This document discusses Infinite Impulse Response (IIR) filters. IIR filters are more computationally efficient than FIR filters as they require fewer coefficients due to using feedback. The document covers IIR filter concepts, properties, design procedures including specification, coefficient calculation, structure selection, and implementation. It also provides examples of coefficient calculation methods like pole-zero placement and the bilinear transform method for converting analog filters to digital IIR filters.
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
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
A Novel Methodology for Designing Linear Phase IIR FiltersIDES Editor
This paper presents a novel technique for
designing an Infinite Impulse Response (IIR) Filter with
Linear Phase Response. The design of IIR filter is always a
challenging task due to the reason that a Linear Phase
Response is not realizable in this kind. The conventional
techniques involve large number of samples and higher
order filter for better approximation resulting in complex
hardware for implementing the same. In addition, an
extensive computational resource for obtaining the inverse
of huge matrices is required. However, we propose a
technique, which uses the frequency domain sampling along
with the linear programming concept to achieve a filter
design, which gives a best approximation for the linear
phase response. The proposed method can give the closest
response with less number of samples (only 10) and is
computationally simple. We have presented the filter design
along with its formulation and solving methodology.
Numerical results are used to substantiate the efficiency of
the proposed method.
ECG COMPRESSION USING
FFT
The electrocardiogram (ECG) is a diagnostic tool that is routinely used to assess the electrical and muscular functions of the heart. Sometimes it is required to send the ECG signals from one place to another place. The ECG signals are compressed at first to reduce the amplitude and frequency and then transferred. ECG signals are compressed by using many techniques. One of the most important technique is FFT.
FFT (Fast Fourier Transform) is a technique used to convert analog signal to digital signal.
In FFT, The total process takes five steps:-
1) Input signal
2) Compression (counter A)
3) Compression (counter B)
4) Recovery of the original signal by using IFFT
5) Error checking
Now the detailed explanation of the above steps is given below
At first the input signal (ECG signal) is taken.
There are two stages for compression. In first stage of compression there is a counter A. It identifies the non-zero values of the signal before compression. After compression if the length of the compressed signal is less than the length of the actual signal, then zero padding is done to make equal the lengths of compressed and actual signal.
Now the signal is passed through the counter B. It identifies the non-zero values after the compression of the signal. Now after compression if the length of the compressed signal is greater than the length of the actual signal, then TRUNCATION of the signal is done.
Now by applying IFFT (Inverse Fast Fourier Transform) the original ECG signal is recovered.
The Error is checked at the last stage.
Compression ratio is given by
CR=(B-A)/B *100
CR-Compression ratio
A-compression in counter A
B-compression in counter B
Compression ratio is a major factor to determine how much compression the signal undergoes.
The compressed signal contains only positive values.
Thus ECG signal is compressed by using FFT technique.
Applications:-
• It finds application in hospitals, when a patient’s report is to be send to another doctor in prenomial place.
Time v Frequency Domain Analysis For Large Automotive SystemsAltair
It has been recognised since the 1960’s that the frequency domain method for structural analysis offers superior qualitative information about structural response; But computational and technological issues have held back the implementation for fatigue calculation until now. Recent technological developments have now enabled the practical implementation of the frequency domain approach and this paper will demonstrate this, with particular reference to the technology limitations that have been overcome, the resultant performance advantages, and accuracy. These techniques are of relevance to all the large automotive OEM’s as well as aerospace T1 suppliers and example case studies from these companies will be included.
In this presentation we can get to know how we can construct a bode plot with suitable examples Of different -different orders.
Along with that a simulation model on MATLAB with graph.
And in this we have explained about the transfer function, poles & zeroes.And the basic concept of stability.
The discrete Fourier transform has many applications in science and engineering. For example, it is often used in digital signal processing applications such as voice recognition and image processing.
2. Introduction
Infinite Impulse Response (IIR) filters are
the first choice when:
Speed is paramount.
Phase non-linearity is acceptable.
IIR filters are computationally more
efficient than FIR filters as they require
fewer coefficients due to the fact that they
use feedback or poles.
However feedback can result in the filter
becoming unstable if the coefficients
deviate from their true values.
3. Concept of Digital IIR filter
h(k), k=0,1,2,…,∞
x(n) (impulse response- y(n)
Input sequence infinite length) output sequence
y(n) h(k ) x(n k )
k 0
4. Properties of an IIR Filter
The general equation of an IIR filter can
be expressed as follows:
b0 b1 z 1 bN z N
H z
1 a1 z 1 aM z M
N
k
bk z
k 0
M
k
1 ak z
k 1
ak and bk are the filter coefficients.
5. Properties of an IIR Filter
The transfer function can be factorised to
give:
z z1 z z 2 z z N Y z
H z k
z p1 z p2 z p N X z
Where: z1, z2, …, zN are the zeros,
p1, p2, …, pN are the poles.
