History

The z-transform was introduced, under this name, by Ragazzini and 

Zadeh in 1952. The modified or advanced Z-transform was later
developed by E. I. Jury, and presented in his bookSampled-Data Control
Systems (John Wiley & Sons 1958). The idea contained within the Z-
transform was previously known as the "generating function method."
 Why z-
Transform
A generalization?of Fourier transform
Why generalize it?
 FT does not converge on all sequence
 Notation good for analysis
 Bring the power of complex variable theory deal with
the discrete-time signals and systems

In mathematics and signal processing, the Z-transform converts a discrete time-domain signal, which

is a sequence of real or complex numbers, into a complex frequency-domain representation.
It can be considered as a discrete equivalent of the Laplace transform. This similarity is explored in the
theory of time scale calculus.
The z-transform is an extension of the discrete time. Fourier transform to a function that is defined on
regions of the complex plane. Some sequences that do not have Fourier transforms will have z-
transforms, but the reverse is true as well. For our purposes, the special significance of z transform is
as a tool to facilitate designing systems with the desired attributes that can be practically implemented.
 X(z) =  x[n] z –n

linear summing time

difference domain
equation solution

z transform
inverse z transform

z transformed algebra z transform

equation solution
Relation btw various
transforms
 z-plane
The z-transform is a function of the complex z variable
Convenient to describe on the complex z-plane
If we plot z=ej for =0 to 2 we get the unit circle
Im
X(z)

z = e  j

Re

Im

Re
Region of Convergence

The region of convergence (ROC) is
the set of points in the complex plane
for which the Z-transform summation
converges.

 
| X ( z ) |  x (
n  
n ) z n
  | x
n  
( n ) || z | n

Right Sided Sequence
 P( z ) where P(z) and Q(z) are
X ( z)  polynomials in z.
Q( z )

Zeros: The values of z’s such that X(z) = 0

Poles: The values of z’s such that X(z) = 

Properties of ROC
Most real signals are analog and in order to utilise the processing power of
modern digital processors it is necessary to convert these analog signals
into some form which can be stored and processed by digital devices. The
standard method is to sample the signal periodically and digitize it with
an A to D converter using a standard number of bits 8, 16 etc. Digital
signal processing is primarily concerned with the processing of these
sampled signals.
A mathematical representation of the sampled signal is shown below.
This is equivalent to modulating a train of delta functions by the
analog signal. The delta function effectively "filters" out the values of
the signal at times corresponding to the zeros in the argument of the
delta function. This process is also referred to as "ideal" sampling
since it results in sampled signals of "zero" width and whose
spectrum is perfectly periodic.
(1)   (2)

This polynomial is called a Z transform. What is the meaning

of Z in this polynomial? The meaning is not that Z should take
on some numerical value; the meaning of Z is that it is the unit
delay operator. For example the coefficients of ZB(Z) = Z + 2Z2
- Z4 -Z5 are plotted in Figure 2. It is the same waveform as in
Figure 1, but it has been delayed.
Digital Filter
 In electronics, computer science and mathematics, a digital filter is a
system that performs mathematical operations on a sampled,
discrete-time signal to reduce or enhance certain aspects of that signal.
 A digital filter system usually consists of an analog-to-digital converter
to sample the input signal, followed by a microprocessor and some
peripheral components such as memory to store data and filter
coefficients etc. Finally a digital-to-analog converter to complete the
output stage. Program Instructions (software) running on the
microprocessor implement the digital filter by performing the necessary
mathematical operations on the numbers received from the ADC.
 Digital filters are commonplace and an essential element of everyday
electronics such as radios, cellphones, and stereo receivers.
Characterization of digital filters
 A digital filter is characterized by its transfer function, or equivalently, its
difference equation. Mathematical analysis of the transfer function can
describe how it will respond to specifications any input. As such,
designing a filter consists of developing appropriate to the problem (for
example, a second-order low pass filter with a specific cut-off frequency),
and then producing a transfer function which meets the specifications.
 The transfer function for a linear, time-invariant, digital filter can be
expressed as a transfer function in the Z-domain; if it is causal, then it has
the form:

 where the order of the filter is the greater of N or M.

 This is the form for a recursive filter with both the inputs (Numerator) and
outputs (Denominator), which typically leads to an IIR infinite impulse response
behaviour, but if the denominator is made equal to unity i.e. no feedback, then
this becomes an FIR or finite impulse response filter.
Difference equation
 In discrete-time systems, the digital filter is often implemented by converting
the transfer function to a linear constant-coefficient difference equation
(LCCD) via the Z-transform. The discrete frequency-domain transfer function
is written as the ratio of two polynomials. For example:

 This is expanded:

and divided by the highest order of z:

 The coefficients of the denominator, ak, are the 'feed-backward' coefficients
and the coefficients of the numerator are the 'feed-forward' coefficients, bk.
The resultant linear difference equation is:

or, for the example above:

rearranging terms

then by taking the inverse z-transform:

 and finally, by solving for y[n]:

 This equation shows how to compute the next output sample, y[n], in
terms of the past outputs, y[n − p], the present input, x[n], and the
past inputs, x[n − p]. Applying the filter to an input in this form is
equivalent to a Direct Form I or II realization, depending on the exact
order of evaluation.
 Implementation
Direct Form 1
 The most straightforward implementation is the Direct Form 1, which
has the following difference equation:

 Here the b0, b1 and b2 coefficients determine zeros, and a1, a2 determine the
position of the poles.
 Flow graph of biquad filter in Direct Form 1:
Digital signal processing (DSP) is concerned with
the representation of signals by a sequence of numbers
or symbols and the processing of these signals.
GOALS OF DSP:
1. to measure, filter and/or compress continuous real-
world analog signals
 convert the signal from an analog to a digital form using an
analog-to-digital converter
 the required output signal is another analog output signal,
which requires a digital-to-analog converter (DAC)
 2.other advantages:
error detection and correction in transmission data &
compression of data.

DSP logirithm standard computers using digital

signal processors or purpose built hardwares.
Some more powerful general purpose
1. microprocessors
2. field-programmable gate arrays (FPGAs)
3. digital signal controllers (mostly for industrial apps
such as motor control)
4.stream processors
DSP DOMAIN:
engineers usually study digital signals in one of the
following domains: time domain (one-dimensional
signals), spatial domain (multidimensional signals),
frequency domain and wavelet domains.
TIME AND SPACE DOMAIN:
The most common processing approach in the time or
space domain is enhancement of the input signal
through a method called filtering. Digital filtering
generally consists of some linear transformation of a
number of surrounding samples around the current
sample of the input or output signal
FREQUENCY DOMAIN:
Signals are converted from time or space domain to the
frequency domain usually through the Fourier transform.
The Fourier transform converts the signal information to
a magnitude and phase component of each frequency.
Z-DOMAIN ANALYSIS:
analog filters are usually analysed on the s-plane; digital
filters are analysed on the z-plane or z-domain in terms
of z-transforms
Most filters can be described in Z-domain (a complex
number superset of the frequency domain) by their
transfer functions. A filter may be analysed in the z-
domain by its characteristic collection of zeroes and
poles.
VARIOUS SUBFIELDS:
1. audio and speech signal processing
2. sonar and radar signal processing
3. sensor array processing
4. digital image processing
5. signal processing for communications
6. control of systems
7. seismic data processing