DSP Tut Solutions

Curtin University of Technology, Australia 3-15
YH Leung (2005, 2006, 2007)

Problems

1. Let ( )
c
x t be a signal with Nyquist rate
o
. Determine the Nyquist rate for each of the
following signals.
(a) ( ) ( 1)
c c
x t x t
(b) ( )
d
c
dt
x t
(c)
2
( )
c
x t
(d) ( ) cos
c o
x t t

2. Consider the system shown below where an ideal sampler is followed by an ideal
lowpass filter to reconstruct ( )
c
x t from its samples ( )
s
x t . From the sampling theorem,
we know that if 2 =
s
T is greater than twice the highest frequency present in ( )
c
x t
and 2 =
c s
, then the reconstructed signal ( )
r
x t will exactly equal ( )
c
x t . If this
condition on the bandwidth of ( )
c
x t is violated, then ( )
r
x t will not equal ( )
c
x t . We
seek to show in this problem that if 2 =
c s
, then for any choice of T , ( )
r
x t and
( )
c
x t will always be equal at the sampling instants, that is,
( ) ( ) , 0, 1, 2,
r c
x nT x nT n = =
x
c
(t) x
s
(t) x
r
(t)
2T T -T 0
t
( )
T
s t

( )
i
H j
c

c
T

To obtain the result, first show

( ) sin ( )
( ) ( )
( )
c c
r c
c
n
t nT
x t x nT T
t nT

which reduces to the following expression when 2 =
c s

( ) sin ( )
( ) ( )
( )
T
r c
n T
t nT
x t x nT
t nT

Then consider the values of for which ( ) sin 0 = .

YH Leung (2005, 2006, 2007)
3. The sequence
[ ] cos ,
4
1

= < <

( )
x n n n
was obtained by sampling a continuous-time signal
( ) ( ) cos ,
c o
x t t t = < <
at a sampling rate of 1000 samples/s. What are two possible positive values of
o
that
could have resulted in the sequence [ ] x n ?

4. The following continuous-time signals ( )
c
x t are sampled. Specify a choice for the
sampling period T that will result in the discrete-time sequences shown. In each case,
is your choice of T unique? If not, give another possible choice of T .
(a) ( ) sin(10 )
c
x t t = , [ ] sin( 4) = x n n
(b) ( ) sin(10 ) (10 )
c
x t t t = , [ ] sin( 2) ( 2) = x n n n
(c) ( ) sin(20 ) cos(40 )
c
x t t t = , [ ] sin( 5) cos(2 5) = x n n n

5. Let ( )
c
h t denote the impulse response of an LTI continuous-time filter and [ ] h n the
impulse response of an LTI discrete-time filter.
(a) If ( ) ( )
t
c
h t e u t
= , determine the frequency response of the continuous-time filter

and sketch its magnitude.
(b) If [ ] ( )
c
h n Th nT = with ( )
c
h t as in Part (a) and
1
10
T = , determine the frequency
response of the discrete-time filter and sketch its magnitude.

6. A complex-valued continuous-time signal ( )
c
x t has the Fourier transform shown below
where
2 1
= . This signal is sampled to produce the sequence [ ] ( )
c
x n x nT = .
0
1
2

( )
c
X j

(a) Sketch the Fourier transform ( )
j
X e of the sequence [ ] x n for
2
= T .
(b) What is the lowest sampling frequency that can be used without incurring any
aliasing distortion, i.e., so that ( )
c
x t can be recovered from [ ] x n ?
(c) Draw the block diagram of a system that can be used to recover ( )
c
x t from [ ] x n
if the sampling rate is greater than or equal to the rate determined in Part (b).
Assume (complex) ideal filters are available.
YH Leung (2005, 2006, 2007)
7. A simple model of a multipath communication channel is shown below where
0 1 < < and 0
d
. Assume that the received signal ( )
c
x t is sampled with a
sampling period T to obtain the sequence [ ] ( )
c
x n x nT = , and that the continuous-time
transmitted signal ( )
c
s t is bandlimited such that ( ) 0 =
c
S j for T .
Delay
d
( )
c
s t ( ) ( ) ( )
c c c d
x t s t s t =

(a) Determine the Fourier transform of ( )
c
x t and [ ] x n in terms of ( )
c
S j .
(b) We want to simulate the multipath system with a discrete-time system by
choosing ( )
j
H e as shown below so that the output [ ] ( )
c
r n x nT = when the
input is [ ] ( )
c
s n s nT = . Determine ( )
j
H e in terms of T and
d
.
( )
j
H e
[ ] ( )
c
s n s nT = [ ] ( ) =
c
r n x nT

Hint. See Eq. (10).
(c) Determine the impulse response [ ] h n in the above figure when (i) =
d
T , and
(ii) 2 =
d
T . Comment on your results.

8. The spectrum of a bandpass continuous-time signal ( )
c
x t is shown below. Determine
the lowest sampling rate that can be used such that the sampled signal sequence does
not suffer from aliasing. Sketch the spectrum of the sampled signal.
f (kHz)
0 6 8
|X
c
(f)|

9. Suppose an ADC has reference voltages V where 5 V V = . For the input signal

1 2
( ) 0.8cos sin
c
x t t t =
where
1 2
, determine the number of bits B that are required so that the signal-
to-quantisation-noise ratio is greater than 50 dB.

YH Leung (2005, 2006, 2007)
Answers

1. (a)
o

(b)
o

(c) 2
o

(d) 3
o

3. 250 or 2250

4. (a) 1 40, 9 40
(b) 1 20 and unique
(c) 1 100, 11 100

6. (b)
s

7. (a)
( )
( ) ( ) 1

=
d
j
c c
X j S j e
2
( )
2 2
1
( ) ( ) ( )

= =
=

k
d
T T
j
j
k k
c c
T T T T
k k
X e S j j S j j e
T T

(b) 1

d
j T
e
(c) (i) [ ] [ ] [ 1] = h n n n ; (ii)
( )
1
2
1
2
sin ( )
[ ] [ ]
( )
n
h n n
n

8. 4 kHz

9. 10 bits

YH Leung (2005, 2006, 2007)
Solutions

1. (OWN Problem 7.4)
Nyquist rate equals twice highest frequency of a signal, i.e., slowest sampling rate for
no aliasing. Let the derived signal be denoted by ( )
c
y t .
(a) ( ) ( ) ( ) ( ) 1

l
= =
l
l
j j
c c c c
Y j X j X j e X j e . Therefore, bandwidth of
( )
c
y t same as ( ) x t and so its Nyquist rate is also
o
.
(b) ( ) ( ) =
c c
Y j j X j . Therefore, bandwidth of ( )
c
y t same as ( )
c
x t and so its
Nyquist rate is also
o
.
(c)
1
2
( ) ( ) ( )
=
c c c
Y j X j X j . Therefore, bandwidth of ( )
c
y t is twice that of
( )
c
x t and its Nyquist rate is 2
o
.
(d) ( ) ( )
1 1
2 2
( ) ( ) ( ) =
c c o c o
Y j X j X j . Therefore, highest frequency of
( )
c
y t is at
3 1
2 2
o o o
= and so its Nyquist rate is 3
o
.

As can be determined from a table of Fourier transforms, the impulse response of the
interpolating filter ( )
i
H j is given by
sin
( )
c
i
t
h t T
t
=
Also, as can be readily seen
( ) ( ) ( )
s c
n
x t x nT t nT
=
=

( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( ) ( )
sin ( )
( )
( )
r s i s i
c i
n
c i
n
c i
n
c
c
n
x t x t h t x h t d
x nT nT h t d
x nT nT h t d
x nT h t nT
t nT
x nT T
t nT

=
= =
' '
1 1
1 1
=
! !
1 1
1 1 + +
=
=

which equals the stated result.
YH Leung (2005, 2006, 2007)
The simplified expression follows by noting that when 2
c s
= , 2 2
c
T T = =
and 2 2 2 1
c s
T T = = = .

Now, when = t mT , m , we have
sin ( )
sin ( )
( ) ( ) ( )
( ) ( )

= =

= =

T
r c c
n n T
mT nT
m n
x mT x kT x nT
mT nT m n

But
1,
sin ( )
0, ( )
' =
1
1
=
!
1

1+
n m
m n
n m m n

( ) ( ) =
r c
x mT x mT

Above result is a special case of Nyquists first method for zero ISI.

3. (OSB Problem 4.2)
First observe that
cos cos 2 ,
4 4
n k n k

1 l 1

= l

( ) ( ) l
l

Now ( ) [ ] ( ) cos
c o
x n x nT nT = =
Therefore 2 ,
1000 4
o
n
k n k

1

=

( )

or 250 2000
o
k =

Alternatively, since the effect of sampling a signal is to replicate its spectrum at integer
multiples of the sampling frequency, we consider the sinusoid with frequency
1
where
after sampling, its spectral line is replicated at
o
. That is,
1 s o
k = .

Now, assuming no aliasing,
250
1000 4
o
o o
n
nT n

= = =
Therefore
1
250 2000
o s
k k = =

YH Leung (2005, 2006, 2007)
4. (OSB Problem 4.11 and 4.4)
(a) Similar to Q3, we consider
10 2 ,
4
nT k n k

1

=

( )

which gives
1
,
40 5
k
T k =

(b) ( )
c
X j is a rectangular function centred at 0 = . Therefore, for a given
sampling frequency, it is impossible to find another sampling frequency such that
the sampled spectra are identical. Hence, we solve

1
10
2 20
n
nT T
= =

(c) With reference to Part (a), we solve, simultaneously

1
20 2
5
nT k n

1

=

( )
and
2
2
40 2
5
nT k n

1

=

( )

i.e.
1
1
100 10
k
T = and
2
1
100 20
k
T =
Therefore, one solution is when
1 2
0 k k = = , i.e., 1 100 T = . Another solution is
when
1
1 k = and
2
2 k = such that 1 100 1 10 11 100 T = = .

(a) From a Fourier transform table, ( ) 1 (1 ) =
c
H j j

YH Leung (2005, 2006, 2007)
(b)
( )
10 1 10
1 1
10 10
[ ] [ ] [ ]
n
n
h n e u n e u n

= = . Therefore,
1 10
( ) 0.1 (1 )
j j
H e e e

=

2 2 10 2 31.42 rad/s = =
s
. The similarity and difference between ( )
c
H j
and ( )
j
H e for 0 31.42 is to be noted.

(a) Sampling frequency is
2 2
2 2 2
s
T = = = . Also,
2 2
T = = .
0

( )
j
X e

(b) Sampling shifts the spectrum by integers multiples of the sampling frequency and
replicates the spectrum there. Therefore, there is no spectral overlap if
s
.

(c)
Sequence to
impulse train
conversion
1
Complex bandpass
filter
[ ] x n ( )
c
x t

YH Leung (2005, 2006, 2007)
(a) Taking Fourier transform of ( )
c
x t , we get
( ) ( ) 1
d
j
c c
X j S j e

l
=
l
l

Thus ( )
c
X j is also bandlimited to T < . The Fourier transform of [ ] x n is given
by Eq. (7) of the Lecture Notes, i.e.,
2
2
( )
2 2
1
( ) ( )
1
( ) ( )

=

= =
=
=

d
T T
j
c
T T
k
j k
c c
T T T T
k k
X e X j jk
T
S j jk S j jk e
T T

(b) From Eq. (7) of the Lecture Notes, the Fourier transform of [ ] s n is given by
2
1
( ) ( )
=
=
j
c
T T
k
S e S j jk
T

Therefore, system can be simulated with the following discrete-time transfer function
( )
( ) 1
( )
d
j
j T j
j
X e
H e e
S e

= =
where, in the last equality, the fact that ( )
c
S j and ( )
c
X j are both bandlimited to
below half the sampling frequency is invoked.

(c) Impulse response is given by
( )
2 2
1 1
[ ] ( ) 1
2 2
d
j T j j n j n
h n H e e d e e d

= =

(i)
d
T =
( ) ( )
( 1)
2
1 1
[ ] 1
2 2
sin sin ( 1)
( 1)
[ ] [ 1]
j j n j n j n
h n e e d e e d
n n
n n
n n
= =
=

Alternatively, we can get [ ] h n directly from Tables and properties of DTFT.
(ii) 2
d
T =
( )
( )
1
2
( )
2
2
1
2
1
2
1 1
[ ] 1
2 2
sin ( )
[ ]
( )
j n
j j n j n
h n e e d e e d
n
n
n
= =

which is non-causal, i.e., discrete-time simulation system is not realisable.
YH Leung (2005, 2006, 2007)
8. In bandpass sampling, sampled signal has no aliasing if sampling rate
s
f satisfies
2 2
1
c B c B
s
f f f f
f
m m

and 2
B s
f f <
where 7 kHz
c
f = , 2 kHz
B
f = , and m is an integer

m
2 16
1 1
c B
f f
m m
=

2 12
c B
f f
m m
=
1 8 12
2 5.33 6
3 4 4

To find lowest sampling rate, we want m as large as possible. Above table shows largest
possible m is 3. Corresponding lowest possible sampling rate is 4 kHz.
f (kHz)
0 6 8
|X
s
(f)|

9. Since the sinusoids have different frequencies,
2 2
2
0.8 1
0.82
2 2
x
= =
Solving
( )
SNR 6.0206 4.7712 20 log
x
V
B
=
we get
( ) ( )
5
0.82
SNR 4.7712 20 log 50 4.7712 20 log
6.0206 6.0206
9.9774
10
x
V
B

l l

l l
= = l l
l l
l
l l
l =
l
=

YH Leung (2005, 2006, 2007)
Problems

1. Let ( )
c
x t be a signal with Nyquist rate
o
. Determine the Nyquist rate for each of the
following signals.
(a) ( ) ( 1)
c c
x t x t
(b) ( )
d
c
dt
x t
(c)
2
( )
c
x t
(d) ( ) cos
c o
x t t

2. Consider the system shown below where an ideal sampler is followed by an ideal
lowpass filter to reconstruct ( )
c
x t from its samples ( )
s
x t . From the sampling theorem,
we know that if 2 =
s
T is greater than twice the highest frequency present in ( )
c
x t
and 2 =
c s
, then the reconstructed signal ( )
r
x t will exactly equal ( )
c
x t . If this
condition on the bandwidth of ( )
c
x t is violated, then ( )
r
x t will not equal ( )
c
x t . We
seek to show in this problem that if 2 =
c s
, then for any choice of T , ( )
r
x t and
( )
c
x t will always be equal at the sampling instants, that is,
( ) ( ) , 0, 1, 2,
r c
x nT x nT n = =
x
c
(t) x
s
(t) x
r
(t)
2T T -T 0
t
( )
T
s t

( )
i
H j
c

c
T

To obtain the result, first show

( ) sin ( )
( ) ( )
( )
c c
r c
c
n
t nT
x t x nT T
t nT

which reduces to the following expression when 2 =
c s

( ) sin ( )
( ) ( )
( )
T
r c
n T
t nT
x t x nT
t nT

Then consider the values of for which ( ) sin 0 = .

