CPEN 304 L04 - Digital - Filters

CPEN 304 Digital Signal Processing
Lecture 4
Digital Filters and properties
October 12, 2021
Dr. Godfrey A. Mills
Email: gmills@ug.edu.gh
Phone: 020-549-6944
Course TA
1 CPEN 304 Lecture 4 Baron Afutu
OUTLINE OF DSP COURSE
Course outline
Digital signal processing
Discrete systems
Z-transforms
Digital filters and properties
IIR digital filter design I (specifications)
IIR digital filter design II (design methods)
FIR digital filter design I
FIR digital filter design II
IIR and FIR digital filter implementation
Number representation
Filter design application
2 CPEN 304 Lecture 4
OUTLINE #4
Digital filtering and characteristics
Time and frequency domain filtering
Digital filter types and characteristics
FIR digital filters
IIR digital filters
Simple FIR digital filters
First order Lowpass and Highpass filters
Higher order Lowpass and Highpass filters
Simple IIR digital filters
First order Lowpass and Highpass filters
Higher order Lowpass and Highpass filters
Second order Bandpass and Bandstop filters
What is filtering
Filtering refers to process of extracting important features
or components of some data while eliminating or suppressing
the contributions from noise or other unwanted features;
We use term filter to describe any system (device or circuits
or program) that is designed to examine each input
request to ensure meeting certain qualifying criteria
or perform some form of discrimination and process
the input accordingly
We use this kind of filter to either suppress noise in signals or
to separate signals into bandwidth channels
We can design filter circuits capable of selectively filtering
out one frequency or a range of frequencies amongst a range
different frequencies in a circuit
Characteristics of filters
Any filter (be it analog or digital) is designed to pass signals of
certain frequency contents without any distortion must have:
frequency response with a maximum magnitude value of
ONE at those frequencies of interest >> passband range
frequency response with magnitude value of ZERO at all
other frequencies to totally block the signal components
with those frequencies >> stopband or attenuation range;
Since analog circuit performs filtering on signals that pass
through the circuit >> it is important for the circuit designer
to understand how to design and construct filters to achieve
the best filtering results
Now, suppose we have a simple LTI discrete filter system that
has a length of 3 and impulse response h[n] as h[n] = {a, b, a}.
Cont’d…..
We can define an expression to describe this filter since we
know the components of the impulse h[n] as
h[0] = a, h[1] = b, and h[2] = a
Using the convolution sum expression, we can determine the
output y[n] of the filter system as :
>> y[n] = h[0]x[n] + h[1]x[n-1] + h[2]x[n-2]
To find the filter transfer function, we perform z-transform
which gives our filter
>> H(z) = h[0] + h[1]z-1 + h[2]z-2
Above filter is non-recursive type of filter (no feedback) and
has coefficients h[0], h[1], and h[2]
Cont’d…..
We can sketch the filter system block diagram, and find the
magnitude response characteristics
To find the frequency response H(.) of the system, we replace
the complex z with ejw in the transfer function H(z) :
>> H(ejw) = e-jw [b + a(ejw +e-jw )] = e-jw [b + 2acos(w)]
We can find the frequency response of system by finding the
discrete Fourier transform of system impulse response h[n]
It is clear from above that we can evaluate a filter system (how
filter performs) performance in time or the frequency domain
To examine the filter in each of the domains require certain
key parameters that define the filter in that domain
Time domain filtering concept
To evaluate how a filter is performing in the time domain, we
rely on the key parameters such as impulse response, a step
response, and frequency response
Step response of a filter system is obtained by integrating the
impulse response while the frequency response is obtained by
from the impulse response by using the Fourier transform
We use the step response of a system to measure or evaluate
how well the designed filter performs in the time domain
Three step response parameters important in design of filters :
Rise time or transition speed of the step response
common way to specify this is to quote the number of
samples between 10% and 90% amplitude levels
Overshoot of the response
Cont’d…..

