from a finite set of possible
values. The difference between unquantized sample and
the quantized output is called the Quantization error.
24

Analog to Digital and Digital to Analog

Conversion (cont.)

3. Coding:

In the coding process, each discrete value is

represented by a b-bit binary sequence.

x(t)

0101...
Sampler

Quantize r

Coder

A/D Converter

25

What is DSP?
Converting a continuously changing waveform
(analog) into a series of discrete levels (digital)

26

What is DSP?
The analog waveform is sliced into equal
segments and the waveform amplitude is
measured in the middle of each segment
The collection of measurements make up
the digital representation of the waveform

27

-1

-1.5
17

15

13

11

0.5

0.22
0.44
0.64
0.82
0.98
1.11
1.2
1.24
1.27
1.24
1.2
1.11
0.98
0.82
0.64
0.44
0.22

1.5

19
-0.22
-0.44 21
-0.64
-0.82
23
-0.98
-1.11
25
-1.2
-1.26
27
-1.28
-1.26
29
-1.2
-1.11
31
-0.98
-0.82
33
-0.64
-0.44 35
-0.22
37

-0.5
0

0
1

What is DSP?

-2
28

Converting Analog into Digital

Electronically(1/3)

The device that does the conversion is

called an Analog to Digital Converter
(ADC)
There is a device that converts digital to
analog that is called a Digital to Analog
Converter (DAC)

29

Converting Analog into Digital

Electronically(2/3)
The simplest form of
ADC uses a resistance
ladder to switch in the
appropriate number of
resistors in series to
create the desired
voltage that is
compared to the input
(unknown) voltage

SW-8
V-high
SW-7
V-7
SW-6
V-6
SW-5
Output

V-5
SW-4
V-4
SW-3
V-3
SW-2
V-2
SW-1
V-1

V-low

30

Converting Analog into Digital

Electronically(3/3)
The output of the
resistance ladder is
compared to the
Analog Voltage
analog voltage in a
Resistance
comparator
Ladder Voltage
When there is a match,
the digital equivalent
(switch configuration)
is captured

Comparator
Output

Higher
Equal
Lower

31

Converting Analog into Digital

Computationally(1/2)
The analog voltage can now be compared with the
digitally generated voltage in the comparator
Through a technique called binary search, the
digitally generated voltage is adjusted in steps
until it is equal (within tolerances) to the analog
voltage
When the two are equal, the digital value of the
voltage is the outcome
32

Converting Analog into Digital

Computationally(2/2)
The binary search is a mathematical technique that
uses an initial guess, the expected high, and the
expected low in a simple computation to refine a
new guess
The computation continues until the refined guess
matches the actual value (or until the maximum
number of calculations is reached)
The following sequence takes you through a
binary search computation
33

Binary Search
Initial conditions

Expected high 5-volts

Expected low 0-volts
5-volts 256-binary
0-volts 0-binary

Analog
5-volts
3.42-volts
2.5-volts

Digital
256
Unknown
(175)
128

Voltage to be converted
3.42-volts
Equates to 175 binary

0-volts

34

Binary Search
Binary search algorithm:
High Low
Low NewGuess
2

First Guess:

Analog
5-volts

Digital
256

3.42-volts

unknown
128

256 0
0 128
2
Guess is Low

0-volts

0
35

Binary Search
New Guess (7):

Analog
5-volts

3.42-volts
176 172
172 174
2

Digital
256
unknown
174

Guess is Low
(but getting really,
really, close)

0-volts

0
36

Binary Search
New Guess (8):

176 174
174 175
2

Analog
5-volts
3.42-volts

Guess is Right On
0-volts

Digital
256

175!

0
37

Binary Search
The speed the binary search is
accomplished depends on:
The clock speed of the ADC
The number of bits resolution
Can be shortened by a good guess (but usually
is not worth the effort)

38

How Does It Work?

Faithful Duplication

Now that we can slice up a waveform and

convert it into digital form, lets take a look
at how it is used in DSP
Draw a simple waveform on graph paper
Scale appropriately

Gather digital data points to represent the

waveform
39

Starting Waveform Used to

Create Digital Data

40

Waveform Created from Digital Data

41

How Does It Work?

