Chapter 4: Problem Solutions

Digital Filters

Problems on Non Ideal Filters

Problem 4.1

We want to design a Discrete Time Low Pass Filter for a voice signal. The specifications are:
Passband Fp 4kHz, with 0.8dB ripple;
Stopband FS 4.5kHz, with 50dB attenuation;
Sampling Frequency Fs 22kHz.

values of H in the Passband and the Stopband, where H is the filter frequency response.
Determine a) the discrete time Passband and Stopband frequencies, b) the maximum and minimum

Solution

a) Recall the mapping from analog to digital frequency 2F Fs , with Fs the sampling fre-
quency. Then the passband and stopband frequencies become p 2 4 22 rad 0.36 rad,
s 2 4.5 22 rad 0.41 rad;
b) A 0.8dB ripple means that the frequency response in the passband is within the interval 1

frequency response within the passband is within the interval 0.9035 H 1.096.
where is such that 20 log10 1 0.8 This yields 100.04 1 0.096. Therefore the

Similarly in the stopband the maximum value is H 105020 0.0031

Problem 4.2

A Digital Filter has frequency response H such that

0.95 H 1.05
0 H 0.005
for 0 0.3
for 0.4
2 Solutions_Chapter4[1].nb

Also let the sampling frequency be Fs 8kHz. Determine the Passband and Stopband frequencies in
kHz, the Passband ripple and the Stopband attenuation in dB.

Solution

20log10 0.005 46dB. The analog passband frequency is 0.3 Fs 2 1.2kHz and the
The passband ripple is given by 20 log10 1.05 0.42dB, and the attenuation in the stopband

stopband 0.4 Fs 2 1.6kHz

Problem 4.3

A continuous time filter has frequency response

HF
1
1


j2F
1000

Determine the passband and stopband frequencies in Hz, assuming a passband ripple of 1dB and
attenuation of 40dB in the stopband. Also determine the half power frequency Fc .

Solution.

1 HF 1 with 20log101 1, which yields 0.12. Since

A passband ripple of 1dB means that the frequency response is within the interval

HF

2F
2

1
then we determine the passband from the equation
1
1000

2F
2


1
1 0.12 0.88
1
1000

HF 104020 0.01 which yields F 15, 914 Hz Notice that this filter has a very long
which yields F 85.9Hz. Similarly for the stopband, we need to determine the frequency where

transition region, as we can see from the plot of its magnitude:

Solutions_Chapter4[1].nb 3

dB
FHz
5000 10000 15000 20000

-10

-20

-30

-40

Problem 4.4

A Digital Filter is defined by the difference equation

yn 0.99yn 1 xn
The filter is clearly recursive. Determine the impulse response hn.
a) Is the filter stable?
b) Would you classify it as Low Pass, Band Pass ... or what?
c) Would you feel comfortable in implementing this on a digital machine?

Solution

a) The filter is stable since its transfer function Hz

11
10.99z
z
z0.99 has one pole at
z 0.99;
b) It is a low pass filter since it has one pole close to z 1, ie 0. This makes the frequency
response "large" at small frequencies. A plot of its magnitude is as follows:
4 Solutions_Chapter4[1].nb

G
40

30

20

10

0.5 1 1.5 2 2.5 3

Problem 4.5

A simple averaging filter is defined as

yn
N xn 1 ... xn N
1

This is clearly an FIR Filter.

a) Let N 4. Determine the transfer function, its zeros and poles;
b) Determine a general form for zeros and poles for any N;
c) By comparing yn and yn 1 determine a recursive implementation. Also the transfer func-
tion, together with its zeros and poles of the recursive implementation. Looking at this example, can
we say that "any" recursive filter is IIR?