Cont’d…..
Overshoot of the response
this must be generally eliminated because it changes
amplitude of the samples in signal >> distortion
Symmetry of top and bottom halves of step
it is desired that upper half part of the step response be
symmetrical with the lower half >> symmetry is
referred to as linear phase > frequency response has
phase that is a straight line
symmetry is needed to make the rising edges look the
same as the falling edges

Cont’d…..

Cont’d…..

Frequency domain filtering
To evaluate how a filter is performing in frequency domain,
we rely on key parameters such as frequency response,
ripples (or distortions) in the pass-band and stop-band,
and transitions (or roll-off)
Frequency (magnitude) response of a filter is characterized by
bands that describes which frequency band the filter should
pass (passband) signals, which band the filter should reject
(stopband) signals, and transition (roll-off) between 2 bands
Different filter design methods lead to different frequency
response characteristics
Some design methods emphasize on low distortions in passband
range of the filter while others stress on the ability of the filter to
attenuate the signal in the stopband
Cont’d…..
Between the passband and the stop band is a transition band,
which is often referred to as the roll-off
fast roll-off of a filter system means we have a very narrow
transition (approaches the ideal condition) while a slow
roll-off means a wide transition
division between the passband transition is what we call the
cut-off frequency
With analog filter design, it is usually defined to be
where the amplitude is reduced to 0.707 (i.e., -3dB)
With digital filter design, it is common to see 99%, 90%,
50%, etc., of the amplitude level

Cont’d…..

Cont’d…..

Digital filter types
Digital filter and algorithms used in DSP come in two form :
>> FIR (finite impulse response) filter algorithms
>> IIR (infinite impulse response) filter algorithms
FIR (filters) are composed of many parts but the key parts that
are used are: multipliers, adders, and delay elements.
Since multipliers are complex parts, we often replace them
with simple parts in arithmetic operations >> example, to
multiply by 8 (23), it is easier to shift number 3 places left
Delay elements can be thought of as registers (implemented
as registers) >> used to store values for short period of
time >> two registers in a row has effect of delaying the
signal by two time units >> output appears as time shifted
version of the input.
FIR digital filters
Suppose a system has an impulse response h[n], then using the
convolution sum, system can be described by input-output as:
>> y[n] = ∑h[k]x[n-k]
Transfer function H(z) of FIR filter system h[n] has polynomial
that can be described in the form of the convolution sum as:
H(z) = ∑h[n]z-n where n ranges from 0 to N
Thus, the time domain of input-output relationship of a causal
FIR filter system of length M can be expressed as :
y[n] = ∑kh[k]x[n-k] for 0 ≤ k ≤ M-1
H(z) = ∑kh[k]z-k for 0 ≤ k ≤ M-1
FIR filters are also known as non-recursive filters because
they are not characterized by an feedback loops
FIR digital filters
Cont’d….
From expression above, if the system impulse response of the
system h[k] is non-zero only at finite points (say, k =1, 2,
3,…M), and zero otherwise for k outside the defined range,
we describe such type of filter system as FIR digital filter.
Thus, a discrete time system is said to be FIR filter system if
its impulse response h[n] has zero-valued samples for n > M
where integer M (must be greater than zero) denotes the
length of the impulse response.

Characteristics of FIR digital filters
FIR digital filters are characterized by following properties:
Usually consists only of zeros (i.e., no poles or poles are
located at the origin of the zero-pole map)
Has no feedback in the system block diagram, which means
that any rounding errors are not compounded by summed
iterations.
Inherently stable due to the fact that all the poles are located
at the origin, which is within the unit circle of the zero-pole
FIR filters however, require considerably more computation
power compared to an IIR filter with the same filter order
or similar sharpness especially when low frequencies cutoffs
are needed

