Digital Signal Processing

IIR digital filter structures

Filter design

1
Basic IIR Digital Filter Structure
 An N-th order IIR digital transfer function is characterised by 2N+1 unique
coefficients, and in general, requires 2N+1 multipliers and 2N two input adders for
implementation

 Direct form IIR filter: Filter structures in which the multiplier coefficients are
precisely the coefficients of the transfer function

 Consider, a 3rd order IIR filter with transfer function

 We can implement H(z) as a cascade

 where

2
Basic IIR Digital Filter Structure
 The filter section H1(z) can be seen to be an FIR filter and can be realised:

 Time domain representation of H2(z) is given by

3
Direct Form IIR Digital Filter Structure

 Cascade of the two structures leads to H(z) and is known as direct form I
structure

 It is noncanonic as it uses 6 delays to realise 3rd order transfer function

 Its transpose (direct form It)

4
Direct Form IIR Digital Filter Structure

 Various other noncanonic direct

form structures can be derived by
simple block diagram
manipulations

 Observe in the direct form

structure (on the right), signal
variables in nodes 1 and 1’ are the
same, so the two top delays can
be shared

 Likewise, the middle and bottom

two delays can be shared

 We get direct form II and its

transpose (direct form IIt) is
also shown

 Now, it is canonic

5
Cascade form IIR Digital Filter Structure
 By expressing the numerator and the denominator polynomials of the
transfer function as a product of polynomials of lower degree, a digital
filter can be realised as a cascade of low-order filter sections

 Consider, for example, P( z ) P1 ( z ) P2 ( z ) P3 ( z )

H ( z)  
H(z)=P(z)/D(z) expressed as D( z ) D1 ( z ) D2 ( z ) D3 ( z )

 Examples of cascade realisations

obtained by different pole-zero
pairings

 Examples of cascade realisations

obtained by different ordering of
sections

6
Cascade form IIR Digital Filter Structure
 There are altogether a total of 36 different cascade realisations of H(z)
based on pole-zero pairings and orderings

 Due to finite wordlength effects, each such realisation behaves

differently

 Usually, the polynomials are factored into a product of 1 st order and 2nd
order polynomials:

 For a first order,

7
Cascade form IIR Digital Filter Structure

 Consider an example:

How?

 The direct form II and cascade

realisations are

 0.44 z 1  1  0.8227 z 1  0.0455 z 2 

H ( z)   1   
1  0.4 z   1  0.8 z 1  0.5 z 2 

Hint: try to remember a general form for direct form II and then reuse it!

8
Parallel Form IIR Digital Filter Structures
 A partial-fraction expansion of the transfer function in z -1 leads to the
parallel form I structure

 Assuming simple poles, the transfer function H(z) can be expressed as

 In the above for a real pole,

 A direct partial fraction expansion of the transfer function in z leads to

the parallel form II structure

 In the above for a real pole,

 We’ll see some examples……….

9
Parallel Form IIR Digital Filter Structures
 A partial fraction expansion of

 in z-1 yields

 0.25  0.25 0.6

H ( z)  1
 1
  0.1
1  (0.4  j 0.5831) z 1  (0.4  j 0.5831) z 1  0.4 z 1

a b a (1  B )  b(1  A)
Use  
1 A 1 B (1  A)(1  B )
Parallel form I

10
Parallel Form IIR Digital Filter Structures
 A partial fraction expansion of

 in z yields

0.24 z 1 (0.1  j 0.1458) z 1 (0.1  j 0.1458) z 1

H ( z)   
1  0.4 z 1 1  (0.4  j 0.5831) z 1 1  (0.4  j 0.5831) z 1

a b a (1  B )  b(1  A)
Use  
1 A 1 B (1  A)(1  B)

Parallel form II

11
FIR filter design
 We have briefly explored the design of FIR filters in Lecture 5
 We’ll look at it again here using MATLAB
 We’ll use the window method to design FIR filters

12
 Ideal filter and impulse response
 An ideal LPF Its impulse
Gain
LPF
1

Inverse DFT

freq
wc

Similar to 

 What is the problem – we can’t have infinite coefficient length

 So, we need to truncate

13
Ideal filter and impulse response
 Let us use a rectangular window

DFT

 We can see that the shape of the LPF has changed

 This is know as Gibbs phenomenon – oscillatory behaviour in the magnitude

responses caused by truncating the ideal impulse response function (i.e. the
rectangular window has an abrupt transition to zero)

 Gibbs phenomenon can be reduced by

 Using a window that tapers smoothly at each end
 Providing a smooth transition from passband to stopband in the magnitude
specifications

14
Filter length

 The filter length (i.e. order) affects the

magnitude response N=13

 Length increase, number of ripples in both

passband and stopband increases but with
N=25
a corresponding decrease in the ripple
widths

 That is as N increase, it gets closer to the Infinite N

ideal LPF

 But height of largest ripples remain the

same independent of length

 Similar oscillatory behaviour could be

observed in the magnitude responses of
the truncated versions of other types of
ideal filters