Solution

a) With N 4 we obtain the transfer function Hz 1 z2 z3 z4 . After normaliza-

4 z
1

tion this becomes

1 z z z1
3 2
Hz
4 z
4
The are four poles at z 0 and three zeros from the solution
4
z3 z2 z 1
1z 0
1z

Therefore the zeros must be such that z4 1, with the exclusion of z 1. That is to say z4 ejk2
for k 1, 2, 3, and therefore the zeros are z jk with k 1, 2, 3, ie z j, 1, j.This is
shown in the z-plane below.
Solutions_Chapter4[1].nb 5

Im

Re

b) Since the transfer function is of the form

N
Hz
N
1 1z

zN 1z

the zeros are of the form z ejk N , k 1, ..., N 1 and the poles are all at z 0.
2

c) Since yn
1 xn 1 ... xn N and
N
yn 1
1
N xn 2 ... xn N 1 by comparing yn and yn 1 we see
that
yn yn 1
N xn 1
1
N xn N 1
1

This yields the transfer function

1 z z 1 N1 N
Hz
N
1z
1
N
1 1z

zN 1z

as we saw before. This is an example of a recursive filter with finite impulse response (FIR).

Problems on FIR Filters

Problem 4.6

We want to design a Low Pass FIR Filter with the following characteristics:
6 Solutions_Chapter4[1].nb

Passband 10kHz,
Stopband 11kHz, with attenuation of 50dB,
Sampling frequency 44kHz
Determine the causal impulse response hn, and an expression for the phase within the passband.
Use one of the standard windows listed in section 4.3.

Solution

First we have to determine the specifications in the digital freq. domain.

Passband: p 2 10 44 0.4545 rad
Stopband: S 2 11 44 0.5 rad

2 p S
Therefore we choose the passband of the ideal filter as C 44 0.477. We
1 21

tion region has a width of 12 N. Since we want a transition region S P 2 44 we deter-

need a Blackman window to satisfy the 50dB attenuation in the stopband. With this window the transi-

mine the filter length N as

44 12
2
N

which yields N 12 22 264. Therefore we choose N 265 and a shift L 132. Finally the
impulse response is
hn hd n 132 wBlackmann
n132 wBlackmann
sin0.4545 n132

which is shown below.

Solutions_Chapter4[1].nb 7

Within the passband the phase is linear and it is given by the expression
H L 132

Problem 4.7

Repeat Problem 2.1 with an equiripple filter using the "remez" function in Matlab. Plot the two
frequency responses and compare the two filters in terms of performance and complexity.

Solution

With Matlab we need first to determine the order of the filter. Use the function "remezord" as follows:

N, fo, mo, k remezord10000, 11000, 1, 0, delta, delta, 44000;

with delta 10 ^ 50 20 the maximum deviation corresponding to 50dB's. This yields an
order N 114, in the sense that the transfer function is of the form
Hz h0 h1z1 ... h114z114
The impulse response hn is obtained as
h remezN, fo, mo, k
where fo, mo and k are from remezord.
8 Solutions_Chapter4[1].nb

Notice that the order of the equiripple filter N 114 is considerably smaller than the order of the
filter designed with the Blackman window in Problem 4.6.

Problem 4.8

Repeat Problem 4.6 using the Kaiser window.

Solution

Wi the Kaiser window we have to determine the parameters N and b from the specifications. In particu-
lar we want an attenuation A 50dB which yields a factor from the expression
0.5842A 210.4 .07886A 21 4.53
Also the filter length is determine from the expression
N
2.285


22 128.717
A8 42
2.285

So we can choose N 129 and L 64. The frequency response of the filter therefore becomes
sin0.4545 n64
wKaisern

n64

Its magnitude is shown below.

Solutions_Chapter4[1].nb 9

Problem 4.9

e0.1F if F 10 Hz
We want to approximate a filter with frequency response

HF
0 if F 11Hz
Let the sampling frequency be Fs 50Hz, and the attenuation in the stopband be 40dB. Determine
the impulse response of a FIR filter which approximates this frequency response. Plot the frequency
response in terms of magnitude and phase to verify that the approximation holds.