Advantages of FIR digital filters
FIR digital filters exhibit linear phase characteristics or
can be easily designed to be linear phase >> linear phase
filters delay the input signal but do not distort its phase
They are simple to implement >> on most DSP processors,
FIR calculations can be done by looping a single instruction
They can be implemented using fractional arithmetic >> can
be implemented using coefficients with magnitude of less than
1 >> important consideration when using fixed point DSPs
They have desirable numeric properties (finite number of bits)
>> because FIR does not have any feedback, fewer bits can be
used in the implementation of FIR filter y can be implement

Disadvantages of FIR digital filters
Sometimes they require more memory and /or calculation or
computational power to achieve a given filter response
characteristics
Certain responses are not practical to implement with FIR
filters

IIR digital filters
From convolution sum for the output of a system with h[n]:
>> y[n] = ∑h[k]x[n-k]
If the system impulse response h[k] non-zero (≠ zero) for
–∞ < k < +∞, we refer to this type of filter as IIR filter.
IIR digital filters thus, have impulse response that lasts forever
due to the presence of feedback within the filter structure
IIR filters are also known as recursive filters are typically
characterized by both feed-forward and feed-back coefficients
in the difference equation
Transfer function of practical IIR filters are characterized by
the ratio of polynomials with zeros and poles of the form:
H(z) = D(z)/P(z) where D(z) = a0+a1z-1+a2z-2 +...+aNz-N
and P(z) = d0+d1z-1+d2z-2+……..+dMz-M
Characteristics of IIR digital filters
IIR digital filters are characterized by following properties:
Usually characterized by feedback topology or have poles
(and perhaps zeros)
Compared with the IIR filters, FIR filters are generally
more expensive to implement but offer several attractive
design characteristics
Generally, IIR filter is preferable if phase distortion is not
important, because implementation of IIR filters involves
fewer parameters. Example, y[n] = 0.9y[n-1]+0.1x[n]
requires less memory and computation
Almost all analog electronic filters are IIR based

Advantages and limitations of IIR
Advantages of IIR digital filters:
Main advantage of digital IIR filter over FIR is its efficiency in
implementation in order to meet specification in terms
passband, stopband, ripple, and/or roll-off >> fewer number
of calculations per time step
Have much better frequency response due to the feedback
Disadvantages of IIR digital filters:
Phase characteristics is not linear which can cause problems to
systems that need linear phase property
Not easy to design to match a particular frequency response
requirement as compared with FIR
Potential instability and for limit cycle behaviuor when idle
due to the feedback system in conjunction with quantization
CPEN 304 Lecture 4
25
Simple Lowpass FIR digital filters
From the moving average filter, we can derive the basic low
order lowpass FIR digital filter which is defined as :
>> y[n] = ∑k(1/M) x[n-k] for 0 ≤ k ≤ M-1
From the moving average expression above, we can recognize
the impulse response h[n] of the system as:
>> h[k] = 1/M for 0 ≤ k ≤ M-1
>> h[k] = 0, otherwise
When we set M = 2, we obtain the first order lowpass FIR
digital filter. Thus, with M = 2, expression reduces to form
>> y[n] = 0.5x[n] + 0.5x[n-1]
The impulse response h[n] then becomes h[n] = {h[0], h[1]}
= h[n] = {½, ½}
Cont’d…..
The transfer function H(z) of the linear difference equation of
the system, the zero-pole map, and the frequency response
H(z=ejw) of the system are given as:

Cont’d…..
From the transfer function, we observe that the system has a
zero at z = -1 and a pole at z =0.
Finding the frequency response of H(z) give the expression :
>> HLP(e-jw) = e-jw/2cos(ω/2)
From above, the magnitude response |H(e-jw)|will be given
by cos(ω/2), which is a decreasing function of the frequency.
From the magnitude response curve, we observe a decreasing
function of frequency from w = 0 to w = p.
The magnitude response has a maximum value of unity (one)
which occurs at w = 0 and a minimum value of zero at w = p.
We refer to the frequency at which ω = ωc on the magnitude
28
response curve
CPEN 304 Lecture 4 as the 3-dB cutoff frequency
Cont’d…..
The 3-dB cutoff frequency is the frequency at which the power
of the magnitude response reduces to half its maximum value
>> identify maximum value of |H(.)|2, find the frequency at which
we obtain half of the maximum, what is that frequency
|H(e-jwc)|2= (1/2)|H(ej0)|2
|H(.)|wc = √(1/2)|H(.)|max
Student to find the dB expression of the above 3-dB point
For our first order lowpass FIR digital filter, the magnitude is:
>> |H(e-jw)|2= [cos(ωc/2)]2= 1/2
Now, to determine the cutoff frequency, (ωc), we set |H(.)|2
equal to square of cos(ωc/2) = half maximum power
|H(e -jwc)2|= cos2(ω /2) = 1/2 >> ω = 2cos-1(1/√2)
29 CPEN 304 Lecture 4 c c
Higher order Low pass FIR filters
Higher order lowpass FIR filters can be obtained by cascading
many of the first order FIR filter or transfer function
FIR filter with order of 3 can be obtained by cascading three
of the HLP(z) = {0.5(1+z-1)}*{0.5(1+z-1)}*{0.5(1+z-1)}
To find the cut-off frequency for higher orders, we generalize
the 3-dB cutoff frequency for the first order lowpass FIR filter
which occurs at ωc = π/2.
We generalize the 3-dB cutoff frequency for a cascade of M
sections of first order FIR lowpass filter to obtain any higher
order FIR lowpass filter as :
>> wc = 2cos-1(2-1/2M)
Example, for M = 3 (gives a second order lowpass FIR filter),
the 3-dB cutoff frequency will be wc = 0.3002p.
Simple High pass FIR digital filters
To obtain the first order highpass FIR digital filter, we
simply replace the “z” in the lowpass transfer function with “-z”
Thus, transfer function H(z) for the first order FIR highpass
filter will be determined from the lowpass expression as :
>> Hhp(z) = 0.5(1-z-1)
Finding the frequency response of Hhp(z) give the expression :
>> Hhp(ejw) = je-jw/2sin(ω/2)
Transfer function of the first order highpass filter above has a
zero as z =1 or ω = 0 , which is in the stopband of the filter
The frequency response also has a monotonically increasing
behaviour from a value of zero to a maximum magnitude of 1
The 3-dB cut-off frequency for the highpass filter also occurs
31
at ω c
CPEN
=
304
π/2.
Lecture 4
Higher order Highpass FIR filters
Higher order highpass FIR filters can be obtained by cascading
several sections of the first order highpass filter function
FIR hiphpass filter of order 3 can be obtained by cascading 3
of the HHP(z) = {0.5(1-z-1)}*{0.5(1-z-1)}*{0.5(1-z-1)}
Alternatively, we can be obtained higher order FIR highpass by
replacing the “z” with “-z” in the moving average expression
>> HHP(z) = 1/M∑(-1)n z-n , for 0 < n < M-1
Example, find the highpass FIR filter for M = 3 and the 3-dB
cutoff frequency of the filter