YH Leung (2005, 2006, 2007)
3. The sequence
[ ] cos ,
4
1

= < <

( )
x n n n
was obtained by sampling a continuous-time signal
( ) ( ) cos ,
c o
x t t t = < <
at a sampling rate of 1000 samples/s. What are two possible positive values of
o
that
could have resulted in the sequence [ ] x n ?

4. The following continuous-time signals ( )
c
x t are sampled. Specify a choice for the
sampling period T that will result in the discrete-time sequences shown. In each case,
is your choice of T unique? If not, give another possible choice of T .
(a) ( ) sin(10 )
c
x t t = , [ ] sin( 4) = x n n
(b) ( ) sin(10 ) (10 )
c
x t t t = , [ ] sin( 2) ( 2) = x n n n
(c) ( ) sin(20 ) cos(40 )
c
x t t t = , [ ] sin( 5) cos(2 5) = x n n n

5. Let ( )
c
h t denote the impulse response of an LTI continuous-time filter and [ ] h n the
impulse response of an LTI discrete-time filter.
(a) If ( ) ( )
t
c
h t e u t
= , determine the frequency response of the continuous-time filter

and sketch its magnitude.
(b) If [ ] ( )
c
h n Th nT = with ( )
c
h t as in Part (a) and
1
10
T = , determine the frequency
response of the discrete-time filter and sketch its magnitude.

6. A complex-valued continuous-time signal ( )
c
x t has the Fourier transform shown below
where
2 1
= . This signal is sampled to produce the sequence [ ] ( )
c
x n x nT = .
0
1
2

( )
c
X j

(a) Sketch the Fourier transform ( )
j
X e of the sequence [ ] x n for
2
= T .
(b) What is the lowest sampling frequency that can be used without incurring any
aliasing distortion, i.e., so that ( )
c
x t can be recovered from [ ] x n ?
(c) Draw the block diagram of a system that can be used to recover ( )
c
x t from [ ] x n
if the sampling rate is greater than or equal to the rate determined in Part (b).
Assume (complex) ideal filters are available.
YH Leung (2005, 2006, 2007)
7. A simple model of a multipath communication channel is shown below where
0 1 < < and 0
d
. Assume that the received signal ( )
c
x t is sampled with a
sampling period T to obtain the sequence [ ] ( )
c
x n x nT = , and that the continuous-time
transmitted signal ( )
c
s t is bandlimited such that ( ) 0 =
c
S j for T .
Delay
d
( )
c
s t ( ) ( ) ( )
c c c d
x t s t s t =

(a) Determine the Fourier transform of ( )
c
x t and [ ] x n in terms of ( )
c
S j .
(b) We want to simulate the multipath system with a discrete-time system by
choosing ( )
j
H e as shown below so that the output [ ] ( )
c
r n x nT = when the
input is [ ] ( )
c
s n s nT = . Determine ( )
j
H e in terms of T and
d
.
( )
j
H e
[ ] ( )
c
s n s nT = [ ] ( ) =
c
r n x nT

Hint. See Eq. (10).
(c) Determine the impulse response [ ] h n in the above figure when (i) =
d
T , and
(ii) 2 =
d
T . Comment on your results.

8. The spectrum of a bandpass continuous-time signal ( )
c
x t is shown below. Determine
the lowest sampling rate that can be used such that the sampled signal sequence does
not suffer from aliasing. Sketch the spectrum of the sampled signal.
f (kHz)
0 6 8
|X
c
(f)|

9. Suppose an ADC has reference voltages V where 5 V V = . For the input signal

1 2
( ) 0.8cos sin
c
x t t t =
where
1 2
, determine the number of bits B that are required so that the signal-
to-quantisation-noise ratio is greater than 50 dB.

YH Leung (2005, 2006, 2007)
Answers

1. (a)
o

(b)
o

(c) 2
o

(d) 3
o

3. 250 or 2250

4. (a) 1 40, 9 40
(b) 1 20 and unique
(c) 1 100, 11 100

6. (b)
s

7. (a)
( )
( ) ( ) 1

=
d
j
c c
X j S j e
2
( )
2 2
1
( ) ( ) ( )

= =
=

k
d
T T
j
j
k k
c c
T T T T
k k
X e S j j S j j e
T T

(b) 1

d
j T
e
(c) (i) [ ] [ ] [ 1] = h n n n ; (ii)
( )
1
2
1
2
sin ( )
[ ] [ ]
( )
n
h n n
n

8. 4 kHz

9. 10 bits

YH Leung (2005, 2006, 2007)
Solutions

Nyquist rate equals twice highest frequency of a signal, i.e., slowest sampling rate for
no aliasing. Let the derived signal be denoted by ( )
c
y t .
(a) ( ) ( ) ( ) ( ) 1

l
= =
l
l
j j
c c c c
Y j X j X j e X j e . Therefore, bandwidth of
( )
c
y t same as ( ) x t and so its Nyquist rate is also
o
.
(b) ( ) ( ) =
c c
Y j j X j . Therefore, bandwidth of ( )
c
y t same as ( )
c
x t and so its
Nyquist rate is also
o
.
(c)
1
2
( ) ( ) ( )
=
c c c
Y j X j X j . Therefore, bandwidth of ( )
c
y t is twice that of
( )
c
x t and its Nyquist rate is 2
o
.
(d) ( ) ( )
1 1
2 2
( ) ( ) ( ) =
c c o c o
Y j X j X j . Therefore, highest frequency of
( )
c
y t is at
3 1
2 2
o o o
= and so its Nyquist rate is 3
o
.

As can be determined from a table of Fourier transforms, the impulse response of the
interpolating filter ( )
i
H j is given by
sin
( )
c
i
t
h t T
t
=
Also, as can be readily seen
( ) ( ) ( )
s c
n
x t x nT t nT
=
=

( ) ( ) ( ) ( ) ( )
( ) ( ) ( )
( ) ( ) ( )
( ) ( )
sin ( )
( )
( )
r s i s i
c i
n
c i
n
c i
n
c
c
n
x t x t h t x h t d
x nT nT h t d
x nT nT h t d
x nT h t nT
t nT
x nT T
t nT

=
= =
' '
1 1
1 1
=
! !
1 1
1 1 + +
=
=

which equals the stated result.
YH Leung (2005, 2006, 2007)
The simplified expression follows by noting that when 2
c s
= , 2 2
c
T T = =
and 2 2 2 1
c s
T T = = = .

Now, when = t mT , m , we have
sin ( )
sin ( )
( ) ( ) ( )
( ) ( )

= =

= =

T
r c c
n n T
mT nT
m n
x mT x kT x nT
mT nT m n

But
1,
sin ( )
0, ( )
' =
1
1
=
!
1

1+
n m
m n
n m m n

( ) ( ) =
r c
x mT x mT

Above result is a special case of Nyquists first method for zero ISI.

First observe that
cos cos 2 ,
4 4
n k n k

1 l 1

= l

( ) ( ) l
l

Now ( ) [ ] ( ) cos
c o
x n x nT nT = =
Therefore 2 ,
1000 4
o
n
k n k

1

=

( )

or 250 2000
o
k =

Alternatively, since the effect of sampling a signal is to replicate its spectrum at integer
multiples of the sampling frequency, we consider the sinusoid with frequency
1
where
after sampling, its spectral line is replicated at
o
. That is,
1 s o
k = .

Now, assuming no aliasing,
250
1000 4
o
o o
n
nT n

= = =
Therefore
1
250 2000
o s
k k = =

YH Leung (2005, 2006, 2007)
4. (OSB Problem 4.11 and 4.4)
(a) Similar to Q3, we consider
10 2 ,
4
nT k n k

1

=

( )

which gives
1
,
40 5
k
T k =

(b) ( )
c
X j is a rectangular function centred at 0 = . Therefore, for a given
sampling frequency, it is impossible to find another sampling frequency such that
the sampled spectra are identical. Hence, we solve

1
10
2 20
n
nT T
= =

(c) With reference to Part (a), we solve, simultaneously

1
20 2
5
nT k n

1

=

( )
and
2
2
40 2
5
nT k n

1

=

( )

i.e.
1
1
100 10
k
T = and
2
1
100 20
k
T =
Therefore, one solution is when
1 2
0 k k = = , i.e., 1 100 T = . Another solution is
when
1
1 k = and
2
2 k = such that 1 100 1 10 11 100 T = = .

(a) From a Fourier transform table, ( ) 1 (1 ) =
c
H j j

YH Leung (2005, 2006, 2007)
(b)
( )
10 1 10
1 1
10 10
[ ] [ ] [ ]
n
n
h n e u n e u n

= = . Therefore,
1 10
( ) 0.1 (1 )
j j
H e e e

=

2 2 10 2 31.42 rad/s = =
s
. The similarity and difference between ( )
c
H j
and ( )
j
H e for 0 31.42 is to be noted.

(a) Sampling frequency is
2 2
2 2 2
s
T = = = . Also,
2 2
T = = .
0

( )
j
X e

(b) Sampling shifts the spectrum by integers multiples of the sampling frequency and
replicates the spectrum there. Therefore, there is no spectral overlap if
s
.

(c)
Sequence to
impulse train
conversion
1
Complex bandpass
filter
[ ] x n ( )
c
x t

YH Leung (2005, 2006, 2007)
(a) Taking Fourier transform of ( )
c
x t , we get
( ) ( ) 1
d
j
c c
X j S j e

l
=
l
l

Thus ( )
c
X j is also bandlimited to T < . The Fourier transform of [ ] x n is given
by Eq. (7) of the Lecture Notes, i.e.,
2
2
( )
2 2
1
( ) ( )
1
( ) ( )

=

= =
=
=

d
T T
j
c
T T
k
j k
c c
T T T T
k k
X e X j jk
T
S j jk S j jk e
T T

(b) From Eq. (7) of the Lecture Notes, the Fourier transform of [ ] s n is given by
2
1
( ) ( )
=
=
j
c
T T
k
S e S j jk
T

Therefore, system can be simulated with the following discrete-time transfer function
( )
( ) 1
( )
d
j
j T j
j
X e
H e e
S e

= =
where, in the last equality, the fact that ( )
c
S j and ( )
c
X j are both bandlimited to
below half the sampling frequency is invoked.

(c) Impulse response is given by
( )
2 2
1 1
[ ] ( ) 1
2 2
d
j T j j n j n
h n H e e d e e d

= =

(i)
d
T =
( ) ( )
( 1)
2
1 1
[ ] 1
2 2
sin sin ( 1)
( 1)
[ ] [ 1]
j j n j n j n
h n e e d e e d
n n
n n
n n
= =
=

Alternatively, we can get [ ] h n directly from Tables and properties of DTFT.
(ii) 2
d
T =
( )
( )
1
2
( )
2
2
1
2
1
2
1 1
[ ] 1
2 2
sin ( )
[ ]
( )
j n
j j n j n
h n e e d e e d
n
n
n
= =

which is non-causal, i.e., discrete-time simulation system is not realisable.
YH Leung (2005, 2006, 2007)
8. In bandpass sampling, sampled signal has no aliasing if sampling rate
s
f satisfies
2 2
1
c B c B
s
f f f f
f
m m

and 2
B s
f f <
where 7 kHz
c
f = , 2 kHz
B
f = , and m is an integer

m
2 16
1 1
c B
f f
m m
=

2 12
c B
f f
m m
=
1 8 12
2 5.33 6
3 4 4

To find lowest sampling rate, we want m as large as possible. Above table shows largest
possible m is 3. Corresponding lowest possible sampling rate is 4 kHz.
f (kHz)
0 6 8
|X
s
(f)|

9. Since the sinusoids have different frequencies,
2 2
2
0.8 1
0.82
2 2
x
= =
Solving
( )
SNR 6.0206 4.7712 20 log
x
V
B
=
we get
( ) ( )
5
0.82
SNR 4.7712 20 log 50 4.7712 20 log
6.0206 6.0206
9.9774
10
x
V
B

l l

l l
= = l l
l l
l
l l
l =
l
=

YH Leung (2005, 2006)
Problems

1. For each of the following transfer functions, state whether or not it is a minimum-phase
system. Justify your answers.
(a)
1 1
1
2
1 1
1 1
3 3
(1 2 )(1 )
( )
(1 )(1 )
z z
H z
z z

+
=
+

(b)
1 1
1 1
4 4
1 1
2 2
3 3
(1 )(1 )
( )
(1 )(1 )
z z
H z
z z

+
=
+

(c)
1
1
3
1 1
2 2
1
( )
(1 )(1 )
j j
z
H z
z z
=
+

(d)
1 1
1
3
1 1
2 2
(1 )
( )
(1 )(1 )
j j
z z
H z
z z

=
+

2. The impulse responses for several different LTI systems are shown below. Find the
group delay associated with each system.
(a) { }
5
1
[ ] 0, 1, 0, 0, 0, 1, 0
n
h n
=
=
(b) { }
4
2
[ ] 0, 1, 1, 2, 2, 1, 1
n
h n
=
=
(c) { }
5
1
[ ] 0, 0, 1, 0, 1, 0, 0
n
h n
=
=
(d) { }
8
2
[ ] 0, 1, 0, 3, 1, 2, 1, 3, 0, 1, 0
n
h n
=
=
(e) { }
6
0
[ ] 0, 1, 2, 0, 2, 1, 0
n
h n
=
=
(f) { }
8
1
[ ] 0, 1, 1, 1, 1, 1, 1, 1, 1, 0
n
h n
=
=

YH Leung (2005, 2006)
3. The figures below show just the zero locations for several different system functions.
For each figure, state whether the system function could be a generalised linear-phase
system implemented by an LCCDE with real coefficients.

(a) (b) (c)

4. Consider a causal LTI system with transfer function

1 1
1
1
( )
1
a z
H z
az

where a is real
(a) For what range of values of a is the system stable?
(b) For
1
2
a = , where are the pole and zero, and what is the ROC?
(c) Find, in terms of a, the impulse response [ ] h n of the system.
(d) Show that the system is all-pass, i.e., the magnitude of the frequency response is a
constant. Specify, in terms of a, the value of the constant.

5. Show that the energy of the output signal of an all-pass system equals the energy of its
input signal, i.e.

2 2
[ ] [ ]
n n
y n x n

= =
=

YH Leung (2005, 2006)
6. Consider an LTI system with input [ ] x n and output [ ] y n . When the input is

( )
( )
sin 0.4
[ ] 5 10cos 0.5
n
x n n
n
= +
the corresponding output is

( ) sin 0.3 ( 10)
[ ] 10
( 10)
n
y n
n

Determine the frequency and impulse responses for the system.
Hint. Sketch ( )
j
X e

and ( )
j
Y e

.