Solution

In the digital domain, let 2F Fs and therefore F Fs 2. Therefore the filter's desired

e52 if 2 5 rad
frequency response becomes

H
0 if F 2.2 5 rad
The ideal filter therefore is going to have a frequency response Hd given by
2.1 5
Hd
e52 if
0 otherwise
and the impulse response
10 Solutions_Chapter4[1].nb

hd n
2.15
2
1 e52 ejnd

2.274050.795775 Cos1.31947 n1. n Sin1.31947 n
5.685128.97758 n2
2.15

tion region of width 8 N. The desired width is 2 50 and therefore N is determined from
Since we want 40dB attenuation in the stopband we can use a hamming window, which has a transi-

the equation

50
2 8
N

and N 50 4 200. Choose N 201 and L 100. This yields the impulse response
hn hd n 100whammingn

Problem 4.10

A bandpass filter needs to be designed, to pass a signal within frequencies 4kHz and 8kHz, with
two transition regions not exceeding F 0.5kHz . Also we want the attenuation in the stopband
not exceeding 50dB, and the same error within the passband. Finally let the sampling frequency be
Fs 44kHz.
a) Determine the impulse response of the ideal filter;
b) Design the filter using the Kaiser window;
c) Design the filter using the equiripple method.
Compare the two frequency responses.

Solution

between 2 4 44 2 11 rad and 2 8 44 4 11 rad. Therefore the impulse

a) In the digital frequency domain we want to design a bandpass filter which passes the frequencies

response of the ideal filter is

11 Sin
hd n
211 2n 4n
411 Sin

11
if n 0
2 ejnd
2 ejnd
1 1 n
211 411 0 if n 0

attenuation A 50dB and the width of the stopband 2 0.5 44 44 rad. Recall the
b) From the formulas of the Kaiser window we determine the parameters and N as follows from the

formulas:
Solutions_Chapter4[1].nb 11

0.5842A 210.4 .07886A 21 4.53

N
A8
2.285
257.434

which yields a window of length N 259 and a time delay L 129. Finally thr impulse response
becomes
hn hd n 129wkaisern

Problem 4.11

You want to design a low pass filter with passband Fp 2kHz and stopband FS 2.5kHz, with
attenuation of at least 40dB. Let the sampling frequency be Fs 10kHz. Using the techniques you
know, determine the design with the least number of coefficients.

Solution

First we translate the specifications into the digital frequency domain:

Passband p 2
10
2 2
5

Stopband S 2
10
2.5
2


2
2 5 10

We know three techniques:

a) Window based: from the desired attenuation we need a hamming window. From the transition
region


N
8
10

we obtain the length of the filter N 81;

b) Kaiser window: applying the formulas with A 40 and 0.1 we obtain
N
408
2.2850.1 45
c) Equiripple Filter: using the matlab function "remezord" we obtain the order N 39 which yields
the lowest complexity. The corresponding frequency response is shown below (magnitude only in
dB's).
12 Solutions_Chapter4[1].nb

Problem 4.12

In the section on equiripple filters we have taken a few things for granted. In this problem try to prove
the following simple facts:
a) when the order of the filter N increases, the maximum error decreases;
b) the solution is unique, in the sense that, for any given order N, there is only one impulse response
h0, ..., hNwhich minimizes the maximum error.

Solution

a) Call hN hN 0, ..., hN N the optimal solution (in the minmax sense) of order N. Then

the vector
hN1 hN 0, ..., hN n, 0 hN1

represents an impulse response of order N 1, not necessarily optimal. Therefore

ehN1 ehN1 ehN

The leftmost inequality is due to the fact that hN1 is not the optimal solution of order N 1, and the
rightmost equality is due to the fact

HN1 HN for all
Solutions_Chapter4[1].nb 13

where HN1 and HN are the frequency responses of hN1 and hN .

b) If there were two different solutions hN and
gN of the same order N having the sign alternation

property, the difference HN GN would have N 1 roots against the assumption of being
polynomials (in cos) of order N.

Problem 4.13

A Hilbert Transform is a filter with frequency response

Hd jsign
with sign 1 for 0 being the signum function.
a) Plot the magnitude and phase of the filter;
b) Determine the impulse response hd n;
c) Determine a causal approximation h0, ..., hN using a rectangular window. Plot the
magnitude of the frequency response for various values of N, say N 40, 60, 120. Does it con-
verge everywhere? In this case what would you call the transition region?

Solution

a) Since Hd jsign 1 for all , and

2 if 0
PhaseH
2 if 0
we obtain the plot below.

| H ( ) |
1

H ( )

2

2
14 Solutions_Chapter4[1].nb

b) The impulse response becomes

hd n
2 jsigne
2
n 2
2 Sin
jn d
n if n 0
1

and hd 0 0.

c) The plots of

HL hd nejn
L
nL

are shown below (magnitude only) for L 20, 40, 60 corresponding to the orders
N 40, 60, 120 respectively. The transition region is defined around the frequency 0.