Simple Low pass IIR digital filters
Recall that IIR digital filters are characterized by feedback
with poles and zeros >> IIR digital filters can therefore be
represented as polynomial function with a constant coefficient.
First order lowpass IIR digital filter will have transfer
function that can be expressed in the form :
>> HLP(z)= {[(1- α)/2]* [(1 + z-1)/(1 - αz-1)]}
Parameter α denotes the stability of the filter and for
stability of the filter system, lαl <1.
Observe that the transfer function H(z) above has a zero
location at z=-1 and a pole location at z=α
Finding frequency response of H(z), we observe that as the
frequency ω increases from 0 to π the magnitude response
decreases from a value of 2 to zero.
Cont’d…..
With positive value of α, the magnitude of the pole increases
from (1-α) to (1+α).
If we set ω=0, and ω=π , we obtain the maximum and
minimum values of the magnitude response.
To determine the 3-dB cutoff frequency for the lowpass first
order IIR filter, we find squared magnitude response and
equate to half the maximum power similar to the FIR as:
jw1 (1 - a ) 2 [1 + cos(wc)]
2
H LP (e ) = =
2 2[1 + a 2 - 2a cos(wc)]
We now proceed to find the value of the cutoff frequency
from the above expression
Cont’d…..
Thus, the frequency response at cutoff frequency will be:
>>|HLP(e-jwc)|2= ½ = [(1- α)2/4 ]*[(1+e-jwc)
*(1+e+jwc)]/[(1 - αe-jwc)(1- αe+jwc)] = 1/2
>> cos(ωc) = 2α/[1+ α2]
The quadratic function above can be solved which will
produced two solutions >> one of the solutions will lead to
stable condition and other will not
Solution resulting in a stable transfer function is given as:
>> α = [1-sin(ωc)]/cos(ωc) >> filter design
With knowledge of the system stability value α, we can
determine transfer function HLP(z) of the lowpass IIR filter
Simple High pass IIR digital filters
We obtain the highpass IIR digital filter from the lowpass IIR
transfer function similar to the FIR as follows:
Replace “z-1” in the lowpass IIR transfer function with “-z-1”
replace the stability parameter “+α” in the coefficient of the
transfer function with “-α”
First order highpass IIR digital filter will have transfer
function that can be expressed in the form :
>> HhP(z) = [(1+ α)/2]*[(1 - z-1)/(1 - αz-1)]
The cutoff frequency for the highpass first order IIR filter is
obtained using the same process outlined earlier, thus, square
of magnitude response is equal to half max power, which gives
>> cos(ωc) = 2α/[1+ α2] and the stability parameter as
>> α = [1-sin(ωc)]/cos(ωc) >> filter design
IIR digital filter example 1
Question 1:
Design a first order highpass IIR digital filter to meet a 3-dB
cutoff frequency of 0.8π (try this 0.95π)
Solution 1:
From the question, cutoff frequency is 0.8π >>
>> sin(ωc) = sin(0.8π) = 0.58778525 (0.1564)
>> cos(ωc) = cos(0.8π) = -0.80902 (-0.9880)
Now using these in the equation for α above we have :
>> α = [1-sin(ωc)]/cos(ωc) = - 0.50952449 (-0.845)
We now substitute into the transfer function expression as:
HhP(z) = [(1+ α)/2]*[(1 - z-1)/(1 - αz-1)]
= 0.24523[(1-z-1)/(1 + 0.50952449z-1)]
Higher order IIR digital filters
Higher order IIR Lowpass filters
Similar to the case of FIR, we obtain higher order IIR filters by
cascading sections of the first order IIR filters
An M-th order IIR lowpass filters can be obtained using the
general structure as:
>> HLP(z) = [(1- α)/2]M * [(1 + z-1)/(1 - αz-1)]M
The 3-dB cut-off frequency ωc of the M-th order lowpass IIR
filter can be obtained by finding the frequency response and
equating to the half maximum power condition
Thus, expression gives solution for the stability parameter as:
>> α = [1+(1-K)cos (ωc) – sin(ωc(2K-K2)1/2]/[1 - K
+ cos(ωc)]
>> K = 2(M-1)/M
Higher order IIR digital filters
Cont’d…….
From the above expression, when M = 1 (first order filter),
the stability expression reduces to the form :
>> α = [1-sin(ωc)]/cos(ωc)
Higher order IIR highpass filters
To find the higher order IIR highpass filter, we modify the
transfer function of the lowpass HLP(z) as was done the in the
first order case
The 3-dB cut-off frequency for the higher order highpass will
be the same as that of the lowpass as demonstrated earlier