15
Common window functions
 Some common window functions

 Low pass filter frequency response


 N=51 and wc 
2

Magnitude response of window

functions with M=25

16
FIR filter design
 Step 1 - Specify the ideal or required frequency response, H D(w)

 Step 2 - Obtain impulse response, hD(n) of the desired filter by evaluating the
inverse Fourier transform

 Important note: we’ll look at hD(n) for standard filter designs in the next slide

 Step 3 - Select a window function and then the number of filter coefficients

 Step 4 - Obtain values of w(n) for the chosen window function and the values of
the actual FIR coefficients, h(n), by multiplying hD(n) with w(n)

h(n)=hD(n) w(n)

17
FIR filter design (Steps 1 and 2)
We can make
use of available
hD(n)

18
FIR filter design (step 3) – window?
 First problem in step 3 - which window function?

 The ratio of the main lobe/side lobe can be used

 Example, the frequency response of Hamming and Blackman windows

 Main lobe centred at ω=0

 Other ripples are called side lobes

19
FIR filter design (step 3) – window?
 To ensure a fast transition from passband to stopband, window should
have a very small main lobe width

 To reduce the passband and stopband ripple , the area under the
sidelobes should be small  so to increase the stopband attenuation, we
need to decrease the sidelobe amplitude

 Most of the time, these two requirements are contradictory

 Example: LPF with length=51 and cut-off at /2

20
FIR filter design (step 3) – length?
 Second problem in step 3 – number of filter coefficients?

 To obtain good (i.e. sharper) transition band  reduce main lobe width 
increase the filter length

 But increased filter length means increased computational complexity

 Eg: Frequency response of Hamming window with length 101 and 201
are shown below
50 50
Magnitude (dB)

Magnitude (dB)
0 0

-50 -50

-100 -100
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample) Normalized Frequency ( rad/sample)

200 200
Phase (degrees)

Phase (degrees)

0 0

-200 -200

-400 -400
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample) Normalized Frequency ( rad/sample)

21
21
 FIR filter design (step 3) – length?
 Using the same window as in the previous slide

 LPF with Hamming window, with normalised cut-off=0.3

50 50
Magnitude (dB)

Magnitude (dB)
0

-50 -50

-100 -100

-150 -150
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample) Normalized Frequency ( rad/sample)
0 0
Phase (degrees)

-1000
Phase (degrees) -2000
-2000

-3000 -4000

-4000
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -6000
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)
Normalized Frequency ( rad/sample)

22
FIR filter design (step 3) – order selection
 To select the ‘suitable’ order i.e. the length
 The lowest order that can meet the requirements

 There are several methods:

 Kaiser, Bellanger, Hermann

 Kaiser’s formula:

N
 
 20 log10  p s  13
N
 
 20 log10  p s  13
14.6(s   p ) / 2 14.6( f s  f p )

 Actual location of transition band is immaterial

 Bellanger’s formula:

2 log10 10 p s 
N  1
3( f s  f p )

 Hermann’s formula is more complex so we will not consider here (but generally it
gives slightly more accurate value for the order)

23
Example – Kaiser’s formula

N
 
 20 log10 0.0115(0.0178)  13
14.6(2000 / 12000  1800 / 12000)

24
Another example
 A requirement exists for an FIR digital filter to meet the following specifications:
 Passband: 150-250 Hz
 Transition width: 50 Hz
 Passband ripple: 0.1 dB
 Stopband attenuation: 60 dB
 Sampling frequency: 1 kHz
 Obtain the filter coefficients
 Solution - The peak ripple values are p=0.0115, s=0.001

 
 20 log10 0.0115(0.001)  13
N

 20 log10  p s  13
 Using Kaiser’s formula: N 
14.6(50 / 1000) 14.6( f s  f p )
 which give N=49.85. So, we could take N=50 (type 1, odd length)

 Note that the passband information is not used to obtain N, so it will be the same
even if we shift the passband or even if we use it for highpass, stopband etc

 If the transition bands are not same then we use the smaller transition band
  bigger transition band requires smaller N, so the smaller transition band
requires a higher order and is more important for the order selection

25
A common mistake
 Example: Obtain the FIR coefficients for a highpass filter with
Fc=600 Hz with FT=2000 Hz 0.4

0.3

 Assume we start with N=100

0.2

0.1

 Use the known impulse response for HPF 0

-0.1

-0.2

-0.3

-0.4
0 20 40 60 80 100 120

Magnitude (dB)
0

-5

-10

 Problem -15
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

 The phase is completely non-linear

Normalized Frequency ( rad/sample)

80

 The magnitude response is not good

Phase (degrees)
60

40

20

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

26
A common mistake – solution 1
0.4
 To obtain linear phase response 0.3

 Coefficients must be symmetry/antisymmetry 0.2

0.1

 So say, change N=201 0

-0.1

-0.2

-0.3

-0.4
0 50 100 150 200 250

50

Magnitude (dB)
0

-50

-100

-150
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

 Linear phase response achieved 5000

But there are still ripples (Gibbs phenomenon)