7. The system function
II
( ) H z represents a Type II FIR generalised linear phase filter
with impulse response
II
[ ] h n . This filter is cascaded with an LTI system whose system
function is
1
(1 ) z
to produce a third system with system function ( ) H z and impulse

response [ ] h n . Prove that the overall filter is a generalised linear phase filter, and
determine what type of linear phase filter it is.

YH Leung (2005, 2006)
Answers

1. (a) Not minimum phase
(b) Is minimum phase
(c) Is minimum phase
(d) Not minimum phase

2. (a) 2; (b) 1.5; (c) 2; (d) 3; (e) 3; (f) 3.5

3. (a) Yes; (b) No; (c) Yes

4. (a) 1 a <
(b) Pole is at
1
2
z a = = and zero is at 1 2 z a = = . ROC is
1
2
z a > = .
(c)
{ }
2
1
[ ] [ ] [ 1]
n
h n a u n u n
a
=
(d) ( ) 1
j
H e a
=

6.
( ) sin 0.3 ( 10)
[ ] 2
( 10)
n
h n
n

7. Type III FIR filter

YH Leung (2005, 2006)
Solutions

A minimum-phase system is a causal stable system with all its poles and zeros inside
the unit circle.
(a) Not minimum phase since has a zero at 2 z =
(b) Is minimum phase
(c) Is minimum phase
(d) Not minimum phase since has a zero at z =

Due to the symmetry of the impulse responses, all systems have generalised linear
phase. The group delay corresponds to the point of symmetry in the impulse response
graph. Thus
(a) { }
5
1
[ ] 0, 1, 0, 0, 0, 1, 0
n
h n
=
= grd ( ) 2
j
H e

=

(b) { }
4
2
[ ] 0, 1, 1, 2, 2, 1, 1
n
h n
=
= grd ( ) 1.5
j
H e

=

(c) { }
5
1
[ ] 0, 0, 1, 0, 1, 0, 0
n
h n
=
= grd ( ) 2
j
H e

=

(d) { }
8
2
[ ] 0, 1, 0, 3, 1, 2, 1, 3, 0, 1, 0
n
h n
=
= grd ( ) 3
j
H e

=

(e) { }
6
0
[ ] 0, 1, 2, 0, 2, 1, 0
n
h n
=
= grd ( ) 3
j
H e

=

(f) { }
8
1
[ ] 0, 1, 1, 1, 1, 1, 1, 1, 1, 0
n
h n
=
= grd ( ) 3.5
j
H e

=

(a) Yes. A zero is accompanied by its complex conjugate and their reciprocals. Zero-
plot could correspond to a Type I linear phase FIR filter.
(b) No, since the system function does not include the complex conjugates of the
zeros.
(c) Yes. All zeros are real and the system function includes the reciprocals of the
zeros. Zero-plot could correspond to a Type II linear phase FIR filter.
YH Leung (2005, 2006)
4. (OSB Problem 5.22bcde)
(a) System is causal, so ROC is z a > . Now, for stability, the ROC must include
the unit circle. Therefore, ( ) H z is stable for 1 a < .

(b) Pole is at
1
2
z a = = and zero is at 1 2 z a = = . ROC is
1
2
z a > = .

(c)
1 1
1 1
1
( )
1 1
a z
H z
az az

=

{ }
1 1
2
1
[ ] [ ] [ 1] [ ] [ 1]
n n n
h n a u n a a u n a u n u n
a

= =

(d)
1 1
2
1 1
1 1
( ) ( ) ( )
j j
j j
a e a e j j j
ae ae
H e H e H e

= =

2 1 2
2 1 2 2 1
1 2 cos 1 1 2 cos 1
( )
1 2 cos 1 2 cos
j
a a a a
H e
a
a a a a a

+ +
= = =
+ +

5. (OSB Problem 5.24b)

2
2
2
2 2
2
1
[ ] ( ) (by Parseval's theorem)
2
1
( ) ( )
2
1
( ) ( ( ) 1 since [ ] is allpass)
2
[ ] (by Parseval's theorem)
j
n
j j
j j
n
y n Y e d
H e X e d
X e d H e h n
x n
=
=
=
= =
=

YH Leung (2005, 2006)
The DTFTs of [ ] x n and [ ] y n are shown below.
0
0
0.5 0.4 0.4 0.5

0.3 0.3
( )
j
X e

( )
j
Y e

5
10 10
10
10
j
e

Therefore, the filter must have following transfer function and impulse response
0

0.3 0.3
( )
j
H e

10
2
j
e

( ) sin 0.3 ( 10)
[ ] 2
( 10)
n
h n
n

The frequency responses of the two component filters can be written as

2
II II
( ) ( )
j j jM
H e A e e

=

( ) ( ) ( )
2 2 2 2 2 2
2 2
( ) 1 2 sin 2sin

+
= = = =
j j j j j j j j
G e e e e e j e e
where
II
( )
j
A e

is a real, even, periodic function of and M is an odd integer. The
transfer function of the cascaded system is given, therefore, by

( )
( )
2 2 2
II II
2
2 2
II
2
( ) ( ) ( ) ( ) 2sin
2 ( ) sin

+
+
= =
=
j j j j jM j j
j j M j
H e H e G e A e e e
A e e

where 1 M M
= + .
Now,
II
( )
j
A e

is an even function of while ( )
2
sin

is an odd function of .
Therefore, ( )
II
2
( ) sin
j
A e

is an odd function of . Also, M is even since M is odd.
The cascaded system is thus a Type III linear phase FIR filter.
YH Leung (2005, 2006, 2007, 2008)
Problems

1. Verify Eq. (29) and Fig. 9.

2. (a) We wish to design a Type I linear phase FIR filter with 6 M = using the window
method. Suppose the desired ideal discrete-time filter is a lowpass filter with
cut-off frequency 0.4
c
= , and the window function is the Hamming window.
Determine the impulse response of the filter.
(b) Repeat Part (a) but for a Type II filter with 5 M = .
(c) Repeat Part (a) but for a Type III filter ( 6 M = ).

3. We wish to design an FIR lowpass filter satisfying the specifications
0.95 ( ) 1.05,
j
H e

< < 0 0.25
0 ( ) 0.1,
j
H e

< < 0.35
by applying a window [ ] w n to the impulse response [ ]
d
h n for the ideal discrete-time
lowpass filter with cutoff 0.3 =
c
. Which of the windows listed in Table 2 of the
Lecture Notes can be used to meet this specification? For each window that you claim
will satisfy this specification, give the minimum length 1 M required for the filter.

4. We wish to design an FIR lowpass filter satisfying the specifications
0.98 ( ) 1.02,
j
H e

< < 0 0.63
( ) 0.15,
j
H e

< 0.65
by applying a Kaiser window to the impulse response [ ]
d
h n of the ideal lowpass filter
with cutoff 0.64
c
= . Find the necessary Kaiser window parameters and M .

YH Leung (2005, 2006, 2007, 2008)
5. We wish to use the Kaiser window method to design a discrete-time FIR filter with
generalised linear phase that meets specifications of the following form:
( ) 0.01,
<
j
H e 0 0.25
0.95 ( ) 1.05,
< <
j
H e 0.35 0.6
( ) 0.01,
<
j
H e 0.65
(a) Determine the minimum length 1 M of the impulse response and the Kaiser
window parameter for the filter to meet the preceding specifications, assuming
the transfer function of the filter is to have the form
2
( ) ( )

=
j j jM
H e A e e
where ( )
j
A e is real and even.
(b) What is the delay of the filter?
(c) Determine the ideal impulse response [ ]
d
h n .

6. We wish to use the Kaiser window method to design an FIR filter with generalised
Type I/II linear phase that meets the following specifications:
0.9 ( ) 1.1,
j
H e

< < 0 0.2
( ) 0.06,
j
H e

< 0.3 0.475
1.9 ( ) 2.1,
j
H e

< < 0.525
This specification is to be met by applying the Kaiser window to the ideal impulse
response associated with the ideal frequency response given by

1, 0 0.25
( ) 0, 0.25 0.5
2, 0.5
j
d
H e

'
1
1
1
1
=
!
1
1
1
1+

(a) What is the maximum value of that can be used to meet this specification?
What is the corresponding value of ? Explain your reasoning.
(b) What is the maximum value of that can be used to meet this specification?
What is the corresponding value of M ? Explain your reasoning.

YH Leung (2005, 2006, 2007, 2008)
7. Let [ ]
d
h n denote the impulse response of an ideal desired system with corresponding
frequency response ( )
j
d
H e

, and let [ ] h n and ( )
j
H e

denote the impulse response and
frequency response, respectively, of an FIR approximation to the ideal system. Assume
that [ ] 0 h n = for 0 n < and n M . We wish to choose the 1 M samples of the
impulse response so as to minimise the mean-square error of the frequency response
defined as

2
2
1
( ) ( )
2
j j
d
H e H e d

(a) Use Parsevals relation to express the error function in terms of the sequences
[ ]
d
h n and [ ] h n
(b) Using the result of Part (a), determine the values of [ ] h n for 0 n M that
minimises
2
.
(c) The FIR filter determined in Part (b) could have been obtained by a windowing
operation. That is, [ ] h n could have been obtained by multiplying the desired
infinite-length sequence [ ]
d
h n by a certain finite-length sequence [ ] w n .
Determine the necessary window [ ] w n such that the optimal impulse response is
[ ] [ ] [ ]
d
h n w n h n = .

8. An ideal discrete-time Hilbert transformer is a system that introduces 90 ( 2
radians) of phase shift for 0 < < and 90 ( 2 radians) of phase shift for
0 < < . The magnitude of the frequency response is constant (unity) for 0 < <
and for 0 < < . Such systems are also called ideal 90-degree phase shifters.
(a) Give an equation for the ideal frequency response ( )
j
d
H e

of an ideal discrete-
time Hilbert transformer that also includes constant (non-zero) group delay. Plot
the phase response of this system for < < .
(b) What type(s) of FIR linear-phase systems (I, II, III, or IV) can be used to
approximate the ideal Hilbert transformer in Part (a)?
(c) Suppose that we wish to use the window method to design a linear-phase
approximation to the ideal Hilbert transformer. Use ( )
j
d
H e

given in Part (a) to
determine the ideal impulse response [ ]
d
h n if the FIR system is to be such that
[ ] 0 h n = for 0 n < and n M .
(d) What is the delay of the system if 21 M = ? Sketch the magnitude response of the
FIR approximation for this case, assuming a rectangular window.
(e) What is the delay of the system if 20 M = ? Sketch the magnitude response of the
FIR approximation for this case, assuming a rectangular window.
YH Leung (2005, 2006, 2007, 2008)
9. Consider the following ideal frequency response for a multiband filter

2
2
, 0 0.3
( ) 0, 0.3 0.6
0.5 , 0.6
j M
j
d
j M
e
H e
e
'
1

1
1
1
=
!
1
1
1

1+

The impulse response [ ]
d
h n is multiplied by a Kaiser window with 48 M = and
3.68 = , resulting in a linear-phase FIR system with impulse response [ ] h n .
(a) What is the delay of the filter?
(b) Determine the ideal impulse response [ ]
d
h n .
(c) Determine the set of approximation error specifications that is satisfied by the FIR
filter, i.e., determine the parameters
1
,
2
,
3
, B , C ,
1 p
,
1 s
,
2 s
, and
2 p
in

1 1
( ) ,
j
B H e B

1
0
p

2
( ) ,
j
H e

1 2 s s

3 3
( ) ,
j
C H e C

2 p

10.
*
Prove Eq. (79). That is, show that in the frequency sampling method, the frequency
response of the designed filter and the desired frequency response satisfy

2 ( ) 2 ( )
1 1
( ) ( ) , 0, ,
k k
M M
j j
d
H e H e k M

= =
= =

11.
*
Derive Eq. (80) of the Lecture Notes. That is, show, in the frequency sampling method
to design FIR filters, the designed impulse response [ ] h n is related to the impulse
response of the ideal filter, [ ]
d
h n , by the time aliasing expression

1
2
[ ( 1)], 0
[ ]
( 1) [ ( 1)],
d
m
m
d
m
h n m M
h n
h n m M
=
'
1
1
=
1
1
1
1
=
!
1
1
1
=
1
1
1+

YH Leung (2005, 2006, 2007, 2008)
12.
*
The ideal magnitude response of a linear phase FIR filter is shown below.

( )
j
d
H e
0
0.2

0.5
1

Design the (Type II) filter using the frequency sampling method with 0 = and
9 M = . List the filter coefficients.

13.
*
It is found that the distortion affecting a digital signal can be modelled by a system
whose impulse response is given by
[0] 1, [1] 1.3, [2] 0.2, [ ] 0, 3 = = = = h h h h n n
Using the frequency sampling method, design an FIR equalising filter which has 5
coefficients. (An equalising filter is expected to remove the effect of the distortion.)

YH Leung (2005, 2006, 2007, 2008)
Answers

2. (a)
6
0
[ ] 0.0050, 0.0290, 0.2331, 0.4, 0.2331, 0.0290, 0.0050
n
h n
=
=
(b)
5
0
[ ] 0, 0.0803, 0.3413, 0.3413, 0.0803, 0
n
h n
=
=
(c)
6
0
[ ] 0.0154, 0.0893, 0.1694, 0, 0.1694, 0.0893, 0.0154
n
h n
=
=

3. Hanning: 1 81 M =
Hamming: 1 81 M =
Blackman: 1 121 M =

4. Passband and stopband are wide enough compared to the transition width for us to say
min 0.02, 0.15 0.02 = =
First-cut design thus proceeds as follows
20 log 33.9794 A = =
0.65 0.63 0.02 = =

0.4
0.5842( 21) 0.07886( 21) 2.6523 A A = =

8
180.9519 181
2.285
A
M
= =

0 0.2 0.4 0.6 0.8 1
0
0.5
1
1.5
f (normalised to half sampling frequency)
M
a
g
n
i
t
u
d
e

R
e
s
p
o
n
s
e

A careful examination of the magnitude response obtained will reveal the first-cut filter
violates the design specification at around 0.63 = . This design violation may be
resolved by increasing M to 182.
YH Leung (2005, 2006, 2007, 2008)
5. (a) Design specification is for a bandpass filter. Bandpass filters involve 4 window
functions as shown below.
11

1 0.5 0 0.5 1
0
0.5
1
1.5
1 0.5 0 0.5 1
0
20
40
60
(x)

Therefore, for each in-band region, there are now 4 interacting window functions.
However, we can safely assume that it is only the two window functions immediately
on either side of an in-band region that have the greatest influence on the ripple in that
in-band region. Accordingly, we proceed as follows.
The narrowest transition region is given by
min (0.35 0.25 ), (0.65 0.6 ) 0.05 = =
which is much smaller than all the in-band widths. Therefore, design equations are
min 0.01, 0.05, 0.05, 0.01 0.01 20 log 40 = = = = A

0.4
0.5842( 21) 0.07886( 21) 3.3953 A A = =

8
2.285
89.1546 90
= =
A
M
0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
1.4
M
a
g
n
i
t
u
d
e

R
e
s
p
o
n
s
e

A careful examination of the frequency response obtained will show the designed filter
violates the design specifications at around 0.657 = . The design violation can be
resolved by increasing M to 91.