Faithful Duplication

Once the waveform is in digital form, the

real power of DSP can be realized by
mathematical manipulation of the data
Using EXCEL spreadsheet software can
assist in manipulating the data and making
graphs quickly
Lets first do a little filtering of noise
42

1/6 Sample Rate

every 6th

150

150

100

100

50
0
-50 0

10

20

30

40

Amplitude

Amplitude

Raw

50
0
-50 0

-100

-100

-150

-150
Time

10

20

30

40

Time

43

1/12 Sample Rate

every 12th

150

150

100

100

50
0
-50 0

10

20

30

40

Amplitude

Amplitude

Raw

50
0
-50 0

-100

-100

-150

-150

Time

10

20

30

40

Time

44

How Does It Work?

Resolution Trade-offs
Bit
Resolution

Sample Rate

High Bit
Count

Good
Duplication

Slow

Low Bit
Count

Poor
Duplication

Fast

High Sample
Rate

Good
Duplication

Slow

Low Sample
Rate

Poor
Duplication

Fast

45

Digital Signal Processing

Lecture -2

46

Sampling of Analog Signals

x[n] = x[nT]

analog signal

1
0.8
0.6
0.4
0.2
0
-0.2
-0.4
-0.6
-0.8
-1
0

1
0.8
0.6
0.4
0.2
0
-0.2
-0.4
-0.6
-0.8
-1
0

sampled signal

Uniform Sampling:

47

Uniform sampling
Uniform sampling is the most widely used sampling scheme.
This is described by the relation
x[n] = x[nT]
- <n<
where x(n) is the discrete time signal obtained by taking samples of the analogue signal x(t) every T seconds.
The time interval T between successive symbols is called the Sampling Period or Sampling interval and its reciprocal 1/T = Fs is called the Sampling Rate (samples per second) or the Sampling Frequency (Hertz).
A relationship between the time variables t and n of continuous time and discrete time signals respectively, can be obtained as

n
t nT
Fs

(1)

48

 A relationship between the analog frequency F and the

discrete frequency f may be established as follows.
Consider an analog sinusoidal signal
x(t) = Acos(2Ft + )
which, when sampled periodically at a rate Fs = 1/T samples
per second, yields
2nF

x[nT] A cos 2FnT A cos

Fs

(2)

But a discrete sinusoid is generally represented as

x[n] A cos 2fn

(3)

Comparing (2) and (3) we get

F
f
Fs

(4)
49

Since the highest frequency in a discrete time signal is f = .

Therefore, from (4) we have
Fmax

Fs
1

2
2T

(5)

or

Fs = 2 Fmax

(6)

Sampling Theorem:
If x(t) is bandlimited with no components of frequencies greater
than Fmax Hz, then it is completely specified by samples taken at
the uniform rate Fs > 2Fmax Hz.
The minimum sampling rate or minimum sampling frequency,
Fs = 2Fmax, is referred to as the Nyquist Rate or Nyquist
50
Frequency. The corresponding time interval is called the Nyquist

Sampling Theorem (cont.)

Signal sampling at a rate less than the Nyquist rate is
referred to as undersampling.
Signal sampling at a rate greater than the Nyquist rate is
known as the oversampling.
Example 1:
The following analogue signals are sampled at a sampling frequency of 40
Hz. Find the corresponding discrete time Signals.
(i) x(t) = cos2(10)t (ii) y(t) = cos2(50)t
Solution:
(i)
10

x1[ n] cos 2

n cos
n
2

40

(ii)

50
n

cos
n

cos(
2

n
/
2
)

cos
n

2
2
40

x2 [ n] cos 2

As, Shows identical in [ x1(n) & x2(n)] sinusoidal signals & indistinguishable. Ambiguity
is there for samples values. x(t) yield same values as y(t) when two are sampled at Fs=40,
then

Note: The frequency F2 = 50 Hz is an alias of F1 = 10 Hz. All of the

51
sinusoids cos2(F1 + 40k)t, t = 1,2,3, are aliases.

