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Filter Intro

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0% found this document useful (0 votes)
11 views

Filter Intro

Uploaded by

somesh.tg2023
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PPTX, PDF, TXT or read online on Scribd
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INTRODUCTION TO FILTERS

1
Need for Filters
• Most of the signals we deal with in real life get corrupted
in some way or another by some unwanted signals.

• In signal processing and analysis, it is imperative to get


rid of these interferences, or at least reduce their effects.

• This is achieved through applying signal Filtering


techniques.

• Filters have two uses: Signal separation and


Signal restoration.
2
• Filter classification
– Based on length Impulse response
– Based on the magnitude spectrum
– Based on the phase spectrum
– Based on the type of input
• Analog filter specification

3
Signal separation
• Signal separation is needed when a signal has been
contaminated with interference, noise, or other signals.

• Imagine a device for measuring the electrical activity of a


human heart (ECG).

• The raw signal will likely be corrupted by the interference


caused by shaking the wire of the ECG lead.

• The measuring the electrical activity of a baby's heart while


still in the womb. The raw signal will likely be corrupted by
the breathing and heartbeat of the mother.

• A filter might be used to separate these signals so that they can


4
be individually analyzed.
Signal restoration
• Signal restoration is used when a signal has been
distorted in some way.

• An audio recording made with poor equipment may


be filtered to better represent the sound as it actually
occurred.

• Another example is the deburring of an image


acquired with an improperly focused lens, or a shaky
camera.

5
• Based
Filter Classification
on the transfer function (It depends on
– the length
Finite of its impulse response
impulse response (FIR) transfer
– function
Infinite
function impulse response (IIR) transfer
• For digital
selective transferresponses,
frequency functions one
withclassification
frequency-
is based
|H(ω)| oron
thethe
formshape of phase
of the the magnitude
function function
θ(ω)

• Based on the magnitude spectrum, one of four


types of ideal filters are usually defined
• Low pass
• High pass
• Band pass
• Band stop

6
Ideal Filter
•An ideal filter is a digital filter designed to pass signal
components of certain frequencies without distortion,
which therefore has a frequency response equal to 1 at
these frequencies, and has a frequency response equal to
0 at all other frequencies

•The range of frequencies where the frequency


response takes the value of one is called the passband

•The range of frequencies where the frequency


response takes the value of zero is called the
stopband

•The transition frequency from a passband to stopband


region is called the cutoff frequency 7
Ideal filter – low pass filter
• Low-pass filters are designed to pass low frequencies, from
zero to a certain cut off frequency and to block high
frequencies
Ideal magnitude frequency response

H e j  1

0
0  

8
Ideal filter – High pass filter
• High-pass filters are designed to pass high frequencies,
from a certain cut off frequency to, and to block low
frequencies
Ideal magnitude frequency response

H e j
1

0
0  

9
Ideal filter – Band pass filter
• Band-Pass Filters: Band-pass filters are designed to pass a
certain frequency range, which does not include zero, and
to block other frequencies

Ideal magnitude frequency response

H e j  1

0
1 2  

10
Ideal filter – Band stop filter
• Band-Stop Filters: Band-stop filters are designed to block
a certain frequency range, which does not include zero,
and to pass other frequencies
Ideal magnitude frequency response

H e j  1

0
1 2  

11
Multi Band filter
• This type of filters generalizes the previous four types of
filters in that it allows for different gains or attenuations in
different frequency bands

Possible ideal magnitude frequency response

H e j  1

0
1  2 3  4 5 6  

12
Based on phase response
The phase response is the relationship between the
phase of a sinusoidal input and the output signal passing
through the system, such as an amplifier or a filter
• Phase delay: a measure of the time delay of the phase
• Group delay: First derivative of phase delay

• Input: j t  ( )
x(t ) e T 

• Output: y (t ) e jt H ( j )
d ( )
Tg 
e jt  H ( j ) e j ( )  d

13
Based on phase response
– Zero phase
One way to avoid any phase distortion is to make sure the frequency response
of the filter does not delay any of the spectral components. Such a transfer
function is said to have a zero – phase characteristic.
– Linear phase
For a causal transfer function with a nonzero phase response, the phase
distortion can be avoided by ensuring that the transfer function has (preferably)
a unity magnitude and a linear-phase characteristic in the frequency band of
interest
– Generalized linear phase
As long as the system response does not change signs for different values of ω,
it can be written as a linear phase transfer function. Therefore, such systems are
said to have generalized linear phase
– Non-linear phase
• Phase response of filter with non linear phase or non constant group delay

14
Filters are further classified as Analog and Digital
filter

• Digital and analog filters both take out unwanted


noise or signal components, but filters work
differently in the analog and digital domains.

• Analog filters will remove everything above or below


a chosen cutoff frequency.

• Digital filters can be more precise in filtering, but the


signal must be digital.

15
• Placing a digital filter in an analog signal chain would
require the analog signal to be converted to a digital
signal before the digital filtering could be applied
and, with any conversion, there are trade-offs in
signal integrity.

• Digital filters work by oversampling and averaging,


and are programmable. But it is wise to apply an
analog filter prior to signal conversion so that all
unwanted frequencies above or below where the
desired signal is reasonably expected to operate are
removed first.

16
Specifications Analog Filter Digital Filter
embedded in a chip that
resistors, capacitors, inductors, operates on digital
Components and op amps signals, such as an MCU,
SoC, processor, or DSP.

Programmable
coefficients No Yes

Complexity Higher, usually seen in high Lower


performance based analog filters.
Accuracy Lower Higher
Higher, it depends on analog
Cost components used in the design. Lower

Additive noise It adds component based thermal It adds digital noise due
introduction noise in the bands to quantization process

Storing Difficult Easy


information

17
Analog Filter Specifications

The magnitude |Ha(jΩ)| of an analog low pass filter is usually


specified as indicated in the figure above

18
• In the passband defined by 0 ≤ Ω ≤ Ωp
1 − δp ≤ |Ha(jΩ)| ≤ 1 + δp, for |Ω| ≤ Ωp

• In the stopband defined by Ωs ≤ |Ω| ≤ ∞


|Ha(jΩ)| ≤ δs, for Ωs ≤ |Ω| ≤ ∞
implying that the magnitude approximate zero within an
error of δs.

• The frequencies Ωp and Ωs are called, respectively, the


passband edge frequency and the stopband edge
frequency.

• The parameter δp and δs is the tolerance of the magnitude


response in the pass band and stop band respectively. 19
• The desired magnitude response in the passband is 1
or (1+δp) and in the stop band is 0.

• Pass band ripple in dB defined by Ap= - 20 log(1-δp).

• Pass band gain at Ω= Ωp, Kp= - Ap =20 log(1- δp)

• Stop band attenuation defined by As = -20 log(δs)

• Stop band gain at Ω= Ωs, Ks=- As =20 log(δs)

20

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