11
See also pp. 473 and pp. 482 of Oppenheim and Schafer with Buck.
YH Leung (2005, 2006, 2007, 2008)
(b) In terms of the first-cut design, since filter is linear phase, its delay is 2 M =
90 2 45 = samples.
(c) Choosing the two
c
s to be half-way between the stopband and passband edges,
impulse response of the first-cut design is

sin 0.625 ( 45) sin0.3 ( 45)
[ ] , (assuming Type I)
( 45) ( 45)
d
n n
h n
n n

=

6. For first-cut design
(a) 2.1809 =
(b) 62.5686 63 64 M =

7. (a) Define ( ) ( ) ( )
j j j
d
E e H e H e

=
such that, by Parsevals relation

2
2 2
1
( ) [ ]
2
j
n
E e d e n
=
= =

where [ ] e n is the inverse DTFT of ( ) ( ) ( )
j j j
d
E e H e H e

= , i.e.,

[ ], 0
[ ] [ ] [ ], 0
[ ],
d
d
d
h n n
e n h n h n n M
h n n M
' <
1
1
1
1
=
!
1
1
1
1+

(b) Clearly,
2
is minimised if [ ] [ ] 0
d
h n h n = , i.e., [ ] [ ]
d
h n h n = for 0 n M .
(c) Since [ ] [ ]
d
h n h n = for 0 n M , the window is a length 1 M rectangular
window.

8. (a) ( ) sgn( )
j j
d
H e j e

= or

2
1 2 ( )
( )
j u
j
d
H e e
l

l
l
=
(b) Types III and IV.
(c)

2
1
2 2
( )
2
1 cos( ) ,
[ ]
0,
'
1
1
1
=
!
1
= 1
1+
M
M M
n
d
M
n n
h n
n

(d) 10.5 samples.
(e) 10 samples.

YH Leung (2005, 2006, 2007, 2008)
9. (a) 24 samples.
(b)

sin0.3 ( 24) sin0.6 ( 24)
1
( 24) 2 ( 24)
[ ] [ 24]
n n
d
n n
h n n

=
(c) The tightest set of parameters are:
1
0.25
p
= ,
1
0.35
s
= ,
2
0.55
s
= ,
2
0.65
p
= ,
1
0.0075634 = ,
2
0.011345 = ,
3
0.0037817 = , 1 B = , and
0.5 C = . These parameters are tightest in the sense that, for example, we can
keep
1
at 0.0075634 and increase
2
and
3
; or keep
1 1 s p
at 0.1 but make
2 2
0.1
p s
.

11.
*
From Eqs. (76) and (77), we have
1 1
1 1
1 1
1 1
2 2
0
2 2
0
2 2
0
2 ( ) 2 ( )
1
[ ] ( )
1
1
[ ]
1
1
[ ]
1
1
[ ]
1
k k
M M
k k
M M
k k
M M
k
M M
M
j j n
d
k
M
j n j n
d
k n
M
j n j n
d
n k
j n n j n n
d
h n H e e
M
h n e e
M
h n e e
M
h n e e
M

= =

= =

=
' '
1 1
1 1
=
! !
1 1
1 1 + +
' '
1 1
1 1
=
! !
1 1
1 1 + +

0
M
n k
= =
' '
1 1
1 1
! !
1 1
1 1 + +

But
1
2 ( )
0
1, ( ) ( 1),
0, otherwise
k
M
M
j n n
k
M n n p M p
e

=
' =
1
1
=
!
1
1+

1
2 ( 1)
2
1
2
1
[ ] [ ( 1)] 1
1
[ ( 1)]
[ ( 1)], 0
[ ( 1)]( 1) ,
M
j p M
d
p
j p
d
p
d
p
p
d
p
h n h n p M e M
M
h n p M e
h n p M
h n p M
=
=
=
'
1
1
=
1
1
1
1
=
!
1
1
1
=
1
1
1
+

Finally, making the substitution p m = completes the proof.

12.
*
[ ] 0.036277, 0.038158, 0.033333, 0.196956, 0.344146, h n =

9
0
0.344146, 0.196956, 0.033333, 0.038158, 0.036277
n=

YH Leung (2005, 2006, 2007, 2008)
Solutions

1. (Made up)
Recall Eq. (22). The amplitude part of ( )
j
W e

is given by

( )
( )
amp
sin ( 1) 2
( )
sin 2
j
M
W e

=
which is maximum at 0 = and

( )
( )
( )
( )
0
amp
0 0
sin ( 1) 2 cos ( 1) 2 ( 1) 2
( ) lim lim 1
sin 2 cos 2 1 2
j
M M M
W e M

= = =

First zero-crossing of
amp
( )
j
W e

after 0 = occurs at
( 1) 2
o
M =
i.e.
2
1
o
M
=
Therefore
4
ml
1
2

= =
o
M

thus verifying Eq. (29).

To obtain Fig. 9, we proceed as follows.

( ) ( )
1 1 1 1
amp
2 2 2 2 2
1 1
( ) cos sin
sin 2
sin 2
j M M M
d
W e
d

=
Suppose
sl
is the frequency where the peak side-lobe of
amp
( )
j
W e

occurs.
sl
can be
found by setting the above derivative to zero, that is, by solving

( )
sl 1
sl
2 2
( 1) sin tan

=
M
M
There is, however, no explicit expression for
sl
. It can only be found numerically
using, for example, the steepest descent method. Once
sl
is found, we substitute it into
the expression for
amp
( )
j
W e

to get its peak side-lobe level. Fig. 9 is obtained by
plotting

sl sl
amp amp
0
amp
( ) ( )
1
( )
20log 20log

=
j j
j
W e W e
M
W e

YH Leung (2005, 2006, 2007, 2008)
2. (Made up)
(a) Ideal impulse response and window function are given, respectively, by

3
sin 0.4 ( 3)
[ ]
( 3)
, 0.0624, 0.0935, 0.3027, 0.4, 0.3027, 0.0935, 0.0624,
d
n
n
h n
n
=

and

6
2
6
0
[ ] 0.54 0.46cos
n
n
w n

=
=

6
0
0.08, 0.31, 0.77, 1, 0.77, 0.31, 0.08 [ ]
n
w n
=
=

6
0
[ ] [ ] [ ]
0.0050, 0.0290, 0.2331, 0.4, 0.2331, 0.0290, 0.0050
d
n
h n h n w n
=
=
=

(b) For 5 M = , the corresponding expressions are

2.5
sin 0.4 ( 2.5)
[ ]
( 2.5)
, 0, 0.2018, 0.3742, 0.3742, 0.2018, 0,
d
n
n
h n
n
=

and

5
2
5
0
[ ] 0.54 0.46cos
n
n
w n

=
=

5
0
0.08, 0.3979, 0.9121, 0.9121, 0.3979, 0.08 [ ]
n
w n
=
=

5
0
[ ] [ ] [ ]
0, 0.0803, 0.3413, 0.3413, 0.0803, 0
d
n
h n h n w n
=
=
=

(c) For 6 M = and a Type III response, the corresponding expressions are

( )

2
3
2sin 0.2 ( 3)
[ ]
( 3)
, 0.1919, 0.2879, 0.2199, 0, 0.2199, 0.2879, 0.1919,
d
n
n
h n
n
=

and

6
2
6
0
[ ] 0.54 0.46cos
n
n
w n

=
=

6
0
0.08, 0.31, 0.77, 1, 0.77, 0.31, 0.08 [ ]
n
w n
=
=

6
0
[ ] [ ] [ ]
0.0154, 0.0893, 0.1694, 0, 0.1694, 0.0893, 0.0154
d
n
h n h n w n
=
=
=

YH Leung (2005, 2006, 2007, 2008)
Passband and stopband are wide enough compared to the transition width for us to
assume maximal passband and stopband errors are, respectively,
20log(0.05) 26.0206 = =
p

and 20log(0.1) 20 = =
s

Therefore, a window with a peak approximation error of less than 26 dB is required.
Table 2 indicates the Hanning, Hamming and Blackman windows are suitable.
Now, main-lobe width required is
ml
0.35 0.25 0.1 = = =
s p
. Therefore,
minimum lengths required are
Hanning:
8
0.1
1 1 81 M

= =
Hamming:
8
0.1
1 1 81 M

= =
Blackman:
12
0.1
1 1 121 M

= =
The actual minimum lengths required are, in fact, smaller. They can be found by using a
trial-and-error procedure as discussed in Example 1 of the Lecture Notes.

See Answers.

See Answers.

YH Leung (2005, 2006, 2007, 2008)
We first note that the narrowest transition region is given by
min (0.3 0.2 ), (0.525 0.475 ) 0.05 = =
which is much smaller than the two passband widths ( 0.2 0.2 < < , and 0.525 <
< wrapped to 0.525 < < ), and the stopband width ( 0.3 0.475 < < )
In the stopband, the ripple due to the right window is twice the ripple due to the left
window since the ripples are scaled by the heights of the discontinuity. Accordingly, the
minimum requirements on the Kaiser window are
12

0.1 1
2 2
min 0.1, 0.06, 0.06 , 0.03 = =
(a) 20 log 30.4576 = = A

0.4
0.5842( 21) 0.07886( 21) 2.1809 = = A A

(b)
8
2.285
62.5686 63
= =
A
M
But desired filter is, essentially, a bandstop filter. Therefore, we must increase M to 64
to get a Type I filter.
A careful examination of the magnitude response obtained will show the designed filter
violates the design specifications at around 0.462 = . The design violation can be
tamed by increasing M by a further 6 to 70.
Alternatively, recognising the right window is amplified by 2, we conjecture our rule of
thumb regarding the setting of may be compromised in the stopband. Determining
as follows

0.1 1
2 2
min 0.1, 0.06, 0 0. .06 , 0.027 9 = =
yields 66 = M . The revised design meets the design specifications immediately.
0 0.2 0.4 0.6 0.8 1
0
0.5
1
1.5
2
2.5
M
a
g
n
i
t
u
d
e

R
e
s
p
o
n
s
e

12
Note: With some thought, it should be clear that and are NOT the allowed in-band and transition
region tolerances, but rather, they are the and the Kaiser windows must have in order for the resulting
filter to meet the design specification. It just so happens that in lowpass and highpass designs, and
often equal the tightest in-band and transition region tolerances, respectively.
YH Leung (2005, 2006, 2007, 2008)
See Answers.

(a) A possible expression for the ideal frequency response is
( ) sgn( )
j j
d
H e j e

=
where is a positive constant. Other possible expressions are

2
sgn( )
( )
j
j
d
H e e

=
[ [
2
1 2 ( )
( )
j u
j
d
H e e

=

2
( ) 1 2 ( )
j
j
d
H e u e

=
2
2
( )
j
d
H e

0

(b) Hilbert transformers have a phase jump of at 0 = . Thus only Types III and
IV can be used.

(c) The inverse DTFT of ( )
j
d
H e

is given by

( ) ( )

2 2
2 2
0
0
0
( ) ( )
( ) ( )
0
( ) ( ) 1 1
( ) ( )
1
( )
1
[ ]
2
1
2
1
1 1
2
1 cos( ) ,
0,

l l

l l
l l

= =

=
' '
1 1
1 1
=
! !
1 1
1 1
+ +
=
'
1
1
=
!
=
+

j j
j j
j n j n
d
je je j n j n
n n
j n j n
n n
n
h n e e d e e d
e e
e e
n n
n
1
1

For the windowed FIR filter to be linear phase, it must be anti-symmetric about
2 M . The ideal impulse response is anti-symmetric about n = . Therefore, we
must choose 2 M = .

YH Leung (2005, 2006, 2007, 2008)
(d) If 21 M = , the delay is 2 10.5 samples M = , and since M is odd, we have a
Type IV filter. A Type IV filter is characterised by
0
( ) 0
j
H e = .
0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
1.4
M
a
g
n
i
t
u
d
e

R
e
s
p
o
n
s
e

(e) If 20 M = , the delay is 2 10 samples M = , and since M is even, we have a
Type III filter. A Type III filter is characterised by
0
( ) ( ) 0
j j
H e H e

= = .
0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
1.4
M
a
g
n
i
t
u
d
e

R
e
s
p
o
n
s
e

(a) Delay is 2 24 samples M = .

(b) Ideal filter can be viewed as the sum of an ideal lowpass filter with passband
0 = to 0.3 and an ideal highpass filter with gain 0.5 and passband 0.6 =
to . Also, in view of the phase response of the ideal filter and the fact that M is
even, the actual filter will be Type I. The ideal impulse response is thus given by

sin0.3 ( 24) sin ( 24) sin0.6 ( 24)
1
( 24) 2 ( 24) ( 24)
sin0.3 ( 24) sin0.6 ( 24)
1
( 24) 2 ( 24)
[ ]
[ 24]

=
=
n n n
d
n n n
n n
n n
h n
n

YH Leung (2005, 2006, 2007, 2008)
(c) Clearly, 1 B = and 0.5 C = . We next determine A. Given 3.68 = , we solve

0.4
0.1102( 8.7) , 50
3.68 0.5842( 21) 0.07886( 21) , 21 50
0, 21
A A
A A A
A
'
1
1
1
1
= =
!
1
1
< 1
1+

Now, 21 A since 0 . Suppose 50 A . Evaluating
3.68
0.1102
8.7 A= =
42.09 , we get a contradiction. Therefore, we must have 21 50 A . Solving
13

0.4
3.68 0.5842( 21) 0.07886( 21) A A =
we obtain 42.4256 A= whereupon

8 42.4256 8
2.285 2.285 48
0.09991 0.1

= = =
A
M

and hence
0.1
1
2
0.3 0.25
= =
p

0.1
1
2
0.3 0.35
= =
s

0.1
2
2
0.6 0.55
= =
s

0.1
2
2
0.6 0.65
= =
p

We now attempt to determine
1
,
2
,
3
. As can be seen, the ideal filter is a
bandstop filter where the lower passband has a gain of 1 while the upper passband
has a gain of 0.5. Therefore, for the lower and upper passbands, it follows that

20
1
10 0.0075634
A

= =
and
20
3
0.5 10 0.0037817
A

= =

13
The equation can be solved as follows. Define, for convenience, 21 x A = such that the equation becomes
0.4
3.68 0.5842 0.07886 x x =
After some simple manipulations, we get
0.4
(3.68 0.5842 ) 0.07886 x x =
x can be found by first making an initial guess, say, 20 x = , then substituting this guess into the RHS
of the above equation to get another value for x (as given by the LHS of the above equation). The new value of
x, as given by the LHS of the above equation, is then substituted back into the RHS of the above equation to
get another value for x. We keep on repeating this iterative process until x has converged to within a certain
number of significant digits.
Another method to solve the equation is as follows. Let
5
x y = . The equation then becomes
5 0.4 5 2 5
3.68 0.5842( ) 0.07886 0.5842 0.07886 y y y y = =
We next use a polynomial root-finding program to find the roots of the above equation. These roots are
1,2,3,4,5
0.918864 2.241332, 1.845809, 1.841769 0.957267 y j j =
Clearly, only
3
1.845809 y = is useful since A is real. Thus,
5
3
21 21.425625 x A y = = = .
YH Leung (2005, 2006, 2007, 2008)
In the stopband, the approximation error is governed by the two windows on
either side of it, and the maximum approximation error is given by
1 3
.