6. Properties of an IIR Filter .. continued
Time domain theoretical equation of IIR
y(n) h(k ) x(n k )
k 0
For the implementation of the above
equation we need the difference equation:
N M
y(n) b x(n k ) b y (n k )
k 0 k k 1k
b(0) x(n) b(1) x(n 1) b(2) x(n 2)
a(1) y(n 1) a(2) y(n 2)
8. IIR Direct Form I and Direct Form II
Structures
x[n] y[n]
+
Z-1 b0 Z-1
x[n-1] y[n-1] IIR
Z-1 b1
-a1 Z-1 Direct
x[n-2] Form I
y[n-2]
b2
-a2
x[n] + + y[n]
b0
Z-1 IIR Direct
-a1 b1 Form II
Z-1
(Canonic)
-a2 b2
y(n) b(0)x(n) b(1)x(n 1) b(2)x(n 2) a(1) y(n 1) a(2) y(n 2)
9.
10. Design Procedure
To fully design and implement a filter five
steps are required:
(1) Filter specification.
(2) Coefficient calculation.
(3) Structure selection.
(4) Simulation (optional).
(5) Implementation.
11.
12.
13. Filter Specification - Step 1
|H(f )| pass-band stop-band
1
fc : cut-of f f requency fs/2 f (norm)
(a)
|H(f )| pass-band transition band stop-band |H(f )|
(dB) (linear)
p 1 p
0 1
1 p
pass-band
-3
ripple
stop-band
ripple
s s
fs/2 f (norm)
fsb : stop-band f requency
fc : cut-of f f requency
fpb : pass-band f requency
(b)
14. Coefficient Calculation - Step 2
There are two different methods available
for calculating the coefficients:
Direct placement of poles and zeros.
Using analogue filter design.
Impulse invariant
Bilinear z-transform etc
15. Pole-zero Placement Method
All that is required for this method is the
knowledge that:
Placing a zero near or on the unit circle in
the z-plane will minimise the transfer
function at this point.
Placing a pole near or on the unit circle in
the z-plane will maximise the transfer
function at this point.
To obtain real coefficients the poles and
zeros must either be real or occur in
complex conjugate pairs.
16. Analogue to Digital Filter Conversion
This is one of the simplest method.
There is a rich collection of prototype
analogue filters with well-established
analysis methods.
The method involves designing an
analogue filter and then transforming it to
a digital filter.
The two principle methods are:
Bilinear transform method
Impulse invariant method.
17. Realisation Structures - Step 3
Direct Form I:
N
k
bk z
Y z k 0 b0 b1 z 1 bN z N
H z M
X z k 1 a1 z 1 aM z M
1 ak z
k 1
Difference equation:
N M
yn bk x n k ak y n k
k 0 k 1
This leads to the following structure…
20. Bilinear Transform Method
Practical example of the bilinear
transform method:
The design of a digital filter to approximate a
second order low-pass analogue filter is
required.
The transfer function that describes the
analogue filter is (This is an analog
BUTTERWORTH filter):
1
H s
s2 2s 1
The digital filter is required to have:
Cut-off frequency =100 Hz
Sampling frequency of 1 kHz.
21. Example: Design of a LP filter design using
Bilinear Transformation
Design a digital equivalent of a 2nd order
Butterworth LP filter with a
cut-off freq fc=100 Hz
Sampling frequency fs=1000 Hz.
The normalized cut off frequency of the digital filter
ω=2πfc/fs=2πfc 100/1000=0.628
Now equivalent analog filter cut-off freq
Ω=ktan(ω/2)= 1.tan(0.628/2)=0.325 rads/sec
22. Design of a LP filter design using Bilinear
Transformation ..continued
H(s) for a Butterworth filter
1
H ( s)
s2 2s 1
Now the transfer function of this Butterworth filter
becomes (putting s=s/Ω)
1
H ( s)
s 2 s
2 1
0.325 0.325
23. Design of a LP filter design using Bilinear Transformation
..continued
Now convert the analog filter H(s) into equivalent
digital filter H(z) by applying the bilinear z-transform-
1
z 1 1 z
s 1
z 1 1 z
1
H ( z)
1 1 z 1 2 2 1 z 1
1
0.3252 1 z 1 0.325 1 z 1
0.067 0.135z 1 0.067 z 2
1 1.1429 z 1 0.4127 z 2
24. Design of a LP filter design using Bilinear Transformation
..continued
From the previous we get the filter coefficients-
{bk}Coefficients {ak}coefficients
b0=0.067 a1= -1.1429
b1=0.0135 a2=0.4127
b2=0.067
The time domain difference equation is obtained
y(n) b(0) x(n) b(1) x(n 1) b(2) x(n 2)
a(1) y(n 1) a(2) y(n 2)
0.67 x(n) 0.135x(n 1) 0.67 x(n)
1.1429 y(n 1) 0.4127 y(n 2)
25. Direct realization of the design
x(n) y(n)
b0 + +
b0=0.067
z-1 z-1
b + + a
1 1
b1=0.135 a1=1.1429
z-1 z-1
b a
2 2
b2=0.067 a2= -0.4127
26. Matlab Code for the BZT Method
clear all
fc=100;fs=1000;
wp=2*pi*fc*(1/fs);
OmegaP=2*fs*tan(wp/2);
Omega=tan(wp/2);
Hs_den1=[(1/Omega)^2,((2^0.5)/Omega),1];
Hs_den=[(1/OmegaP)^2,((2^0.5)/OmegaP),1];
Hs_num=1;
Hs_denN=Hs_den/Hs_den(1);
Hs_numN=Hs_num/Hs_den(1);
[b,a]=bilinear(Hs_numN,Hs_denN,fs);
freqz(b,a,fs)