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Dsp Paper Part1

The document discusses properties of discrete-time systems, including linearity, stability, and causality, along with the convolution sum and Fourier transform properties. It compares analog and digital filters, highlighting their advantages and disadvantages, and outlines the differences between FIR and IIR filters. Additionally, it covers applications of digital signal processing in various fields and introduces the Z-transform and its properties.

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tanish gehlot
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0% found this document useful (0 votes)
4 views93 pages

Dsp Paper Part1

The document discusses properties of discrete-time systems, including linearity, stability, and causality, along with the convolution sum and Fourier transform properties. It compares analog and digital filters, highlighting their advantages and disadvantages, and outlines the differences between FIR and IIR filters. Additionally, it covers applications of digital signal processing in various fields and introduces the Z-transform and its properties.

Uploaded by

tanish gehlot
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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2023

Properties of discrete time system

Static, linear, casual, time variant , stable


https://youtu.be/JQl_2zLmQ1k?si=jN9WGIOG4JatH39n
Convolution Sum - Properties - Graphical Method (youtube.com)

Convolution using graphical method


Direct form 1 and direct form 2 realisation
Properties of discrte time fourier transform

Periodicity

Time shifting , reversal, frequency

Parseval’s theroem
Determine fourier transform of discrtet time sequence
Z transform and pole -zero plot

Linear constant coefficient fifferential equation


Question

A discrete-time LTI system H has input x[n] and output y[n] related by the
linear constant coefficient difference equation y[n] − 1 2 y[n − 1] = x[n] + 1 3
x[n − 1]. (a) Find the transfer function H(z). Note: you can find the functional
form of H(z), but in this part you do not yet have enough information to find
the ROC. In parts (c) and (e) you will be given more information so that you
can find the ROC. (b) Give a pole-zero plot for H(z). (c) Now assume that the
system frequency response H(e jω) exists. For this assumption, give the ROC of
H(z) and find the system impulse response h[n]. (d) Under the assumption of
part (c) – that H(e jω) exists – is the system causal? Is it stable? (e) Now
assume that the system H is unstable and is not causal. For these assumptions,
give the ROC of H(z) and find the impulse response h[n]
Analog and Digital filters advantages and disadvantages
Characteristics Analog Filter Digital Filter
Working signals These filters work with These filters work with digital
analog or actual signals samples of the signal
Representation These filters are represented These filters are represented by
by linear differential linear difference equations
equations
Components Implementing such filters Implementing such filters requires
requires resistors, inductors, adders, subtractors, and delays
and capacitors
Frequency Approximation problem is Special coefficients are designed in
response computed in order to order to meet the expected
achieve the desired frequency response
frequency response
Stable & Causal Transfer function G(s) Transfer function G(s) should be a
Response should be a rational rational function of z-transform z,
function of laplace variable whose coefficients are real
s, whose coefficients are real numbers.
numbers.
Stability & Poles of transfer function Poles of transfer function should lie
Causality in terms should lie on left-half of s- inside the unit circle of z-plane
of Poles plane
Environmental Because of components These types of filters are less
changes tolerance, these filters are sensitive towards environmental
more sensitive to changes, noise, and disturbances.
environmental changes
Flexibility They are less flexible in They are more flexible because
nature software and control programs can
be modified easily according to the
requirements.
Adaptability These types of filter are less These types of filter are more
adaptive; we have to adaptive because they are
redesign the filter if we want programmable; which means that
to make any changes. we can make any changes in it
without affecting the filter circuitry.
Additive noise These filters introduce These filters introduce digital noise
thermal noise due to because of quantization process
components
Cost Higher because of analog Less costly
components involvement
Coefficients Not programmable Programmable; that’s why easy to
make changes
Design Difficult to design and then Significantly easy to design and
simulate because of several simulate on software program
components
ADC, DAC, AND In such filters, there is no These filters need high performance
DSP Requirement need for ADC, DAC, and DSP ADC, DAC, and DSP tools

Advantages of Analog Filters


o It is quite easy to apply, as there is no need for a microprocessor.
o There is no need to write a program/algorithm.
o Easy RC filters need very few components.
Design butterworth filter
Ciercular convolution
Applications of a Digital signal processing system
We use digital signal processing in:

• Telecommunication
• For echo cancellation.
• Equalization – Think about tuning your radio for bass and
treble).
• Filtering – Removing unwanted signals using specially
designed filters like the Infinite Impulse Response Filter (IIR).
• Multiplexing and repeating signals.
• Instrumentation and Control
• In designing Phase Locked Logic (PLL).
• Noise reduction circuits.
• Compression of signals.
• Function generators.
• Digital Image Processing
• Compression of an image.
• Enhancement, reconstruction, and restoration of an image.
• Analysis or face detection (like Snapchat).
• Speech Processing
• Digital audio synthesis.
• Speech recognition and analysis.
• Medicine
• X-rays, ECGs, EEGs.
• Signal filtering
• Noise removal and shaping of signal spectrums.
• Military
• Sonar and navigation.
• Analysis after tracking in radars.
• Consumer electronics
• Music players
• Professional music turntables (like the ones DJs use).
Properties of roc of z transform
FIR vs IIR Filters

What is FIR?
FIR stands for Finite Impulse Response. It is a type of digital filter where the output depends only
on the input values and a finite number of previous input values.