20
2 1 3
1.5 10 0.011345
A

= = =
Note above specification are tightest in the sense that, for example, we can keep
1
at 0.0075634 and increase
2
and
3
; or keep
1 1 s p
at 0.1 but make
2 2
0.1
p s
.

10.
*
(Made up)
By definition,
0
( ) [ ]
M
j j n
n
H e h n e

=
=
. Therefore, from Eqs. (77) and (76)

1
1 1
2
0 0
2 2
0 0
1
( ) [ ]
1
1
( )
1
k
M
k k
M M
M M
j n
j j n
n k
M M
j j n
j n
d
n k
H e H k e e
M
H e e e
M
= =
= =
l
l
=
l
l
l
l
l
=
l
l
l

At
1
2
k
M
= , we have

1 1 1 1
1 1 1
2
1 1
2 2 2 2
0 0
2 2 2
0 0
2 ( )
0 0
1
( ) ( )
1
1
( )
1
1
( )
1
1
1
k k k k
M M M M
k k k
M M M
k
n
M M
M M
j j j n j n
d
n k
M M
j j n j n
d
k n
M M
j j k k
d
k n
d
H e H e e e
M
H e e e
M
H e e
M
H
M

= =
= =

= =
l
l
=
l
l
l
l
l
=
l
l
l
l
l
=
l
l
l
=

1
2
0
1, ( 1),
( )
0, otherwise
k
M
M
j
k
M k k p M p
e
=
'
= 1
1
!
1
1+

But , 0, , k k M
= . Therefore, second summation is non-zero only when 0 k k
= ,
or k k
= , leading to

1 1 1
2 2 2 1
( ) ( ) ( 1) ( )
1
k k k
M M M
j j j
d d
H e H e M H e
M

= =

11.
*
(Made up)
See Answers.

YH Leung (2005, 2006, 2007, 2008)
12.
*
(Made up)
Frequency points are taken from 0, 0.2 , 0.4 , 0.6 , 0.8 , , 1.2 , 1.4 , 1.6 , 1.8 =

9 9 9 9
2 2 2 2
9
0.2 0.4 0.4 0.2
1 1
3 3
0
[ ] 1, , , 0, 0, 0, 0, 0, ,
j j j j
k
H k e e e e

=
=
Filter coefficients are given by

2
1
0
9
0
1
[ ] [ ]
1
0.036277, 0.038158, 0.033333, 0.196956, 0.344146,
0.344146, 0.196956, 0.033333, 0.038158, 0.036277
=
=
=
nk
M
M
j
k
n
h n H k e
M

which can be found with the following MATLAB code:

M = 9 ;
H = [1 exp(-j*M/2*0.2*pi) 1/3*exp(-j*M/2*0.4*pi) 0 0 0 0 0 ...
1/3*exp(j*M/2*0.4*pi) exp(j*M/2*0.2*pi)] ;
for n=0:M
k = 0:M ;
h(n+1) = sum(H .* exp(j*2*pi*n*k/(M+1))) / (M+1) ;
end
or M = 9 ;
H = [1 exp(-j*M/2*0.2*pi) 1/3*exp(-j*M/2*0.4*pi) 0 0 0 0 0 ...
1/3*exp(j*M/2*0.4*pi) exp(j*M/2*0.2*pi)] ;
n = 0:M ;
m = n' ;
Txf = exp(j*2*pi*m*n/(M+1)) ;
h = Txf * H.'/(M+1)

YH Leung (2005, 2006, 2007, 2008)
13.
*
(From Zoubir/Pham)
Transfer function of distorting system is given by
2
2
0
( ) [ ] 1 1.3 0.2

=
= =
j j n j j
n
H e h n e e e
Therefore, ideal equaliser has transfer function
2
1 1
( )
( ) 1 1.3 0.2

= =

j
d
j j j
G e
H e e e

Sampling ( )
j
d
G e at 2 5 = k , 0, , 4 = k (or
1
2
2 ( ) 5 = k ), we get
2 5
2 5 4 5
1
[ ] ( ) , 0, , 4
1 1.3 0.2
0.4, 0.3679 0.4017, 0.03060 1.742,
0.03060 1.742, 0.3679 0.4017

= = =

=

j k
d
j k j k
G k G e k
e e
j j
j j

whence

4
2 5
0
1
[ ] [ ] 0.2394, 0.4468, 0.5329, 0.6035, 0.6779
5
=
= =
j nk
k
g n G k e
Alternatively, often we recognised that in equalisation, accepting some delay in the
equalised signal can result in a better recovery of the original signal. As such, we
require
( ) ( )

=
j j j D
d
H e G e e
where D is some positive real number. Suppose 1 = D . Repeating the above design
steps, we get
2 5
2 5
2 5 4 5
[ ] ( ) , 0, , 4
1 1.3 0.2
0.4, 0.4957 0.2257, 0.9991 1.4272,
0.9991 1.4272, 0.4957 0.2257

= = =

=

j k
j k
d
j k j k
e
G k G e k
e e
j j
j j

and

4
2 5
0
1
[ ] [ ] 0.6779, 0.2394, 0.4468, 0.5329, 0.6035
5
=
= =
j nk
k
g n G k e

YH Leung (2005, 2006, 2008)
Problems

1. Derive the following transform relations that arise in the impulse invariance method
(a)
2 1
2 2
1
1 ( )
( )
1 ( )
aT
aT
T e z
s a
e z

l
l

(b)
1
2 2 1 2 2
sin
( ) 1 2 cos ( )
aT
o
o
aT aT
o o
T e T z
s a e T z e z

l
l
l
l

l
l

(c)

1
2 2 1 2 2
1 cos
( ) 1 2 cos ( )
aT
o
aT aT
o o
T e T z
s a
s a e T z e z
l
l
l

l
l

2. (a) Suppose we wish to design a discrete-time filter using the impulse invariance
method with an ideal continuous-time lowpass filter as a prototype. The prototype
filter has a cut-off frequency of 2000 rad/s
c
= , and the impulse invariance
transformation uses 0.2 ms T = . What is the cut-off frequency
c
for the
resulting discrete-time filter?
(b) Repeat Part (a) if the design method is the bilinear transformation method.

3. (a) An ideal discrete-time lowpass filter with cut-off frequency
4
c
= was designed
using the impulse invariance method with 0.1 ms T = . What was the cut-off
frequency
c
for the prototype continuous-time ideal lowpass filter?
(b) An ideal discrete-time highpass filter with cut-off frequency
2
c
= was designed
using the bilinear method with 1 ms T = . What was the cut-off frequency
c
for
the prototype continuous-time ideal highpass filter?

YH Leung (2005, 2006, 2008)
4. Consider a continuous-time system with system function
1
( )
c
H s
s
=
This system is called an integrator since the output ( ) y t is related to the input ( ) x t by
( ) ( )
t
y t x d

Suppose a discrete-time system is obtained by applying the bilinear transformation to
( )
c
H s .
(a) What is the transfer function ( ) H z of the resulting discrete-time system? What is
its impulse response [ ] h n ?
(b) If [ ] x n is the input and [ ] y n is the output of the resulting discrete-time system,
write the LCCDE that is satisfied by the input and output. What problems do you
anticipate in implementing the discrete-time system using this LCCDE?
(c) Obtain an expression for the frequency response ( )
j
H e

of the discrete-time
system. Sketch its magnitude and phase responses for 0 . Compare them
with the magnitude and phase responses of the continuous-time integrator. Under
what conditions could the discrete-time integrator be considered a good
approximation to the continuous-time integrator?
Now, consider the continuous-time differentiator which has transfer function
( )
c
G s s =
Suppose a discrete-time system is obtained by applying the bilinear transformation to
( )
c
G s .
(d) What is the transfer function ( ) G z of the resulting discrete-time system? What is
its impulse response [ ] g n ?
(e) Obtain an expression for the frequency response ( )
j
G e

of the discrete-time
system. Sketch its magnitude and phase responses for 0 . Compare them
with the magnitude and phase responses of the continuous-time differentiator.
Under what conditions could the discrete-time differentiator be considered a
good approximation to the continuous-time differentiator?
(f) The continuous-time integrator and differentiator are exact inverses of one
another. Is the same true of their discrete-time approximations?

YH Leung (2005, 2006, 2008)
5. Impulse invariance and bilinear transformation are two methods for designing discrete-
time filters. Both methods transform a continuous-time transfer function ( )
c
H s into a
discrete-time transfer function ( ) H z . Answer the following questions by indicating
which method(s) will yield the desired result:
(a) A minimum-phase continuous-time system has all its poles and zeros in the LH
s-plane. If a minimum-phase continuous-time system is transformed into a
discrete-time system, which method(s) will result in a minimum-phase discrete-
time system?
(b) If the continuous-time system is an all-pass system, its poles will be at locations
k
s in the LH s-plane, and its zeros will be at corresponding locations
k
s in the
RH s-plane. Which design method(s) will result in an all-pass discrete-time
system?
(c) Which design method(s) will guarantee that

0
0
( ) ( )
j
c
H e H j

=
=
=
(d) If the continuous-time system is a bandstop filter, which method(s) will result in a
discrete-time bandstop filter?
(e) Suppose that
1
( ) H z ,
2
( ) H z and ( ) H z are transformed versions of
1
( )
c
H s ,
2
( )
c
H s and ( )
c
H s , respectively. Which design method(s) will guarantee that
1 2
( ) ( ) ( ) H z H z H z = whenever
1 2
( ) ( ) ( )
c c c
H s H s H s = ?
(f) Suppose that
1
( ) H z ,
2
( ) H z and ( ) H z are transformed versions of
1
( )
c
H s ,
2
( )
c
H s and ( )
c
H s , respectively. Which design method(s) will guarantee that
1 2
( ) ( ) ( ) H z H z H z = whenever
1 2
( ) ( ) ( )
c c c
H s H s H s = ?
(g) Assume that two continuous-time transfer functions satisfy the condition

2
1
2
2
, 0 ( )
( )
, 0
j
c
j
c
e H j
H j
e
'
1

1
=
!
1
< 1
+

If
1
( ) H z and
2
( ) H z are transformed versions of
1
( )
c
H s and
2
( )
c
H s ,
respectively, which design method(s) will result in discrete-time systems such that

2
1
2
2
, 0
( )
( )
, 0
j
j
j
j
e
H e
H e
e
'
1
< <
1
=
!
1
< < 1
+

(Such systems are called 90-degree phase splitters.)
(h) Suppose ( )
c
H s has constant group delay, i.e.
( ) ( ) , ( )
j
c
H j A j e A j
=
Which design method(s) will guarantee that ( )
j
H e

also has constant group
delay?

YH Leung (2005, 2006, 2008)
6. A discrete-time lowpass filter with the following design specification is to be realised as
an IIR filter.
Stop-band attenuation: 20 dB
Pass-band deviation: 1 dB
Pass-band edge: 0.2
Stop-band edge: 0.55
Design the minimum order IIR filter by applying the bilinear transformation on a
continuous-time Butterworth filter.

7. Repeat Question 6 but use the impulse invariance method.

8. Repeat Question 6 except perform the design by applying the bilinear transformation
on a continuous-time Type I Chebyshev filter.

9. A discrete-time lowpass filter with the following design specification is to be realised as
an IIR filter.
0.95 ( ) 1.05,
j
H e

< < 0 0.25
0 ( ) 0.1,
j
H e

< < 0.4
Design the minimum order IIR filter by applying the bilinear transformation on a
continuous-time Type I Chebyshev filter.

YH Leung (2005, 2006, 2008)
Answers

2. (a) 0.4 rad
c
=
(b) 0.3571 rad
c
=

3. (a) 2500 rad/s
c
=
(b) 2000 rad/s
c
=

4. (a)
1
1
1
2
1
( )
z
T
z
H z
= ; ( )
2
[ ] [ ] [ 1]
T
h n u n u n =
(b) ( )
2
[ ] [ 1] [ ] [ 1]
T
y n y n x n x n = ; system is unstable
(c)
2 2
( ) cot
j
T
j
H e

=

(d)
1
1
1
2
1
( )
z
T
z
G z
= ;
( )
2
[ ] 2( 1) [ ] [ ]
n
T
g n u n n =
(e)
2
2
( ) tan
j j
T
G e

=

(f) Yes, provided the same T is used in both bilinear transforms.
Both the discrete-time integrator and differentiator give good approximations only for
low frequency (relative to the sampling frequency) input signals.
YH Leung (2005, 2006, 2008)
5. (a) Bilinear
(b) Bilinear
(c) Bilinear; impulse invariance only if
2
0
( ) 0
k
c k
T
k
H j

(d) Bilinear
(e) Bilinear
(f) Bilinear and impulse invariance
(g) Bilinear
(h) Impulse invariance only if ( )
c
H j is suitably bandlimited.