N40
1.2
1
0.8
0.6
0.4
0.2

3 2 1 1 2 3
N60
1.2
1
0.8
0.6
0.4
0.2

3 2 1 1 2 3
Solutions_Chapter4[1].nb 15

N120
1.2
1
0.8
0.6
0.4
0.2

3 2 1 1 2 3

Problems on IIR Filters

Problem 4.14

Using the Bilinear Transformation, determine the order N and the cut off frequency c of the analog
prototype filter for the following discrete time design:
a) passband 8kHz;
b) stopband 9kHz;
c) passband ripple 0.5dB;
d) stopband attenuation 40dB;
e) sampling frequency Fs 44kHz.

Solution

First we define the problem in the digital frequency domain:

2 Fp
p
Fs 1.1424 rad
2 FS
S
Fs 1.2852 rad

Then we determine the specifications of the Analog Prototype:

16 Solutions_Chapter4[1].nb

2 56554.2 rad sec

p

2 65876.0 rad sec

p Fs 2 Tan
S
S Fs 2 Tan
Now from the Passband Ripple we determine e as

p 100.520 0.944

p 1 0.349
1 2

Then we determine the order of the filter

2

2
log
1s
N
2
37.0762

2 log
s
s
p

which yields N 38. Finally we determine the cut off frequency of the filter as
c p 1N 58141.4 rad sec
and therefore its frequency response becomes

p
H
1

2N
1

2N
p 1N

1
2 1

The following plot shows the poles of the filter in the s-plane:
Solutions_Chapter4[1].nb 17

-5 -4 -3 -2 -1

-2

-4

-6

The frequency response (magnitude only) is shown next:

20000 40000 60000 80000 100000

-25

-50

-75

-100

-125

-150

-175
18 Solutions_Chapter4[1].nb

Problem 4.15

A 4-th order Butterworth filter has cut off frequency c 200 rad sec.
a) Determine the zeros and poles of the transfer function;
b) What would be its passband and stopband frequencies if we want 1dB ripple in the passband and
40dB attenuation in the stopband?
c) If we apply a Bilinear Transformation with sampling frequency Fs 1kHz, determine the zeros
and poles in the z-plane.

Solution

c 200 rad sec spaced by an angle of 360 2N 360 8 45 degrees. The poles are
a) Recall that an N th order Butterworth Filter has poles on a circle with radius

shown in the figure below

s plane

and they are given by p1,4 200ej58 and p2,3 200ej78 . All zeros are at s .

b) From the frequency response H 8 we solve for p and S , as

1

1 200

Hp 10120 530.673 rad sec

HS 104020 1986.89 rad sec

c) Applying the formula for the Bilinear Transformation each pole is mapped as
s
2
z
2
T
s
s

Ts
Solutions_Chapter4[1].nb 19

This yields poles in the z-plane at 0.673045 j 0.433479 and 0.53675 j0.143193 and
four zeros at z 1.

Problem 4.16

Repeat Problem 4.14 using a Chebyshev filter. Which one would you choose if complexity is an issue?

Solution

The specifications of the prototype filter are the same, since they depend on the original specifications
of the filter and on the bilinear transformation. Recall them here for convenience:
2 Fp
p
Fs 1.1424 rad
2 FS
S
Fs 1.2852 rad

2 56554.2 rad sec

p

2 65876.0 rad sec

p Fs 2 Tan
S
S Fs 2 Tan
p 10120 0.944
S 104020 0.01

p 1 0.349
1 2

Then use the formulas for the Chebyshev Filter, from section 4.4, to obtain N 12 (the complexity of
the filter) and the frequency response as shown below.