Some Elementary Discrete Time signals

Unit Impulse or unit sample sequence:
It is defined as
1,
0

n0
n 0

In words, the unit sample sequence is a signal that is zero

everywhere, except at t = 0.
1
0.8
0.6
0.4
0.2
0
-3

-2

-1

Unit impulse function

52

Some Elementary Discrete Time signals

Unit step signal
It is defined as
1,
u[n ]
0

2
1.8
1.6
1.4
1.2
1
0.8
0.6
0.4
0.2
00

n0
n0

7
53

Some Elementary Discrete Time signals

Unit Ramp signal
It is defined as
n, n 0
r[n]
0 n0

6
5
4
3
2
1
00

54

Some Elementary Discrete Time signals

Exponential Signal
The exponential signal is a sequence of the form
x[n] = an,
for all n
If the parameter a is real, then x[n] is a real signal. The
following figure illustrates x[n] for various values of
a.
0<a<1

-1<a<0

a>1
a<-1
55

Some Elementary Discrete Time signals

Exponential Signal (cont)
when the parameter a is complex valued, it can be expressed as
j
where r and are now the parameters. Hence we may express
x[n] as

a re

x[n] r n e j r n cos n j sin n

Since x[n] is now complex valued, it can be represented

graphically by plotting the real part

x
[
n
]

r
cos

n
R
as a function of n, and separately plotting the imaginary part
n

asx a[nfunction
] r n sinofnn. (see plots on the next slide)
I

56

xR[n] = (0.9)ncos(n/10)

0.5
0
-0.5
0

10

20

30

40

50

60

50

60

1
xI[n] = (0.9)nsin(n/10)

0.5
0
-0.5
0

10

20

30

40

57

Exponential Signal (cont.)

|x[n]|

Alternatively, the signal x[n] may be graphically represented by the

amplitude or magnitude function
|x[n]| = rn
and the phase function
[n] = n
The following figure illustrates |x[n| and [n] for r = 0.9 and = /10.

[n]

10

10

0
-
-

58

Discrete Time Systems

A discrete time system is a device or algorithm that operates
on a discrete time signal x[n], called the input or excitation,
according to some well defined rule, to produce another
discrete time signal y[n] called the output or response of the
system.
We express the general relationship between x[n] and y[n] as
y[n] = H{x[n]}
where the symbol H denotes the transformation (also called
an operator), or processing performed by the system on x[n]
to produce y[n].
x[n]

Discrete Time System

H

y[n]
59

Classification of Discrete Time Systems

Static versus Dynamic Systems

A discrete time system is called static or memory-less if its
output at any instant n depends at most on the input sample
at the same time, but not on the past or future samples of the
input. In any other case, the system is said to be dynamic or
to have memory.
Examples: y[n] = x2[n] is a memory-less system, whereas the
following are the dynamic systems:
(a) y[n] = x[n] + x[n-1] + x[n-2]
(b) y[n] = 2x[n] + 3x[n-4]
60

Time Invariant versus Time Variant Systems

A system is said to be time invariant if a time delay or time
advance of the input signal leads to an identical time shift in
the output signal. This implies that a time-invariant system
responds identically no matter when the input is applied.
Stated in another way, the characteristics of a time invariant
system do not change with time. Otherwise the system is said
to be time variant.
Example1: Determine if the system shown in the figure is
time invariant or time variant.
y[n]
Solution: y[n] = x[n] x[n-1]
x[n]
+
Now if the input is delayed by k units
in time and applied to the system, the
-1
Z
Output is
y[n,k] = n[n-k] x[n-k-1]
(1)
On the other hand, if we delay y[n] by k units in time, we obtain
y[n-k] = x[n-k] x[n-k-1]
(2)
61
(1) and (2) show that the system is time invariant.

Time Invariant versus Time Variant Systems

Example 2: Determine if the following systems are time invariant or

time variant.
(a) y[n] = nx[n] (b) y[n] = x[n]cosw0n
Solution:
(a) The response to this system to x[n-k] is
y[n,k] = nx[n-k]
(3)
Now if we delay y[n] by k units in time, we obtain
y[n-k] = (n-k)x[n-k]
= nx[n-k] kx[n-k]
(4)
which is different from (3). This means the system is time-variant.
(b) The response of this system to x[n-k] is
y[n,k] = x[n-k]cosw0n
(5)
If we delay the output y[n] by k units in time, then
y[n-k] = x[n-k]cosw0[n-k]
which is different from that given in (5), hence the system is time
variant.
62

