Salient Features:
The magnitude response is nearly constant(equal to 1) at lower frequencies
There are no ripples in passband and stop band
The maximum gain occurs at Ω=0 and it is H(Ω)=1
The magnitude response is monotonically decreasing
As the order of the filter ‘N’ increases, the response of the filter is more close to the ideal response
Using Chebyshev filter design, there are two sub groups,
Type-I Chebyshev Filter
Type-II Chebyshev Filter
The major difference between butterworth and chebyshev filter is that the poles of butterworth filter lie on the circle while the poles of chebyshev filter lie on ellipse.
Salient Features:
The magnitude response is nearly constant(equal to 1) at lower frequencies
There are no ripples in passband and stop band
The maximum gain occurs at Ω=0 and it is H(Ω)=1
The magnitude response is monotonically decreasing
As the order of the filter ‘N’ increases, the response of the filter is more close to the ideal response
Using Chebyshev filter design, there are two sub groups,
Type-I Chebyshev Filter
Type-II Chebyshev Filter
The major difference between butterworth and chebyshev filter is that the poles of butterworth filter lie on the circle while the poles of chebyshev filter lie on ellipse.
The presentation covers sampling theorem, ideal sampling, flat top sampling, natural sampling, reconstruction of signals from samples, aliasing effect, zero order hold, upsampling, downsampling, and discrete time processing of continuous time signals.
The presentation covers sampling theorem, ideal sampling, flat top sampling, natural sampling, reconstruction of signals from samples, aliasing effect, zero order hold, upsampling, downsampling, and discrete time processing of continuous time signals.
Embedded systems increasingly employ digital, analog and RF signals all of which are tightly synchronized in time. Debugging these systems is challenging in that one needs to measure a number of different signals in one or more domains (time, digital, frequency) and with tight time synchronization. This session will discuss how a digital oscilloscope can be used to effectively debug these systems, and some of the instrumentation considerations that go along with this.
- Obtained the Fast Fourier Transform of signals.
- Designed and Validated Low Pass, High Pass, and Band Pass filters in compliance with the specifications.
- Produced and compared graphs of the results upon processing.
Design Technique of Bandpass FIR filter using Various Window FunctionIOSR Journals
Abstract: Filter is one of the most important part of communication system. Without digital filter we cannot think about proper communication because noise occurs in channel. For removing noise or cancellation of noise we use various type of digital filter. In this paper we propose design technique of bandpass FIR filter using various type of window function. Kaiser window is the best window function in FIR filter design. Using this window we can realize that FIR filter is simple and fast. Keywords: FIR filter, LTI, bandpass filter, MATLAB
Design Technique of Bandpass FIR filter using Various Window FunctionIOSR Journals
Abstract: Filter is one of the most important part of communication system. Without digital filter we cannot
think about proper communication because noise occurs in channel. For removing noise or cancellation of
noise we use various type of digital filter. In this paper we propose design technique of bandpass FIR filter
using various type of window function. Kaiser window is the best window function in FIR filter design. Using
this window we can realize that FIR filter is simple and fast.
Keywords: FIR filter, LTI, bandpass filter, MATLAB
A mind map connecting important DSP concepts on LTI System. The mind map is divided into three sections, which are Filtering, Analysis and Design. The mind map is also showing basic Matlab functions related to the processes available on the mind map. Concepts such as LTI system, IIR, FIR, time-domain, frequency-domain, z-domain, and Fourier transform are mapped on a single page diagram.
2. SECTION OUTLINE:
Introduction to Digital filter design
Introduction to FIR Filter
Design of FIR Filter using WINDOW Techniques
Introduction to IIR Filter
Design of IIR Filter using Approximation Method
MATLAB Code of Designed Filters
3. Introduction:
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.