6. 2.319332 3 = N N
Assuming 1 T = , 0.813973 1.088743 0.951358 =
c c

( )( )
1 3
1 2 1
0.0428552 (1 )
1 0.909227 0.441019 1 0.355308
( )

=
z
z z z
H z

7. 2.939072 3 = N N
Assuming 1 T = , 0.787017 =
c

1
1 2 1
0.787017(1 0.769359 )
0.787017
1 1.047895 0.455201 1 0.455201
( )

=
z
z z z
H z

8. 0.508847 =
1.874807 2 = N N
Assuming 1 T = , 0.649839 =
c

1 2
1 2
0.070422 (1 )
1 1.199678 0.515739
( )

=
z
z z
H z

9. 0.4708 =
3.2652 4 = N N
Assuming 1 T = , 0.8284
c
=
( )( )
1 4
1 2 1 2
0.0047234 (1 )
1 1.2811 0.8121 1 1.4135 0.5634
( )
z
z z z z
H z

=

YH Leung (2005, 2006, 2008)
Solutions

1. (Includes OSB Problem 7.1a)
(a)
2
1
1
( )
( )
at
s a
t e u t

= L

2
[ ] ( ) [ ] ( ) [ ]
anT aT n
c
h n T h nT T nT e u n T n e u n

= = =
and from a table of z-transforms,

( )
1
2
1
( ) 2
1 ( )
( )
aT
aT
e z
e z
H z T

=

(b)
( )
2 2
1 1
2
( )
o
o o
o
j
s a j s a j
s a

=
Therefore, from Eq. (49)

( ) ( ) ( ) 1 1
1
1 2 2
2
1 ( ) 1 ( )
( 2 sin )
2
1 2 cos
( )
a j T a j T
o o
aT
o
aT aT
o
j
T T
e z e z
e j T z jT
e T z e z
H z

l

l
l
=
=

(c)
( )
2 2
1 1 1
2
( ) o o
o
s a
s a j s a j
s a

=

( ) ( ) ( ) 1 1
1
1 2 2
1
2
1 ( ) 1 ( )
2 (2cos )
2
1 2 cos
( )
a j T a j T
o o
aT
o
aT aT
o
T T
e z e z
e T z
T
e T z e z
H z

l

l
l
=
=

2. (OSB Problems 7.9 and 7.10 (modified))
(a)
3
2000 (0.2 10 ) 0.4 rad
c c
T
= = =
(b)
( )
( )
3
2000 (0.2 10 ) 1 1
2 2
2tan 2tan 0.3571 rad
c
T
c

= = =

3. (OSB Problems 7.11 and 7.12)
(a)
3
4
0.1 10
2500 rad/s
c
c
T

= = =
(b)
( ) ( )
3
2
2 2
2 2
1 10
tan tan 2000 rad/s
c
c
T

= = =
YH Leung (2005, 2006, 2008)
(a)
1
1
2 1
1
1
1
1
2
1
( ) ( )
z
T
z
z
T
c
s
z
H z H s
= =
( )
2
[ ] [ ] [ 1]
T
h n u n u n =

(b) LCCDE given by
( )
2
[ ] [ 1] [ ] [ 1]
T
y n y n x n x n =
Discrete-time system is not implementable since it has a pole on the unit circle
(at 1 z = ) and so is unstable.

(c)
2 2
2
2 2
2
2cos
1
2 2 2 2 2
2 sin 1
( ) cot
j j j
j j j
e e e j
T T T T
j
j e e e
H e

= = = =

As can be seen, approximation is good only for low frequency (relative to the
sampling frequency) input signals.

(d)
1
1
2 1
1
1
1
1
2
1
( ) ( )
z
T
z
z
c
s T
z
G z G s
= =

( ) ( )
1
2 2
[ ] ( 1) [ ] ( 1) [ 1] 2( 1) [ ] [ ]
n n n
T T
g n u n u n u n n
= =

(e)
2 2
2
2 2
2
2 sin
1 2
2 2 2
2
2cos 1
( ) tan
j j j
j j j
j
e e e j j
T T T T
e e e
G e

= = = =

Again, approximation is good only for low frequency (relative to the sampling
frequency) input signals.

YH Leung (2005, 2006, 2008)
(f) If the same value of T is used in both bilinear transforms, then

2
2 2 2
( ) ( ) cot tan 1
j j j
T
j T
H e G e

= =
i.e., the two discrete-time systems are inverses of each other.

5. (OSB Problems 7.25 and 7.41)
(a) Recall bilinear transformation maps LH s-plane to inside unit circle and RH
s-plane to outside unit circle, so all poles and zeros of ( )
c
H s are mapped to inside
unit circle. In other words, ( ) H z is also minimum phase.
Impulse invariance method does not preserve minimum phase, as can be seen
from following counter-example.
( )
( )( )
2 1
1 2 1 1 2 1
1 (9 8 )
( 10)
9 8 9 8
( 1)( 2) 1 2
1 ( ) 1 ( ) 1 ( ) 1 ( )
T T
T T T T
T e e z
s
T T
s s s s
e z e z e z e z

= =
It can be verified the zero of ( ) H z lies outside the unit circle if 1 T = .

(b) A continuous-time all-pass system has frequency response
( ) constant,
c
H j = < <
but the bilinear transformation performs only a one-to-one frequency mapping,
specifically, < < is mapped into < < through =
( )
1
2tan 2 T
. Accordingly, the discrete-time frequency response is given by

( ) constant,
j
H e

= < <
which is also all-pass.
Impulse invariance method results in
2
( ) ( )
j
c
T T
k
H e H j j k
=
=

The aliasing terms can destroy the all-pass nature of ( )
c
H s .

(c) Bilinear transformation maps
0
( )
c
H j
=
to
0
( )
j
H e

=
. Therefore, it always
satisfies required condition.
2 2
0
0
0
( ) ( ) ( ) ( 0)
j
c c c
T T T
k k
k
H e H j j k H j k H j

=
= =
=
= =

Therefore, required condition is satisfied only if
2
0
( ) 0
c
T
k
k
H j k

YH Leung (2005, 2006, 2008)
(d) Bilinear transformation only warps the frequency axis. Therefore, a continuous-
time bandstop filter will be transformed to a bandstop filter in discrete-time.
2
( ) ( )
j
c
T T
k
H e H j j k
=
=

and the aliasing terms may fill in the stop band.

(e) By the bilinear transformation,
1 1
2 1 2 1
1 1
1 1
1 1
2 1 2 1
1 1
1 1
1 2
1 2
1 2
( ) ( ) ( ) ( )
( ) ( )
( ) ( )
z z
T T
z z
z z
T T
z z
c c c
s s
c c
s s
H z H s H s H s
H s H s
H z H z

= =
= =
= =
=
=

Therefore, desired result holds.
Suppose
1 ( ) 1 ( )
1 1
1 2
( ) ( ) ( )
b a b a
c c c
s a s b s a s b
H s H s H s

= = =
Impulse invariance method yields
( )
( )( )
1
1 1 1 1
( ) ( )
( )
1 1 1 1
( )
aT bT
aT bT aT bT
e e z
T b a T b a
T
b a
e z e z e z e z
H z

= =
But, clearly
1 1 1 2
1 1
( ) ( ) ( )
aT bT
T T
e z e z
H z H z H z

=
i.e., desired result does not hold for impulse invariance method.

(f) By the bilinear transformation,

1 1
2 1 2 1
1 1
1 1
1 1
2 1 2 1
1 1
1 1
1 2
1 2
1 2
( ) ( ) ( ) ( )
( ) ( )
( ) ( )
z z
T T
z z
z z
T T
z z
c c c
s s
c c
s s
H z H s H s H s
H s H s
H z H z

= =
= =
= =
=
=

Therefore, desired result holds.
For the impulse invariance method,

2
2 2
1 2
2 2
1 2
( ) ( )
( ) ( )
( ) ( )
j
c
T T
k
c c
T T T T
k
c c
T T T T
k k
H e H j j k
H j j k H j j k
H j j k H j j k
=

= =
=
=
=

YH Leung (2005, 2006, 2008)
i.e.
1 2
( ) ( ) ( )
j j j
H e H e H e

=
Therefore, desired result also holds for impulse invariance method.

(g) By the bilinear transformation,
2
1 1
2
( ) ( tan )
j
c
T
H e H j

= and
2
2 2
2
( ) ( tan )
j
c
T
H e H j

=
2
1
1 2
2
2 2 2
2
( tan )
( )
( tan ) 2 ( )
, 0
, 0
j
c
T
j
c
T
j
H j
H e
H j j H e
e
e

'
1
< <
1
1
= =
!
1
< < 1
1+

i.e., bilinear transformation will give desired result.
In the case of the impulse invariance method,
2
1 1
( ) ( )
j
c
T T
k
H e H j j k
=
=
and
2
2 2
( ) ( )
j
c
T T
k
H e H j j k
=
=

Therefore, the phase relationship is not guarantee to hold because of aliasing.

(h) The bilinear transformation warps the frequency axis. Therefore, the resulting
discrete-time filter will no longer have a constant group delay.
The impulse invariance method will also not result in a constant group delay
because of aliasing, unless ( )
c
H j is bandlimited to
T T

< < .

YH Leung (2005, 2006, 2008)
6. We first portray the specification graphically as follows
( )
j
H e

1
1
1
2

where
10 1 1 1
20log (1 ) 1 1 0.891251 g = = =

10 2 2 2
20log ( ) 20 0.1 g = = =
0.2
p
=
and 0.55 =
s

For the bilinear transformation method, the corresponding band edges of the
continuous-time filter, assuming 1 T = , are given by

( )
0.2
2
2tan 0.649839
p
= =
and
( )
0.55
2
2tan 2.341699
s
= =

Therefore, the minimum order of the continuous-time Butterworth filter is given by

( )
2 2
2 1
2 2
1 2
1
1
log
1
2.319332 3
2
log
s
p
g g
g g
N N

( )
= =

We next determine the cut-off frequency
c
. We require
at
p
= ,
( )
6
2
1
1
1
p c
g

or
( )
1
2
6
1
1 1
0.813973
p
c
g
l
l
=
and at
s
=
( )
6
2
1
2
1
s c
g

or
( )
1
2
6
2
1 1
1.088743
s
c
g
l
l
=
We choose
c
to be the average of the above two limits, i.e., 0.951358 =
c
.

YH Leung (2005, 2006, 2008)
The poles of ( )
c
H s are thus given by

1,2
0.951358 120 = s
and
3
0.951358 = s
leading to

( )
( )
3 1 2
1 2 3 1 2 3
2
1 1
(1 )(1 ) 1 ( )( )
0.861057
0.951358 0.905082 0.951358
( )

= =
=
s s s
c
s s s s s s s s s s s s
s s s
H s

Taking bilinear transform, we get

2
1 1 1
1 1 1
1 3
1 2 2
0.861057
1 1 1
2 0.951358 2 0.905082 2 0.951358
1 1 1
0.861057 (1 )
4(1 ) 0.9513585(1 ) 0.905082 (1
( )

1
1 1 1 1

( ) ( ) (( ) )
( )

=
=
z z z
z z z
z
z z
H z
( )( )
( )( )
1 2 1 1
1 3
1 2 1
) 2(1 ) 0.951358(1 )
0.0428552 (1 )
1 0.909227 0.441019 1 0.355308

=
z z z
z
z z z

0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
(x)
|
H
(
e
j
)
|

7. For the impulse invariance method, the band edges of the continuous-time filter are
given, assuming 1 T = , by

1
0.2 =
and
2
0.55 =

YH Leung (2005, 2006, 2008)
Therefore, the minimum order of the continuous-time Butterworth filter is given by

( )
( )
2 2
2 1
2 2
1 2
2
1
1
1
log
1
2.939072 3
2
log
g g
g g
N N

= =
For
c
, we find the
c
that will cause the filter to just meet the passband edge
specification. This will result in the filter more than meeting the requirement at the
stopband edge but this is prudent in view of the aliasing in frequency response of the
discrete-time filter, which is a characteristic of the impulse invariance method. Thus,

( )
1
1
2
6
1
1 1
0.787017
c
g
l
l
= =
and the poles of ( )
c
H s are given by

1,3
0.787017 120 = s
and
2
0.787017 = s
resulting in

3 1 2
1 2 3 1 2 3
1 1 1
1 1 1
( )

= =
k k k
c
s s s s s s s s s s s s
H s
where
3
1 2 3
3 3
1 1
( ) ( )
= =

= =

c
i n i n
n n
n i n i
s s s
i
s s s s
k
The discrete-time transfer function is given by

3 1 2
1 1 1
1 2 3
1 1 1
1
1 2
1 1 1
0.393508 0.227192 0.393508 0.227192
0.787017
1 (0.523948 0.425064) 1 0.455201 1 (0.523948 0.425064)
0.787017(1 0.769359 )
0.
1 1.047895 0.455201
( )

=
=
=
s s s
k k k
e z e z e z
j j
j z z j z
z
z z
H z
1
787017
1 0.455201

z

0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
(x)
|
H
(
e
j
)
|

YH Leung (2005, 2006, 2008)
8. Recall, for the bilinear transformation method, the corresponding band edges of the

( )
0.2
1
2
2tan 0.649839
= =
and
( )
0.55
2
2
2tan 2.341699
= =

Proceeding with the design of the continuous-time Chebyshev I filter

2
1
1
1
0.508847

= =
g
g

( )
2 2
2 1
2 2
1 2
2
1
1 1
1
1
cosh
1.874807 2
cosh
g g
g g
N N
= =

1
0.649839 = =
c

2
1 1
1 4.170247

= =

( )
1 1
1
2
0.776215
= =
N N
a

( )
1 1
1
2
1.265903
= =
N N
b
Poles:
1,2
0.356676 0.581690 = s j

2
1
1
( 0) 0.891251

= =
c
H j

1 2
1 2
0.891251
( )( )
( )

=
s s
c
s s s s
H s

1 2
1 2
1 1 1 2
1 1
1 2
1 1
1 1
0.891251 0.070422 (1 )
1 1.199678 0.515739
2 2
( )

1 1

( )( )
= =
z z
z z
s s z
z z
s s
H z
0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
(x)
|
H
(
e
j
)
|

YH Leung (2005, 2006, 2008)
9. First, since the maximum gain (1.05) does not equal 1, we must scale the amplitude gain
of the design specification as follows.

0.95
1.05
( ) 1,
j
H e

< < 0 0.25

0.1
1.05
0 ( ) ,
j
H e

< < 0.4

Next, for the bilinear transformation method, the corresponding band edges of the

( )
0.25
1
2
2tan 0.8284
= =
and
( )
0.4
2
2
2tan 1.4531
= =

We now proceed with the design of the continuous-time Type I Chebyshev filter:

0.95
1
1.05
g =

0.1
2
1.05
g =

2
1
1
1
0.4708
g
g

= =

( )
2 2
2 1
2 2
1 2
2
1
1 1
1
1
cosh
3.2652 4
cosh
g g
g g
N N
= =

1
0.8284
c
= =

2
1 1
1 4.4721

= =

( )
1 1
1
2
0.3833
N N
a
= =

( )
1 1
1
2
1.0709
N N
b
= =
Poles:
1,4
0.1215 0.8197 s j =

2,3
0.2933 0.3395 s j =

2
1
1
( 0) 0.9048
c
H j

= =

1 2 3 4
1 2 3 4
0.9048
( )( )( )( )
( )
s s s s
c
s s s s s s s s
H s

=
YH Leung (2005, 2006, 2008)
By the bilinear transform the discrete-time filter is given by

( )
1 2 3 4
1 1 1 1
1 1 1 1
1 4 2 3
1 1 1 1
1 1 1 1
1 4
1 2
0.9048
2 2 2 2
0.1313(1 )
5.1726 6.6268 4.2006 5.374
( ) 1.05
z z z z
z z z z
s s s s
s s s s
z
z z
H z

' 1 1'' 1 1'

1 11 1
1 11 1
! !! !