Linear versus Non-linear

Systems
A system H is linear if and only if
H[a1x1[n] + a2x2[n]] = a1H[x1[n]] + a2H[x2[n]]
for any arbitrary input sequences x1[n] and x2[n], and any
arbitrary constants a1 and a2.
a1
x1[n]
y1[n]
+
H
a2
x2[n]
x1[n]
x2[n]

H
H

a1
a2

If y1[n] = y2[n], then H is linear.

y2[n]

63

Causal versus Noncausal Systems

A system is said to be causal if the output of the system at
any time n [i.e. y[n]) depends only on present and past
inputs but does not depend on future inputs.
Example: Determine if the systems described by the
following input-output equations are causal or
noncausal.
(a) y[n] = x[n] x[n-1] (b) y[n] = ax[n] (c) y[n] x[k ]
(d) y[n] = x[n] + 3x[n+4] (e) y[n] = x[n2]
(f) y[n] = x[-n]
Solution: The systems (a), (b) and (c) are causal,
others are non-causal.
n

64

Stable versus Nonstable Systems

A system is said to be bonded input
bounded output (BIBO) stable if and
only if every bounded input produces a
bounded output.

65

z-transform
Transform techniques are an important role in the analysis of
signals and LTI system.
Z- transform plays the same role in the analysis of discrete time
signals and LTI system as Laplace transform does in the analysis of
continuous time signals and LTI system.
For example, we shall see that in the Z-domain (complex Z-plan)
the convolution of two time domain signals is equivalent to
multiplication of their corresponding Z-transform.
This property greatly simplifies the analysis of the response of LTI
system to various signals.
DSP

Slide 66

1-The Direct Z- Transform

The z-transform of a sequence x[n] is

X ( z ) x[ n ] z

Where z is a complex variable. For convenience, the z-transform of a

signal x[n] is denoted by
X(z) = Z{x[n]}

We may obtain the Fourier transform from the z transform by

making the substitution X ( z ) e . This corresponds to
restricting z 1 Also with z r e j
,
j

X (r e

x[n](r e

) n

That is, the z-transform is the Fourier transform of the sequence x[n]r - n . for r=1
this becomes the Fourier transform of x[n].
The Fourier transform therefore corresponds to the z-transform evaluated on the
unit circle:
DSP

Slide 67

z-transform(cont:

The inherent periodicity in frequency of the Fourier transform

is captured naturally under this interpretation.
The Fourier transform does not converge for all sequences - the infinite
sum may not always be finite. Similarly, the z-transform does not
converge for all sequences or for all values of z.
For any Given sequence the set of values of z for which the z-transform
converges is called the region of convergence (ROC).
DSP

Slide 68

z-transform(cont:

The Fourier transform of x[n] exists if the sum n x[ n ]

converges. However, the z-transform of x[n] is just the Fourier
transform of the sequence x[n]r -n. The z-transform therefore exists

(or converge) if
n

X ( z)

This leads to the condition

x[ n]r

x[ n] z

for the existence of the z-transform. The ROC therefore consists of a

ring in the z-plane:

In specific cases the inner radius of this ring may include the origin, and the outer
radius may
, then
DSP extend
Slide 69 to infinity. If the ROC includes the unit circle
z 1
the Fourier transform will converge.

z-transform(cont:
Most useful z-transforms can be expressed in the form
X ( z)

P( z )
,
Q( z )

where P(z) and Q(z) are polynomials in z. The values of z for

which P(z) = 0 are called the zeros of X(z), and the values with
Q(z) = 0 are called the poles. The zeros and poles completely
specify X(z) to within a multiplicative constant.
In specific cases the inner
radius of this ring may include
the origin, and the outer radius
may extend to infinity. If the
z 1
ROC includes the unit circle
, then the Fourier
transform will converge.
DSP

Slide 70

Example: right-sided exponential sequence

Consider the signal x[n] = anu[n]. This has the z-transform

X ( z)

a u[n]z (az )
n

n 0

Convergence requires that

az 1

which is only the case if

az 1 1.or

equivalently z a .