The filter order is determined from the formula

2
1S 1 1S

2 2

log

S 2
S
N

log
p




S

p 1

which yields N 12. Then the poles of the filter in the s-plane are computed as
20 Solutions_Chapter4[1].nb

2 1 1

;
1N

p 1 2
r1 ;

p 2 1
2

r2 ;
2 k 1 2 k 1
2
p r2 cos j r1 sin , k 0, ..., N 1

2N 2 2N 2
The poles in the s-plane are shown below. Notice the two different scales for the Real and Imaginary
axis

40000

20000

-8000 -6000 -4000 -2000

-20000

-40000

The plot of the frequency response is shown below. If you compare it with the Butterworth filter in
Problem 4.14, notice that you obtain the same attenuation with a lower complexity (N 12 for Cheby-
shev and N 38 for Butterworth).

20000 40000 60000 80000 100000

-20

-40

-60

-80

-100
Solutions_Chapter4[1].nb 21

Problem 4.17

We want to implement the analog filter with transfer function

Hs
2s1

s2 s1

by a discrete time approximation, using the Bilinear Transformation. Let Fs 10Hz be the sampling
frequency.
a) Determine zeros and poles of both the analog filter Hs and the discrete time implementation
Hz;
b) Determine the Linear Difference Equation of the discrete time implementation;
c) Plot the frequency responses of the digital filter H and the analog filter H. Verify that you
can obtain one from the other by the appropriate frequency transformation.

Solution

a) The zeros of the analog system are s 1 2, and s (yes this is a zero too!). The poles are
the solution of s2 s 1 0 which yields s ej23

Applying the mapping sz

s
2
z
2
T
s
s

Ts

we can verify that the zeros are mapped as

s 1 2 z 0.9512
s z 1
and the poles
s ej23 z 0.947743 j 0.0822827

b) Since we know the zeros and the poles we can write the transfer function to be
z1z0.9512
z0.947743j 0.0822827z0.947743j
Hz K



0.0822827

with the constant K to be determined. Combining terms we obtain

z 0.0488z0.9512
2
Hz K
z2 1.8955z
0.9050
We can determine K by matching the value at one frequency component, say at s 0 z 1:
Hs s0
1 Hz z1 K
z 0.0488z0.9512
z1 10.2737 K
2
2s1

Equating the two we obtain K 1 10.2737 0.0973361. Finally the transfer function
s2 s1 z2 1.8955z 0.9050
22 Solutions_Chapter4[1].nb

z 0.0488z0.9512
2
Hz 0.0973361
z2 1.8955z
0.9050
and the difference equation

0.09763361 xn 0.0488xn 1 0.9512xn 2

yn 1.8955yn 1 0.9050yn 2

Problem 4.18

An integrator has transfer function

Hs
1
s

a) Determine a discrete time implementation using Euler's approximation, with sampling frequency
Fs 10Hz. Sketch the frequency response, and verify that it approximates the integrator for low
frequencies;
b) Same as a), using the Bilinear Transformation.

Solution

a) By the Euler approximation

1
s
1z
Ts

we obtain
Hz 0.1
z
z1

The frequency response is

j j
H 0.1
ej 1 0.1
e e

For small, we can approximate ej 1 and sin 2 2, and therefore we see that, for
2j ej2 sin2

small the discrete time system behaves like an integrator with frequency response
H j2F FFs 2
j Fs
1 1

The following plot compares the frequency response H with the ideal integrator. We can see
that in this case the Euler approximation gives a good approximation.
Solutions_Chapter4[1].nb 23

60

40

20

0.5 1 1.5 2 2.5 3

-20

b) With the Bilinear Transformation we have

1
Hz
2
1z1
Ts 1z

and the frequency response is

2


cos
H
0.1

The following plot compares the frequency response H with the ideal integrator. Even here
2 j sin 2

we can see that in this case the Euler approximation gives a good approximation. Notice that at
z 1 (ie ), the frequency response is zero, and in the dB plot it goes to .

100

50

0.5 1 1.5 2 2.5 3

-50

-100

-150

Problem 4.19

You want to design an analog Band Pass Filter which passes the frequencies in the interval
5kHz F 6kHz
24 Solutions_Chapter4[1].nb

with 1dB ripple in the passband. Let the the filter be Butterworth with order N 4.
a) Determine the frequency transformation Low Pass to Band Pass you would use;
b) Determine the frequency response of the corresponding Low Pass Filter, together with its zeros and
poles;
c) Determine the zeros and poles of the Band Pass Filter and its transfer function.