1 11 1 ( )( ) ( )( )
1 11 1 + ++ +

=
=
( )
( )( )
1 2
1 4
1 2 1 2
7 7.5974 3.0279
0.0047234 (1 )
1 1.2811 0.8121 1 1.4135 0.5634
z z
z
z z z z

=

where the gain of the original design specification has been restored by the last
multiplier 1.05
0 0.2 0.4 0.6 0.8 1
0
0.2
0.4
0.6
0.8
1
1.2
(x)
|
H
(
e
j
)
|

YH Leung (2005, 2006, 2007, 2008)
Problems

1. Compute the DFT of each of the following finite-length sequences considered to be of
length N (where N is even):
(a) [ ] [ ] = x n n
(b) [ ] [ ], 0 1 =
o o
x n n n n N
(c)
1, even, 0 1
[ ]
0, odd, 0 1
'
1
1
=
!
1

1+
n n N
x n
n n N

(d)
2
2
1, 0 1
[ ]
0, 1
'
1
1
=
!
1

1
+
N
N
n
x n
n N

(e)
, 0 1
[ ]
0, otherwise
'
1

1
=
!
1
1+
n
a n N
x n

2. Consider the finite-length sequence
3
0
[ ] 6, 5, 4, 3
n
x n
=
= . Determine the sequences
(a)
1
4
[ ] [ 2 ], 0 3 x n x n n =
(b)
2
4
[ ] [ ], 0 3 x n x n n =

3. A finite-duration sequence [ ] x n of length 8 has the 8-point DFT [ ] X k shown below.
k
[ ] X k
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7

A new sequence of length 16 is defined by
[ 2], even
[ ]
0, odd
x n n
y n
n
'
1
1
=
!
1
1+

Sketch the 16-point DFT of [ ] y n .

YH Leung (2005, 2006, 2007, 2008)
4. Two 8-point sequences
1
[ ] x n and
2
[ ] x n shown below have DFTs
1
[ ] X k and
2
[ ] X k
respectively. Determine the relationship between
1
[ ] X k and
2
[ ] X k .
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
a
b
c
d
e
n n
e
d
a
b
c x
1
[n] x
2
[n]

5. Two finite-length sequences [ ] x n and
1
[ ] x n are shown below.
0
1
2 3 4 5 6 7 0 1 2 3
4
5 6 7
n n
x [n] x
1
[n]
1 1
1 1
1 1

The N-point DFTs of these sequences, [ ] X k and
1
[ ] X k , respectively, are related by
2
2
1
[ ] [ ]
N
j k
X k X k e
=
where N is an unknown constant. Can you determine a value of N consistent with
the figures drawn for [ ] x n and
1
[ ] x n ? Is your choice for N unique? If so, justify
your answer. If not, find another choice of N consistent with the information given.

6. Consider the real finite-length sequence [ ] x n shown below.
n
[ ] x n
0 1 2 3 4 5 6 7 -1
4
3
2
1

(a) Determine the finite-length sequence [ ] y n whose 6-point DFT is
2
6
4
[ ] [ ]
j k
Y k X k e

=
where [ ] X k is the 6-point DFT of [ ] x n
(b) Determine the finite-length sequence [ ] w n whose 6-point DFT is
[ ] Re [ ] W k X k =
(c) Determine the finite-length sequence [ ] q n whose 3-point DFT is
[ ] [2 ], 0, 1, 2 Q k X k k = =

YH Leung (2005, 2006, 2007, 2008)
7. [ ] x n is a real-valued finite-length sequence of length 10 and is nonzero in the interval
from 0 to 9, i.e.,
[ ] 0, 0, 10
[ ] 0, 0 9
x n n n
x n n
= <

( )
j
X e

denotes the DTFT of [ ] x n , and [ ] X k denotes the 10-point DFT of [ ] x n .
Determine a choice for [ ] x n so that [ ] X k is real valued for all k and
( ) ( ) ,
j j
X e A e

= <
where ( ) A is real and is a nonzero real constant.

8. Let [ ] x n be a real-valued 5-point sequence whose 7-point DFT is denoted by [ ] X k .
If Re [ ] X k is the 7-point DFT of [ ] g n , show that [0] [0] g x = , and determine the
relationship between [1] g and [1] x .

9. Suppose we have two 4-point sequences [ ] x n and [ ] h n as follows:
( )
2
[ ] cos , 0, 1, 2, 3
[ ] 2 , 0, 1, 2, 3
n
n
x n n
h n n
= =
= =

(a) Calculate the 4-point DFT [ ] X k .
(b) Calculate the 4-point DFT [ ] H k .
(c) Calculate
4
[ ] [ ] [ ] y n x n h n = by doing the circular convolution directly.
(d) Calculate [ ] y n of Part (c) by multiplying the DFTs of [ ] x n and [ ] h n and
performing an inverse DFT.

10. Consider the two 4-point sequences
3
1 0
[ ] 1, 0, 2, 1
n
x n
=
= and
3
2 0
[ ] 0, 1, 0,
n
x n a
=
=
where a is unknown. Suppose the 4-point circular convolution of
1
[ ] x n and
2
[ ] x n is
given by

4
3
1 2
0
[ ] [ ] [ ] 1, 1, 1, 1
n
y n x n x n
=
= =
Can you specify a uniquely? If so, what is a? If not, give two possible values of a.

YH Leung (2005, 2006, 2007, 2008)
11. In this exercise, we investigate the plausibility of the zero-insertion method described in
the Lecture Notes for zero-padding in frequency, particularly when the original DFT
has an even number of points.
Consider the continuous-time signal
( ) sin 2 cos 4 =
c
x t A t B t
Clearly, ( )
c
x t is periodic with period 1 sec and bandwidth 2 Hz. Suppose ( )
c
x t is
sampled over 1 sec at 4, 5, 6 and 7 Hz with the first sample taken at 0 = t to yield
the 4-, 5-, 6- and 7-point sequences
4
[ ] x n ,
5
[ ] x n ,
6
[ ] x n and
7
[ ] x n , respectively. Show
that

3
4
0
[ ] 0, 2 , 4 , 2
=
=
k
X k j A B j A

4
5
5
4 0
[ ] 0, 2 , 2 , 2 , 2
=
=
k
X k j A B B j A

5
6
6
4 0
[ ] 0, 2 , 2 , 0, 2 , 2
=
=
k
X k j A B B j A

6
7
7
4 0
[ ] 0, 2 , 2 , 0, 0, 2 , 2
=
=
k
X k j A B B j A

12. Consider the finite-length sequence
5
0
[ ] 1, 1, 1, 1, 1, 1
=
=
n
x n . Let ( ) X z be the z-
transform of [ ] x n . If we sample ( ) X z at
(2 4)
=
j k
z e , 0, 1, 2, 3 = k , we obtain
(2 4)
1
[ ] ( )

=
=
j k
z e
X k X z , 0, 1, 2, 3 = k .
Sketch the sequence
1
[ ] x n obtained as the inverse DFT of
1
[ ] X k

13. Suppose
1
[ ] x n is an infinite-length, stable (i.e., absolutely summable) sequence with
z-transform given by
1
1
1
3
1
( )
1
X z
z

Suppose
2
[ ] x n is a finite-length sequence of length N, and its N-point DFT is given by
2
2 1
[ ] ( ) , 0, 1, , 1 j k N
k e
X k X z k N
=
= =
Determine
2
[ ] x n .

14. Consider the two finite-length sequences
3
1 0
[ ] 1, 2, 1, 3
n
x n
=
= and
2
[ ] x n =

5
0
0, 2, 0, 0, 1, 1
n=
. What is the smallest N such that the N-point circular
convolution of
1
[ ] x n and
2
[ ] x n equals the linear convolution of these sequences, i.e.,
such that
1 2 1 2
[ ] [ ] [ ] [ ]
N
x n x n x n x n = ?

YH Leung (2005, 2006, 2007, 2008)
15. Determine a real sequence [ ] x n that satisfies all of the following three conditions:
Condition 1: The DTFT of [ ] x n has the form
1 2
( ) 1 cos cos 2
j
X e A A
=
where
1
A and
2
A are some unknown constants.
Condition 2: The sequence [ ] [ 3] x n n evaluated at 2 n = is 5.
Condition 3: For the 3-point sequence
2
0
[ ] 1, 2, 3
n
w n
=
= , the result of the 8-point
circular convolution of [ ] w n and [ 3] x n is 11 when 2 n = ; i.e.,
7
8
0
2
[ ] [ 3 ] 11
m
n
w m x n m
=
=
=

16. Two finite-length sequences
1
[ ] x n and
2
[ ] x n , which are zero outside the interval
0 99 n , are circularly convolved to form a new sequence [ ] y n .
100
99
1 2 1 2
100
0
[ ] [ ] [ ] [ ] [ ] , 0 99
k
y n x n x n x k x n k n
=
= =

If, in fact,
1
[ ] x n is nonzero only for 10 39 n , determine the set of values of n for
which [ ] y n is guaranteed to be identical to the linear convolution of
1
[ ] x n and
2
[ ] x n .

17. Consider two finite-duration sequences [ ] x n and [ ] y n . [ ] x n is zero for 0 n < , 40 n ,
and 9 30 n < < , and [ ] y n is zero for 10 n < and 19 n , as shown below.
0
0 9
10 19
30 39
n
n
[ ] y n
[ ] x n

Let [ ] w n denote the linear convolution of [ ] x n and [ ] y n . Let [ ] g n denote the 40-point
circular convolution of [ ] x n and [ ] y n :
[ ] [ ] [ ] [ ] [ ]
k
w n x n y n x k y n k
=
= =

40
39
40
0
[ ] [ ] [ ] [ ] [ ]
k
g n x n y n x k y n k
=
= =

YH Leung (2005, 2006, 2007, 2008)
(a) Determine the values of n for which [ ] w n can be nonzero.
(b) Determine the values of n for which [ ] w n can be obtained from [ ] g n . Explicitly
specify at what index values n in [ ] g n these values of [ ] w n appear.

18. Suppose [ ] x n and [ ] y n are two finite-duration sequences, both with the same length N.
Show, directly, that

1
0
[ ] [ ] [ ]
N
m r
x m y n m w n rN

= =
=

where [ ] y n is the periodic extension of [ ] y n , and [ ] w n
is the linear convolution of

[ ] x n with [ ] y n .

19. We want to implement the linear convolution of a 10,000-point sequence with an FIR
impulse response that is 100 points long. The convolution is to be implemented by using
DFTs and IDFTs of length 256.
(a) If the overlap-add method is used, what is the minimum number of 256-point
DFTs and the minimum number of 256-point IDFTs needed to implement the
convolution of the entire 10,000-point sequence? Justify your answer.
(b) If the overlap-save method is used, what is the minimum number of 256-point
DFTs and the minimum number of 256-point IDFTs needed to implement the
convolution of the entire 10,000-point sequence? Justify your answer.
(c) Suppose the DFTs and IDFTs of Parts (s) and (b) are implemented using FFTs
and IFFTs. Compare the number of arithmetic operations (multiplications and
additions) required in the overlap-add method, the overlap-save method, and
direct convolution.

Other problems to attempt from Oppenheim, Schafer with Buck, Chapter 8:
8, 11, 13, 14, 16, 18, 19, 25, 27, 29, 36, 37, 50
YH Leung (2005, 2006, 2007, 2008)
Answers

1. (a) [ ] 1 = X k (b)
2
[ ]

=
o
j kn N
X k e
(c)
2, 0, 2
[ ]
0, otherwise
'
= 1
1
=
!
1
1+
N k N
X k (d) 2
2
1
2, 0
, odd
[ ]
0, even

'
= 1
1
1
1
=
!
1
1
1
1
+
j k N
e
N k
k
X k
k

(e)
2
1
1
[ ]

=
N
j k N
a
ae
X k

2. (a)
3
1
0
[ ] 4, 3, 6, 5
n
x n
=
=
(b)
3
2
0
[ ] 6, 3, 4, 5
n
x n
=
=

3.
k
[ ] Y k
0 1 2 3 4 5 6 7 15

4.
2 1
[ ] ( 1) [ ]
k
X k X k =

5. 5 N =

6. (a)
5
0
[ ] 2, 1, 0, 0, 4, 3
n
y n
=
=
(b)

5
3 3
2 2
0
[ ] 4, , 1, 1, 1,
n
w n
=
=
(c)
2
0
[ ] 5, 3, 2
n
q n
=
=

YH Leung (2005, 2006, 2007, 2008)
7. We first establish the following result. Let [ ] Y k be the DFT of an arbitrary real N-point
sequence. Define
[ ] [ ] [ ] X k Y k Y k
=
Clearly, [ ] X k is real for all k. Now, from the properties of DFTs, the IDFT of [ ] X k
is given by
[ ] [ ] [ ] [ ] [ ] [ ] [ ]
N N
x n y n y n y n y n y n y N n
= = =
Accordingly, if [ ] X k is an N-point DFT and is real for 0 ( 1) k N , then its
IDFT [ ] x n must satisfy

[ ], 1 1
[ ]
arbitrary real , 0
x N n n N
x n
n
'
1
1
=
!
1
=
1+

In the context of this Problem, the above result means

[1] [9]
[2] [8]
[3] [7]
[4] [6]
x x
x x
x x
x x
=
=
=
=

and [0] x and [5] x are arbitrary but real.
Next, recall our earlier work on linear phase FIR filters. The condition on ( )
j
X e

implies [ ] x n is linear phase. Therefore, [ ] x n must satisfy
[ ] [ 1 ], 0 ( 1) x n x N n n N =
or in the context of this Problem,

[0] [9]
[1] [8]
[2] [7]
[3] [6]
[4] [5]
x x
x x
x x
x x
x x
=
=
=
=
=

Combining both sets of conditions on [ ] x n , we see that [ ] x n must be a constant for
0 ( 1) n N .