In digital signal processing, there are two important types
of systems:
Digital filters: perform signal filtering in the time
domain
Spectrum analyzers: provide signal representation in the
frequency domain
4. Digital Filter:
xn yn
Digital Filter
Sampling
frequency
fS
A
D
C
D
A
C
x(t) y(t)
Analog
anti-
aliasing
filter
Analog
smoothing
filter
5. Preliminaries:
The design of a digital filter is carried out in three steps:
Specifications: they are determined by the applications
Approximations: once the specification are defined, we use various
concepts and mathematics that we studied so far to come up with a
filter description that approximates the given set of specifications.
(in detail)
Implementation: The product of the above step is a filter
description in the form of either a difference equation, or a system
function H(z), or an impulse response h(n). From this description
we implement the filter in hardware or through software on a
computer.
6. Classification:
Digital filters are classified into one of two basic forms,
according to how they respond to a unit impulse:
Finite impulse response
Infinite impulse response
7. Finite Impulse Response:
In signal processing, a finite impulse response
(FIR) filter is a filter whose impulse response is
of finite duration, because it settles to zero in finite
time.
FIR digital filters use only current and past input
samples, and none of the filter's previous output
samples, to obtain a current output sample value
8. The transfer function is given by
The length of Impulse Response is N
All poles are at Z=0. .
Zeros can be placed anywhere on the z-plane
1
0
).()(
N
n
n
znhzH
9. Filter Design by Windowing
Simplest way of designing FIR filters
Start with ideal frequency response
Choose ideal frequency response as desired response
Most ideal impulse responses are of infinite length
n
nj
d
j
d enheH
deeH
2
1
nh njj
dd
else0
Mn0nh
nh d
11. Kaiser window
Kaiser window
β Transition
width (Hz)
Min. stop attn
dB
2.12 1.5/N 30
4.54 2.9/N 50
6.76 4.3/N 70
8.96 5.7/N 90
12. Rectangular Window
else0
Mn01
nw
Narrowest main lob
– 4/(M+1)
– Sharpest transitions at
discontinuities in frequency
Large side lobs
– Large oscillation around
discontinuities
– -13 dB
Simplest window possible
14. Kaiser Window
Parameterized equation forming a set
of windows
Parameter to change main-lob width
and side-lob area trade-off
I0(.) represents zeroth-order
modified Bessel function of 1st kind
else0
Mn0
I
2/M
2/Mn
1I
nw
0
2
0
15. MATLAB CODE:
%Design of LPF&HPF using rectangular,hamming and kaiser windows
clc;clear all;close all;
rp=input('enter attenuation in pass band');
rs=input('enter attenuation in stop band');
fp=input('enter pass band frequency');
fs=input('enter stop band frequency');
Fs=input('enter sampling frequency');
wp=2*pi*fp/Fs;
ws=2*pi*fs/Fs;
%formula for FIR filter
num=-20*log10(sqrt(rp*rs))-13;
den=14.6*(fs-fp)/Fs;
n=ceil(num/den);
disp('order of filter is n');
disp(n);
disp('press any key to continue');
pause;
n1=n+1;
%For even order
if(rem(n,2)~=0)
n1=n;
end
%LPF
16. %LPF
s1=input('enter the value for window 0-rectangularLPF 1-kaiserLPF 2-hammingLPF 3-
rectangularHPF 4-kaiserHPF 5-hammingHPF');
switch(s1);
case 0
y=rectwin(n1);
[b,a]=fir1(n,wp,'low',y);
freqz(b,a,512);
case 1
y=kaiser(n1);
[b,a]=fir1(n,wp,'low',y);
freqz(b,a,512);
case 2
y=hamming(n1);
[b,a]=fir1(n,wp,'low',y);
freqz(b,a,512);
case 3
y=rectwin(n1);
[b,a]=fir1(n,wp,'high',y);
freqz(b,a,512);
case 4
y=kaiser(n1);
[b,a]=fir1(n,wp,'high',y);
freqz(b,a,512);
case 5
y=hamming(n1);
[b,a]=fir1(n,wp,'high',y);
freqz(b,a,512);
end
17. Pros & Cons:
FIR filters have the following
advantages:
Exactly linear phase is possible
Always stable, even when quantized
Design methods are generally linear
Efficient hardware realizations
Startup transients have finite
duration
FIR filters have the following
disadvantages:
• Higher filter order than IIR
filters
• Corresponding greater
delays
18. Infinite Impulse Response Filter:
IIR systems have an impulse response function that is
non-zero over an infinite length of time. This is in
contrast to finite impulse response (FIR) filters, which
have fixed-duration impulse responses
IIR filters may be implemented as
either analog or digital filters
19. Cont..
While designing a digital IIR filter , an analog filter
(e.g. Chebyshev filter, Butterworth filter) is first
designed and then is converted to a digital filter by
applying discretization techniques such as Bilinear
transform or Impulse invariance.
21. Chebyshev Filter:
Chebyshev filters are analog or digital filters having a
steeper roll-off and more passband ripple (type I) or
stopband ripple (type II)
Chebyshev filters have the property that they
minimize the error between the idealized and the
actual filter characteristic over the range of the filter,
but with ripples in the passband
23. Butterworth filter
The Butterworth filter is a type of signal processing
filter designed to have as flat a frequency response as
possible in the pass band. It is also referred to as a
maximally flat magnitude filter
24. MATLAB Prototype Filter Design Commands
[B,A] = BUTTER(N,Wn)
[B,A] = CHEBY1(N,R,Wn)
[B,A] = CHEBY2(N,R,Wn)
[B,A] = ELLIP(N,Rp,Rs,Wn)
– N = filter order
– R = pass band ripple (cheby1) or stop-band ripple (cheby2) in
dB. (Rp and Rs respectively for the elliptic filter)
– Wn = cut-off frequency (radians/sec for analog filters or
normalized digital frequencies for digital filters)
– [B,A] = filter coefficients, s-domain (analog filter) or z-domain
(digital filter)
25. Design Example
Filter Specifications:
Butterworth response
Pass-band edges = 400 Hz and 600 Hz
Stop-band edges = 300 Hz and 700 Hz
Pass-band ripple = 1 dB
Stop-band attenuation = -20 dB
Sampling Frequency = 2000 Hz
26. Design Example Results
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
0
0.2
0.4
0.6
0.8
1
1.2
1.4
Frequency (kHz)
Magnitude
Magnitude Response
Band Edges
(-1dB and -20 dB)
27. Design Example
Chebyshev II High-Pass Filter
Filter specifications:
Chebyshev II response (stop-band ripple)
Pass-band edge = 1000 Hz
Stop-band edge = 900 Hz
Pass-band ripple = 1 dB
Stop-band attenuation = -40 dB
Sampling frequency = 8 kHz
28. MATLAB Code for Design Example
>> fs=8000;
>> Wp=[2*1000/fs]; % Pass-band edge normalized digital frequency
>> Ws=[2*900/fs]; % Stop-band edge normalized digital frequency
>> [N,Wn]=cheb2ord(Wp,Ws,1,40); % The “order” command
>> [B,A]=cheby2(N,40,Wn,'high');
% cheby2 is the “filter” command. In this command
% the syntax requires the stop-band attenuation
% as the second parameter
>> fvtool(B,A)
31. Summary of IIR Filter:
IIR filters can be design by pole-zero location
– Digital oscillators: poles on the unit circle
– Notch filters: zeros on the unit circle with nearby poles to
control notch width
Classic analog filters can be designed using the
bilinear transformation
IIR filters have the advantage of smaller filter order for
a given frequency response.
IIR filters have the disadvantages of possible instability
due to coefficient quantization effects and non-linear
phase response.
32. References:
“Design of IIR Filter” by K.S Chandra, M.Tech, IIT-Bombay, Jan-2006
“Digital Filter Design” by Prof. A.G. Constantinides, University of
Auckland, 2006
“FIR Filter Design”, Gao Xinbo,School of E.E., Xidian Univ.
xbgao@ieee.org
“FIR Filter by Windowing”- The lab Book Pages.com
“Digital Signal Processing”, Prof.Ramesh Babu, Pondicherry Govt.
College, TataMcgraw-Hill publication.
Wikipedia.org