Phase (degrees)
 0

-5000

-10000
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

27
A common mistake – solution 2

 Let us use a window, say Hamming

1 0.4

0.9 0.3

0.8
0.2
0.7

0.1
0.6

0.5 0

0.4
-0.1
0.3
-0.2
0.2

0.1 -0.3

 Less ripple in the magnitude 0

0 50 100 150 200 250 -0.4
0 50 100 150 200 250

response (in the passband) 50

Magnitude (dB)
0

-50

 Why is the cut-off frequency in the -100

plot at 0.6  rad/sample? -150

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)
 We used 0.3 to obtain the impulse 5000

response function, right? Phase (degrees)

0

-5000

-10000
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

28
Taking the example further
 Let us actually use the filter after designing it!

20 20

0
0
Power Spectrum Magnitude (dB)

Power Spectrum Magnitude (dB)

-20
-20

-40
-40

-60

-60
-80

-80
-100

-100
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 -120
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Frequency
Frequency

 It could be improved – using higher N and improved window such as

Kaiser window

29
 Using MATLAB fir1 function
 Using MATLAB, filter coefficients can easily obtained using fir1 function!
 Hamming window is used by default
 For the same input x in the previous slide, let us design a bandpass filter
to extract only the component at 650 Hz 0.08

0.06

0.04

0.02

-0.02

-0.04

-0.06

-0.08

20 20 -0.1
0 50 100 150 200 250

0
0
Power Spectrum Magnitude (dB)

Power Spectrum Magnitude (dB)

-20
-20

-40
-40
-60

-60
-80

-80
-100

-100 -120
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Frequency Frequency

30
Improved FIR filter design
 The windows such as Hanning, Hamming etc are known as fixed window designs as
the  ripple values are fixed)

 FIR filter design can also be improved using adjustable window functions such as
Dolph-Chebyshev and Kaiser that provide control over  by means of an additional
parameter characterising the window

 The basis of FIR filter design is to obtain the set of minimal number of coefficients
for the required response

 So, computer based optimisation methods have been invented

 Parks-McClellan algorithm can be used to obtain the FIR coefficients (in MATLAB,
we can design this filter using ‘remez’ function)
 Normally, improved design over the standard window based method
 Known as computer based FIR filter design

 The only other FIR design method is frequency sampling method

 But knowledge of the window based method is sufficient for this course

 Note: Kaiser order and Kaiser window are two different matters, though we normally use them
together

31
IIR filter design
 The most common approach to IIR filter design:
 Convert the digital filter specifications into an analogue prototype
low-pass filter specifications
 Determine the analogue low-pass filter transfer function, Ha(s)
 Transform Ha(s) in the desired digital transfer function G(z)

 The most widely used transformation is the bilinear transformation that

maps the imaginary axis in the s-plane (j) onto the unit circle of the z-
plane

 So, relation between G(z) and Ha(s)

 However, this design requires some knowledge of analogue low-pass

filters and also s-plane and therefore, we will skip this method and
straightaway utilise IIR filter design using MATLAB

32
IIR filter design using MATLAB
 We have done IIR filter design using MATLAB in Lecture 5

 There are several classical IIR filters

 Butterworth, Chebyshev Types I and II, Elliptic
 MATLAB functions
 butter – Butterworth filter with flat passband (no ripples)
 ellip – Elliptic filter with ripples (but normally requiring lower order than
Butterworth for same transition band)
 cheby1 - Chebyshev filter controlling peak-to-peak ripple in the passband
 cheby2 - Chebyshev filter controlling the amount of stopband ripple

 First step is to estimate the order of the filter given the specifications – using
functions: buttord, cheb1ord, cheb2ord, ellipord etc

 Next, obtain the coefficients, B and A using functions: buttord, besself, cheby1,
cheby2, ellip, etc

 Finally, do the filtering using filtfilt function

 The frequency response of the filter can be obtained using function freqz(B,A)

33
IIR filter design using MATLAB -example
 Design an elliptic IIR low-pass filter with the specifications: F p=0.8 kHz,
Fs=1 kHz, FT=4 kHz, p=0.5 dB, s=40 dB

 MATLAB code:
 [N,Wn]=ellipord(0.4,0.5,0.5,40)
 [B,A]=ellip(N,0.5,40,Wn)

0
Magnitude (dB)

-50

-100

-150
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

0
Phase (degrees)

-100

-200

-300

-400
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

34
IIR filter design using MATLAB -example
 To filter a signal, use filtfilt function

20 20

0 0

Power Spectrum Magnitude (dB)

Power Spectrum Magnitude (dB)

-20 -20

-40 -40

-60 -60

-80 -80

-100 -100

-120 -120
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Frequency Frequency

35