Examples of FIR Filters

Examples of FIR filters include low-pass filters, high-pass filters, band-pass filters, and notch
filters.

Uses of FIR Filters

FIR filters are commonly used for audio and image processing, noise reduction, and signal
processing applications.
What is IIR Filters?
IIR stands for Infinite Impulse Response. It is a type of digital filter where the output depends not
only on the input values but also on a combination of past input and output values.

Examples of IIR Filters

Examples of IIR filters include Butterworth filters, Chebyshev filters, elliptic filters, and Bessel
filters.

Uses of IIR Filters

IIR filters are widely used in communication systems, control systems, biomedical signal
processing, and many other applications.

Differences Between FIR and IIR Filters


Difference FIR Filters IIR Filters
Area

Filter Type FIR filters have a finite impulse response. IIR filters have an infinite impulse
response.

Stability FIR filters are always stable. IIR filters can be unstable.

Phase FIR filters have a linear phase response. IIR filters can have non-linear phase
Response response.

Impulse FIR filters have a symmetric impulse IIR filters can have both symmetric and
Response response. asymmetric impulse responses.

Design FIR filters are relatively easier to design IIR filters are relatively more complex to
Complexity and implement. design and implement.

Filter Order FIR filters generally have a higher filter IIR filters generally have a lower filter
order. order.

Frequency FIR filters have a linear phase frequency IIR filters can have non-linear phase
Response response. frequency response.

Low- FIR filters have a better low-frequency IIR filters may have a poor low-frequency
Frequency roll-off. roll-off.
Roll-off

Memory FIR filters require more memory due to IIR filters require less memory due to
Requirement their longer impulse response. their infinite impulse response.
Real-Time FIR filters are more suitable for real-time IIR filters are generally used for offline
Processing processing. processing.

Conclusion:

In summary, FIR filters have a finite impulse response, are always stable, have a linear phase
response, and are relatively easier to design. However, they require more memory and generally
have a higher filter order. On the other hand, IIR filters have an infinite impulse response, can be
unstable, have a non-linear phase response, and are relatively more complex to design. They
require less memory and generally have a lower filter order. The choice between FIR and IIR
filters depends on the specific requirements and constraints of the application.

Fast fourier transform (FFT)


2019

A system that possesses two basic properties namely linearity and time invariant is
known as linear time-invariant system or LTI system.

There are two major reasons behind the use of the LTI systems −

• The mathematical analysis becomes easier.


• Many physical processes through not absolutely LTI systems can be approximated with the
properties of linearity and time-invariance.
BIBO stability
Bounded Signal
A signal whose magnitude is a finite value is called the bounded signal. A sine wave is an
example of bounded signal.
Time shifting and time reversal

Z-Transform
The Z-transform is a mathematical tool which is used to convert the difference equations
in discrete time domain into the algebraic equations in z-domain. Mathematically,
if x(n) is a discrete time function, then its Z-transform is defined as,
Time Reversal Property of Z-Transform
Statement – The time reversal property of Z-transform states that the reversal or
reflection of the sequence in time domain corresponds to the inversion in z-domain.

Or

From book thoda sa eplained kam hai book mein but last vali language likhna shi hai
Time Shifting Property of Z-Transform
From book
Determine impulse response
LTI system
https://youtu.be/A4tSHx-icLs?si=WHCtfsrE9-y5RRB-
CIRCULAR SHIFT AND DUALITY
DSP#25 Duality property of DFT in Digital Signal Processing || EC Academy - YouTube

4 POINT DFT SEQUENCE


RALIX-2 DECIMATION IN FREQUENCY
IIR BILLINEAR TRANSFORMATION
advantages of the Bilinear Transformation method for designing IIR
filters
• The mapping is purely one-to-one. While it is not necessarily the same with
impulse invariance method.
• There is no bother of the aliasing effect.
• A stable analog filter can be transformed into a stable digital filter
• There are no restrictions on the type of filters that can be transformed. While
the other two methods are limited to Low Pass Filters and an even more limited
class of Bandpass filters.
• There is one to one transformation from the ‘s’ plane to the ‘z’ plane.
• All characteristics of the amplitude response of the analog filter are preserved
when designing the digital filter. Consequently, the pass band ripple and the
minimum stop band attenuation are preserved.

disadvantages of the Bilinear Transformation method for


designing IIR filters
• Frequency Warping is the only disadvantage, as the mapping is
non-linear we have to perform pre-warping.

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