In the ROC, the series converges to

1
z
X ( z ) (az )

, z a,
1
n 0
1 az
za

1 n

since it is just a geometric series.

DSP

Slide 71

2-Properties of the region of convergence

The properties of the ROC depend on the nature of the signal. Assuming that the
signal has a finite amplitude and that the z-transform is a rational function:
The ROC is a ring or disk in the z-plane, centered on the origin

(0 R z L ).

The Fourier transform of x[n] converges absolutely if and only if the ROC of
the z-transform includes the unit circle.
The ROC cannot contain any poles.
If x[n] is finite duration (ie. zero except on finite interval ( N1 n N 2 ).
at z=0 or
), then the ROC is the entire Z-plan except perhaps
z=
.
If x[n] is a right-sided sequence then the ROC extends outward from the
outermost finite pole to infinity.
If x[n] is left-sided then the ROC extends inward from the innermost nonzero
pole to z = 0.
A two-sided sequence (neither left nor right-sided) has a ROC consisting of a
ring in the z-plane, bounded on the interior and exterior by a pole (and not
containing any poles).
The
ROC
DSP
Slideis72a connected region.

3 - The inverse z-transform

Formally, the inverse z-transform can be performed by evaluating a

Cauchy integral. However, for discrete LTI systems simpler methods
are often sufficient.

A-Inspection method: If one is familiar with (or has a table

of) common z-transform pairs, the inverse can be found by
inspection. For example, one can invert the z-transform

X ( z)

1
1
1
z 1
2

Using Z-transform pair

, z

1
2,

1
a u[ n ]

,........ for z a .
1 az 1
By inspection we recognise that
n
1
x[n]
u[ n ],

2
n

Also, if X(z) is a sum of terms then one may be able to do a term-byterm DSP
inversion
Slide 73 by inspection, yielding x[n] as a sum of terms.

3 - The inverse z-transform

B-Partial fraction expansion:

For any rational function we can obtain a partial fraction expansion,
and identify the z-transform of each term. Assume that X(z) is
expressed as a ratio of polynomials in z-1:

X ( z)

M
k 0
N

bk z

k
k

ak z
It is always possible to factorX(z) as
k 0

X(z)

b0
a0

1 c z
1 d z
M

k 1
N

k 1

where the ck' s are the nonzero and poles of X(z).

DSP

Slide 74

The
inverse
z-transform
Partial fraction expansion (Continue:)

If M<N and the poles are all first order, then X(z) can be expressed
N
Ak
as
X(z)
,
1
k 1 1 d k z
in this case the coefficients A k are given by

A k 1 d k z 1 X ( z )

z dk

If M>N and the poles are first order, then an expression of the form
cab be used, and Brs be obtained by long division of the numerator.
M-N

X(z) Br z

Ak
1

1 dk z
The A k ' s can be obtained using M N
r 0

DSP

Slide 75

k 1

3 - The inverse z-transform Partial fraction expansion

The most general form for partial fraction expansion,

which can also deal with multiple - order poles, is

X(z)

M-N

B z
r 0

k 1, k i

Ak
1 dk z

Cm

m 1

1 d z

1 m

Ways of finding the C m ' s can be found in most standard

DSP texts. The terms B r z r correspond to shifted and
scaled impulse sequences, and invert to terms of the
form B r [n - r]. The fractional term s
A
k

1 d k z 1
correspond to exponentia l sequences. For these terms the
ROC properties must be used to decide whether the sequences
are left - sided or right - sided.
DSP

Slide 76

C-

Power Series Expansion

If Z transform is given as power series in form

X z

x[n] z

.................. [ 2] z x[ 1] z 1 x[0] x[1] z 1 [ 2] z ......

then any value in the sequence can be found by identifying the

coefficient of the appropriate power of z-1.

DSP

Slide 77

4- Properties of the z-transform

if X(z) denotes the z-transform of a sequence x[n] and the ROC of X(z) is
indicated by Rx, then this relationship is indicated as

x[ n]
z X ( z ),

ROC

Rx

Furthermore, with regard to nomenclature, we have two sequences such that

x1 [ n]
z X 1 ( z ),
x2 [ n]
z X 2 ( z ),

ROC
ROC

R x1
R x2

ALinearity: The linearity property is as follows:

ax1[n] bX 2 (n)
z aX 1[ z ] bX 2 ( z ),

ROC contains R x1 R x1 .