Solution

From the specifications we determine the lower and upper frequencies as

L 2 5000
U 2 6000
Then we choose a prototype Butterworth filter of order N 4 and with arbitrary cut off frequency,
say
C 1
which has poles at
2 k1
pk C
2
2
4 , k 0, 1, 2, 3

C s2 L U
In order to apply the proper transformation (Low Pass to Band Pass)

s U L
qs

we compute the poles of the bandpass filter from the equations

qs pk , k 0, 1, 2, 3
Each equation is quadratic and it yields two solutions. As a total we have 2 4 8 poles for the
bandpass filter which are given by
poles 1303.33 37418.3 , 1101.14 31613.4 ,
3004.15 35515.3 , 2800.76 33110.8 , 3004.15 35515.3 ,
2800.76 33110.8 , 1303.33 37418.3 , 1101.14 31613.4
The filter has also four zeros at s 0, due to the fact that q0 . From the zeros and the poles
we determine the transfer function. The magnitude of the frequency response is shown below
Solutions_Chapter4[1].nb 25

-16
610
-16
510
-16
410

-16
310
-16
210

-16
110

10000 20000 30000 40000 50000 60000

Problems Involving Material from Previous Chapters

Problem 4.20

Recall that, when we apply a digital filter to a continuous time signal, we need two analog filters:
Anti-Aliasing and Reconstruction. Due to hardware constraints and cost, these two filters cannot have
a very large order, and they represent a constraint in our design. Suppose we want to design a digital
filter for a signal having bandwidth of 8kHz, and we have to use a 5 pole Butterworth filter
(commercially available for low cost) for both antialiasing and reconstruction filters. Also we want
1dB passband ripple and 50dB attenuation in the stopband.

a) What do you think would be the minimum sampling frequency we can have, with these two analog
filters. (Hint: recall what is the passband and what is the stopband of the analog antialising and recon-
struction filters in terms of signal bandwidth and sampling frequency);

b) The 5 pole Butterworth filter you buy, often is based on switched capacitor technology. This allows
to select the cut off frequency c fairly easily, by adjusting the frequency of an oscillator. Given the
sampling frequency you determined in question Q1, determine a suitable value for the cut off fre-
quency c of the filter.
26 Solutions_Chapter4[1].nb

Solution

a) From the specifications of the problem, we have to pass all frequencies up to 8kHz. This yields
p 2 8000 rad/sec
We want 1dB ripple in the passband. This gives a value of as
0.509
Therefore the frequency response becomes

H
10


1

12 p

with e and p as given. Now we have to find the stopband frequency, from the requirement of 50dB
attenuation. This leads to the equation

H

p
1 5020


10 10
1
2

which we solve for W. This yields the stopband frequency

S 181950. rad/sec
or, in Hertz,
S
FS
2 28958.2 Hz

Now the sampling frequency Fs has to be such that

FS Fs 8000 Hz
which yields a sampling frequency
Fs FS 8000 28958.2 8000 36.9582kHz
b) The cut-off frequency C of the filter is obtained by solving

H



1 1
10

1 2 2
p

which yields C 57537.7 rad sec. The figure below illustrates the problem.
Solutions_Chapter4[1].nb 27

FPASS

FSTOP

8. 0 Fs 8.0 Fs F (kHz )

Sampling freq.

Problem 4.21

In your CD the data is sampled at 44.1kHz (CD quality), and we want to have a good sound quality up
to 21kHz. If you had a to use an analog Butterworth filter as a reconstruction filter, what would be the
order of your filter? Do you think you can reasonably build a filter with that complexity? (I do not
think so either!) The chapter on Multi Rate DSP is going to show you how the CD technology solves
this problem.

Solution

Since we want the filter to pass the signal and reject all frequencies above Fs 2, we can see that
passband and stopband frequencies become
p 2 21000 rad sec
S 244100 2 2 22050rad sec
Assuming a 1dB passband ripple and 40dB attenuation in the stopband, this would yield a frequency
response of the form

H
2
12


1

N
p

with e=0.509 and P as given. For an attenuation of 40dB we obtain N 108 after solving

HS
2
12


1
S

N 0.01
p

for the order N.