IIR higher order filters example 1
Question 1:
Design a first order lowpass IIR digital filter to meet a 3-dB
cutoff frequency of 0.4π. Now, cascade the single stage filter
section to realize an IIR lowpass filter with an order of 4.
Solution 1:
For first order, we set M = 1 in the equation and obtain K = 1
Using the cutoff frequency ωc = 0.4π and K = 1 in the above
expression gives α = 0.1584
For the 4th-order , we set M = 4 in the equation and obtain
the value of K = 23/4 = ??
Using the cutoff frequency ωc = 0.4π and K = 23/4 in the
above expression gives α = -0.2510
40
We can
CPEN then4 substitute α and K into the HLP(z) equation
304 Lecture
Simple Bandpass IIR digital filters
Recall the circuit diagram for bandpass filter as comprising a
cascade of a single pole lowpass and highpass filters >>
Bandpass filters are characterized by second order filters.
Second order bandpass IIR digital filter will have
transfer function that can be expressed in the form :
HBP(z) = [(1- α)/2]*[(1 - z-2)/(1 - β(1+ α)z-1+αz-2)]
Similarly, the square magnitude function is given by:
2
H ( e jw ) =
(1 - a ) 2
(1 - cos 2w )
BP
4 [1 + b 2 (1 + a ) 2 + a 2 - 2 b (1 + a 2 ) cos w + 2a cos 2w
Magnitude response function above assumes unity value when
the frequency ω=ω0, called center frequency of the Bandpass

Simple Bandpass IIR digital filters
Cont’d…..
This center frequency, ω0 of the bandpass filter is given by:
>> ω0 = cos-1(β) = >> β = cos(ω0)
The frequencies ω1 and ω2 where the squared magnitude
(|H(.)|2) goes to half (1/2) are referred to as the 3-dB cutoff
frequencies and the difference assuming is ω2 > ω1 is known
as the 3-dB Bandwidth.
Thus, the 3-dB bandwidth BW is
>> BW = ω2 – ω1
>> = cos(ω2 – ω1) = 2α/[1+ α2]
>> BW = ω2 – ω1 = cos-1(2α/[1+ α2])

Bandpass IIR filters example 1
Design a 2nd order IIR Bandpass filter with a centre frequency
of ωo = 0.4π and a 3dB bandwidth (BW) of 0.1π.
To determine the transfer function of the filter, we need to
find the values for all the parameters in the expression.
Using the design values, we determine the parameters as:

Bandpass IIR filters example 1
Cont’d…..
Note that in this bandpass example, when we solve the
quadratic equation for α, we obtain two solutions as:
>> α1=1.37638192 and α2 = 0.726542528.
These solution will give us two transfer functions. However,
to know which of the two transfer functions is the appropriate
choice, we have to test for the stability of the two systems.
In this case, we check for the poles of the two systems. If the
magnitude is outside the unit radius ROC criteria, then we
ignore it as an unstable system.
Student to check this calculations.

Simple Bandstop IIR digital filters
Similar to the other IIR digital filters, the bandstop filter can
be determined from the following transfer function:

Exercises for digital filters
Question 1:
(a) Design a 5th order lowpass FIR digital filter to meet the 3-
dB cut-off frequency of 0.24π using a cascade of five first
order lowpass filters
(b) Design a 6th order highpass FIR digital filter to meet the 3-
dB cut-off frequency of 0.24π using a cascade of six first
order highpass filters
Question 2:
(a) Design a first order lowpass IIR digital filter to meet the
normalized 3-dB cut-off frequency of 0.3π
(b) Design a 5th order lowpass IIR digital filter to meet the
normalized 3-dB cut-off frequency of 0.3π using a cascade of
five first order lowpass filters
Exercises for digital filters
Cont’d……
Question 3:
(a) Design a first order highpass IIR digital filter to meet the
normalized 3-dB cut-off frequency of 0.3π
(b) Design a 5th order highpass IIR digital filter to meet the
normalized 3-dB cut-off frequency of 0.3π using a cascade of
five first order highpass filters
Question 4:
(a) Design a 2nd order bandpass IIR digital filter to meet a
centre frequency of 0.7π and bandwidth of 0.15π
(b) Design a 2nd order bandstop IIR digital filter to meet a
centre frequency of 0.55π and bandwidth of 0.25π

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