BTime Shifting: The time shifting property is as follows:

x[n n0 ]
z z X ( z ),
n0

ROC R x

(The ROC may change by the possible addition or deletion of z =0 or z = .)

This is easily shown:

Y ( z ) x[ n n ] z
0

DSP

n0

Slide 78

x[ m] z

n0

x[ m] z
n

X ( z ).

( m n0 )

C-

Multiplication by an exponential sequence

The exponential multiplication property is

z
x
[
n
]

X [ z / z0 ],
z0
n

ROC z0 R x ,

where the notation z 0 Rx , indicates that the ROC is scaled by z (that is,
inner and outer radii of the ROC scale by z ). All pole-zero locations are
similarly scaled by a factor z0: if X(z) had a pole at z z then X(z/z0)
will have a pole at z=z0z1.
0

If z0 is positive and real, this operation can be interpreted as a shrinking or

expanding of the z-plane | poles and zeros change along radial lines in the zplane.
If z0 is complex with unit magnitude (z0 = ejw0) then the scaling operation
corresponds to a rotation in the z-plane by and angle w0, That is, the poles and
zeros rotate along circles centered on the origin. This can be interpreted as a
shift in the frequency domain, associated with modulation in the time domain
by ejw0n. If the Fourier transform exists, this becomes

e x[n] X e
j 0 n

DSP

Slide 79

j ( 0 )

D-

Differentiation

The differentiation property states that

dX ( z )
,
dz

nx[n]
z z

ROC R x .

This can be seen as follows: since

X ( z)

x[ n ] z

n -

We have

dX ( z )
z
z (n) x[n]z n1 nx[n]z n z{nx[n]}.
n
dz

Example: second order pole

The z-transform of the sequence
Can be found
n

a u[ n]
z
to be

1
1 z 1

x[n] na n u[n]
z a,

d
1
az 1

X(z)

1
1 2
DSP Slide 80
dz 1 az
1 az

z a.

E-

Conjugation

This property is

x * [n] z X * ( z*),
FHere

ROC R x .

Time reversal.

x * [ n]
z X * (1 / z*),

ROC

1
.
Rx

The notation 1/Rx means that the ROC is inverted, so if Rx is the set
of values such that rR z rL , then the ROC is the set of values of z su
that 1 / r l z 1/rR .

Example: Time-reversed exponential sequence

The Signal x[ n] a n u[ n] is a time-reversed version of a nu[n]. The

z-transform is therefore

1
a z
X ( z)

,
1 1
1 az 1 a z
1 1

DSP

Slide 81

z a Rx.
1

G-

Convolution

This property state that

x1[n] * x2 [n]
z X 1 ( z ) X 2 ( z ),
Here

ROC contains

x * [ n]
z X * (1 / z*),

ROC

R x1 R x2 .

1
.
Rx

Example: evaluating a convolution using the z-transform

The z-transforms of the signal x1[n] =anu[n] and x2[n] = u[n] are

X 1 ( z)

a n z n
n 0

and

1
,
1
1 az

z a

1
,
z 1
1
n 0
1 az
For a 1, The z-transforms of the convolution y[n] = x 1[n] *x2[n] is
1
z2
Y ( z)

z 1
1
1
1 az 1 az z a z 1
.X 2 ( z)

1
Y DSP
( z ) Slide
82

1
1
1 az 1 az

z2
z a z 1

z 1

Some common z-transform pairs are:

DSP

Slide 83

I-

Relationship with the Laplace transform:

Continuous-time systems and signals are usually described by the Laplace

transform. Letting z = esT , where s is the complex Laplace variable

s d j ,
we
have
( d j ) T
dT
j T
z e
e e
.

Therefore
z e dT and z T 2f/f s 2 / s ,
where ws is the sampling frequency. As varies from to , the s-plane is
mapped to the z-plane:
The j axis in the s-plane is mapped to the unit circle in the z-plane.
The left-hand s-plane is mapped to the inside of the unit circle.
The right-hand s-plane maps to the outside of the unit circle.
DSP

Slide 84