8. [0] [0] g x = ;
1
2
[1] [1] g x =

9. (a)
4
4
3
0
[ ] 1 0, 2, 0, 2
k
j
k
X k e
=
= =
(b)
2 4 6
4 4 4
3
0
[ ] 1 2 4 8 15, 3 6, 5, 3 6
k k k
j j j
k
H k e e e j j

=
= =
(c)(d)
3
0
[ ] 3, 6, 3, 6
n
y n
=
=
YH Leung (2005, 2006, 2007, 2008)
10. 1 a =

11. Sampling ( )
c
x t at 4, 5 and 6 Hz, we get

3
4 0
[ ] , ( ), , ( )
=
=
n
x n B A B B A B
2 2
5
5 5 5 5
4
2 2
5 5 5 5
0
[ ] , sin cos , sin cos ,
sin cos , sin cos

=
=

n
x n B A B A B
A B A B

1 1
6
2 2
5
1 1
2 2
0
[ ] , ( 3 ), ( 3 ),
, ( 3 ), ( 3 )
=
=

n
x n B A B A B
B A B A B

DFTs can be found, after some algebra, through the DFT matrices

4
1 1 1 1
1 1
1 1 1 1
1 1
FFT

l
l
l
=
l
l
l
l
j j
j j

2 5 4 5 4 5 2 5
4 5 2 5 2 5 4 5
5
4 5 2 5 2 5 4 5
2 5 4 5 4 5 2 5
1 1 1 1 1
1
1
1
1
FFT

l
l
l
l
l
=
l
l
l
l
l
l
j j j j
j j j j
j j j j
j j j j
e e e e
e e e e
e e e e
e e e e

3 2 3 2 3 3
2 3 2 3 2 3 2 3
2 3 2 3 2 3 2 3
3 2 3 2 3 3
6
1 1 1 1 1 1
1 1
1 1
1 1 1 1 1 1
1 1
1 1
FFT

l
l
l
l
l
l
=
l
l
l
l
l
l
l
j j j j
j j j j
j j j j
j j j j
e e e e
e e e e
e e e e
e e e e

12.
3
1
0
[ ] 2, 2, 1, 1
n
x n
=
=

13.
( )
( )
1
3
1 1
2
3
1
[ ]
N
n
x n
=

14. 9 N =

YH Leung (2005, 2006, 2007, 2008)
15. From Condition 1, we can write

1 2
2 2
1 2
2 2
( ) 1 cos cos 2 1 ( ) ( )
A A
j j j j j
X e A A e e e e

= = (1)
Comparing the above expression with the definition of the DTFT
( ) [ ]
j jn
n
X e x n e

=
=

(2)
we see that [ ] x n is the non-causal finite-length sequence

2 1 1 2
2
2 2 2 2
2
[ ] , , 1, ,
A A A A
n
x n
=
= (3)
Next, from Condition 2
( ) ( )
2 2
[ ] [ 3] [ 3] [ 1] 5
n n
x n n x n x
= =
= = = (4)
Combining (3) and (4), we get
[ 1] [1] 5 x x = = (5)
and
1
10 A = (6)
Finally, with reference to Condition 3, we first remark that [ 3] x n is a causal 6-point
sequence

2 1 1 2
5
2 2 2 2
0
[ 3] 0, , , 1, ,
A A A A
n
x n
=
= (7)
Therefore, the linear convolution of [ 3] x n with the causal 3-point sequence [ ] w n
will yield a causal 3 6 1 8 = point causal sequence. Moreover, from our study on
the relationship between linear and circular convolutions, we see the 8-point circular
convolution of [ 3] x n with [ ] w n will equal their linear convolution. In other words,

8
[ ] [ 3] [ ] [ 3] w n x n w n x n = (8)
It can be shown, using the table method to calculate linear convolutions, that

7
0
[ ] [ 3] 0, , 5 2 , 11 3 , 22, 13 , 15 2 , 3
n
w n x n a a a a a a
=
= (9)
where
2
[ 2] [2] 2 a x x A = = = . Hence, from Condition 3

2
[ ] [ 3] 5 2 11
n
w n x n a
=
= = (10)

2
3 or 6 a A = = (11)
Summarising

2
2
[ ] 3, 5, 1, 5, 3
n
x n
=
=

16. 39 99 n

YH Leung (2005, 2006, 2007, 2008)
17. (a) 10 28 n and 40 58 n
(b) [ ] [ ] g n w n = for 19 39 n , and [ ] [ 40] g n w n = for 0 9 n

19.
Multiplications Addtions
Overlap-add 528,384 798,813
Overlap-save 536,576 804,864
Direct convolution 1,009,900 999,801

YH Leung (2005, 2006, 2007, 2008)
Solutions

(a)
1
2 2 0
0
[ ] [ ] 1
N
j kn N j k N
n
X k n e e

=
= = =

(b)
1
2 2
0
[ ] [ ]
o
N
j k n N j kn N
o
n
X k n n e e

=
= =

(c)
2
2
4
1
1
1 2 4
1
0 0
even
2, 0, 2
[ ]
0, otherwise
N
j k
j k N
N
e j kn N j km N
e
n m
n
N k N
X k e e
= =
'
= 1
1
= = = =
!
1
1+

(d)
1
2
2
2
1 2
2
2 1
2 2
1 1
1
1
1 0
2, 0
, odd
[ ]
0, even
N
j k
j k
N
j k N
j k N
j k
N
N
j kn N
e e
e
e
e n
N k
k
X k e
k
=
'
= 1
1
1
1
= = = =
!
1
1
1
1
+

(e)
2 2
2 2
1 1
1 ( ) 1 2 2
1 1
0 0
[ ] ( )
j k N N N j k
j k N j k N
N N
ae a e n j kn N j k N n
ae ae
n n
X k a e ae

= =
= = = =

2
2 2
1 1
1 ( ) 2 2
1
0 0
1
1
[ ] ( )
j k N N
j k N
N
j k N
N N
ae n j kn N j k N n
ae
n n
a
ae
X k a e ae
=
= = = =

It is easier to solve the problems graphically.
We start by plotting [ ] x n , the periodic extension of [ ] x n , then plot [ 2] x n and [ ] x n .
1
[ ] x n and
2
[ ] x n are just the first four points of [ 2] x n and [ ] x n , starting at 0 n = .
n
[ ] x n
0 1 2 3 0 1 2 3
6
5
4
3
n
[ 2] x n
0 1 2 3 0 1 2 3
6
5
4
3
0 1 2 3 0 1 2 3
[ ] x n
n

Thus,
3
1
0
[ ] 4, 3, 6, 5
n
x n
=
=
and
3
2
0
[ ] 6, 3, 4, 5
n
x n
=
=
YH Leung (2005, 2006, 2007, 2008)
2
16
2 2 2 2 2
16 16 16 16 16
2 2 2 2 2
16 16 16 16 16
2
16
15
0
0 1 2 14 15
0 1 2 14 15
(2 )
0
[ ] [ ]
[0] [1] [2] [14] [15]
[0] 0 [1] [7] 0
[ ]
j kn
n
j k j k j k j k j k
j k j k j k j k j k
j k n
n
Y k y n e
y e y e y e y e y e
x e e x e x e e
x n e
=
=
=
=
=
2
8
7
7
0
[ ]
j kn
n
x n e
=
=

(i) For 0 7 k , clearly [ ] [ ] Y k X k =
(ii) For 8 15 k , we observe that

2 2 2 2
8 8 8 8
8 ( 8) j kn j kn j n j k n
e e e e

= =
[ ] [ 8] Y k X k =
In other words, [ ] Y k is just [ ] X k at 0 7 k , and repeated at 8 15 k
k
[ ] Y k
0 1 2 3 4 5 6 7 15

The two sequences are related through the circular shift

2 1
8
[ ] [ 4 ] x n x n =
Therefore, from time shifting property of DFTs

2
8
4
2 1 1 1
[ ] [ ] [ ] [ ]( 1)
j k
jk k
X k X k e X k e X k

= = =

If
2
2
1
[ ] [ ]
N
j k
X k X k e
= , then by the time shifting property of DFTs

1
[ ] [ 2 ]
N
x n x n =
From the figures, one can conclude that 5 N = and it is unique.

YH Leung (2005, 2006, 2007, 2008)
(a) We define firstly the zero-padded 6-point sequence

5
0
[ ] 4, 3, 2, 1, 0, 0
n
x n
=
=
By the time shifting property of DFTs, it follows that

5
0 6
[ ] [ 4 ] 2, 1, 0, 0, 4, 3
n
y n x n
=
= =
(b) First observe that

1
2
[ ] Re [ ] [ ] [ ] W k X k X k X k
= =
Therefore, from the properties of DFTs

1 1
6 6 2 2
[ ] [ ] [ ] [ ] [ ] w n x n x n x n x n
= =
But
5
0 6
[ ] 4, 0, 0, 1, 2, 3
n
x n
=
=
Hence

5 5
1
0 0 2
5
3 3
2 2
0
[ ] 4, 3, 2, 1, 0, 0 4, 0, 0, 1, 2, 3
4, , 1, 1, 1,
n n
n
w n
= =
=
=
=

(c) The DFT of [ ] x n has been decimated by 2. Therefore, by the time-aliasing result
Eq. (39)

2 2
0 0
2
0
[ ] [ 3 ] [ 3] [ ], 0, 1, 2
1, 0, 0 4, 3, 2
5, 3, 2
r
n n
n
q n x n r x n x n n
=
= =
=
= = =
=
=

See Answers.

YH Leung (2005, 2006, 2007, 2008)
Suppose [ ] G k is the DFT of [ ] g n . It then follows that

1
2
[ ] Re [ ] [ ] [ ] G k X k X k X k
= =
whereupon

1 1
7 7 2 2
[ ] [ ] [ ] [ ] [ ] g n x n x n x n x n
= =
Hence

7
1 1
[0] [0] [ 0 ] [0] [0] [0]
2 2
g x x x x x = = =
and
7
1 1 1 1
[1] [1] [ 1 ] [1] [6] [1] 0 [1]
2 2 2 2
g x x x x x x = = = =
where [6] 0 x = since [ ] X k was found by zero-padding [ ] x n to 7 points.

(a)
3
0
[ ] 1, 0, 1, 0
=
=
n
x n

2 4 6
4 4 4
4
4
3
2 4
0
3
0
[ ] [ ] [0] [1] [2] [3]
1
0, 2, 0, 2
k k k
k
j j j
j kn
n
j
k
X k x n e x x e x e x e
e

=
= =
=
=

(b)
3
0
[ ] 1, 2, 4, 8
=
=
n
h n

2 4 6
4 4 4
2 4 6
4 4 4
3
2 4
0
[ ] [ ] [0] [1] [2] [3]
1 2 4 8
k k k
k k k
j j j
j kn
n
j j j
H k h n e h h e h e h e
e e e

=

= =
=

(c)
[ ]
[ ]
h n
x n
1 2 4 8 1 2 4 8
1 1 2 4 8 1 2 4 8
0 0 0 0 0 0 0 0 0
-1 -1 -2 -4 -8 -1 -2 -4 -8
0 0 0 0 0 0 0 0 0
0 = n

4
3
0
[ ] [ ] [ ] 3, 6, 3, 6
n
y n x n h n
=
= =
YH Leung (2005, 2006, 2007, 2008)
(d) [ ] [ ] [ ] Y k X k H k =

( )( )
( ) ( )
( ) ( )
4 2 4 6
4 4 4 4
2 4 6 4 6 8 10
4 4 4 4 4 4 4
2 4 6 4 6 0 2
4 4 4 4 4 4 4
2 4 6
4 4 4
1 1 2 4 8
1 2 4 8 2 4 8
1 2 4 8 2 4 8
3 6 3 6
k k k k
k k k k k k k
k k k k k k k
k k k
j j j j
j j j j j j j
j j j j j j j
j j j
e e e e
e e e e e e e
e e e e e e e
e e e

=
=
=
=

But

2 2 4 6
4 4 4 4
3
0
[ ] [ ] [0] [1] [2] [3]
kn k k k
j j j j
n
Y k y n e y y e y e y e

=
= =

Comparing the two expressions for [ ] Y k , we get

3
0
[ ] 3, 6, 3, 6
n
y n
=
=

It can be readily shown that

4
3
1 2
0
[ ] [ ] 1, 1 2 , , 2
n
x n x n a a a
=
=
Therefore, we require
1 2 1, 1, and 2 1 a a a = = =
which are satisfied by 1 a = .

11. (Made up)
See Answers.

YH Leung (2005, 2006, 2007, 2008)
From the expression for time-aliasing, Eq. (40), we have (for
1
4 = N )

2 1 1
[ ] [ ] [ ], 0, , ( 1) = = x n x n N x n n N
[ ] x n
n 0
5
[ 4] x n
n 0
4
n 0 3
1
[ ] x n


2
[ ] x n is
1
[ ] x n time aliased to N samples. Also, from Table of z-transforms,

( )
1
1
3
[ ] [ ]
n
x n u n =
Therefore, for 0 1 n N ,

( ) ( )
( ) ( )
( ) ( )
( )
( )
1
3
0
1 1
2 1
3 3
0
1 1
3 3
1 1
3 3
0
1 1
3
1
[ ] [ ] [ ]
N
n rN n rN
r r r
n rN
r
m
n N
m
n
x n x n rN u n rN

= = =
= = =
=
l
=
l
l
=


1
[ ] x n is a 4-point sequence while
2
[ ] x n is a 6-point sequence. It follows from
pp. 7-24 of the Lecture Notes that we are required to zero pad both sequences to
4 6 1 9 = points, i.e., 9 N = .

See Answers.
YH Leung (2005, 2006, 2007, 2008)
If
1
[ ] x n is nonzero for 10 39 n only, then the linear convolution
1 2
[ ] [ ] x n x n is
nonzero for (0 10) (39 99) n , i.e., 10 138 n , only. (See Problem 2 of
Discrete-Time Signals and Systems a Review.)
Now, from Section 7.3 of the Lecture Notes, we see that a 100-point circular
convolution of a 40-point sequence with a 100-point sequence will result in the first 39
points of the circular convolution not equalling the corresponding linear convolution,
and the last 39 points of the linear convolution not being found. Also, from a closer
study of Fig. 8, one can deduce that the 10 leading zeros in
1
[ ] x n do not affect the
previous conclusion, i.e. the 100-point circular convolution finds only points 39 to
138 39 99 = of the linear convolution.

(a) Decomposing [ ] x n as follows
0
0 9
10 19
30 39
n
n
[ ] y n
[ ] x n
1
[ ] x n
2
[ ] x n

we see that
( )
1 2 1 2
[ ] [ ] [ ] [ ] [ ] [ ] [ ] [ ] [ ] [ ] w n x n y n x n x n y n x n y n x n y n = = =
The first term
1
[ ] [ ] x n y n is nonzero from 0 10 10 n = = to 9 19 28 n = = ,
while the second term
2
[ ] [ ] x n y n is nonzero from 30 10 40 n = = to n =
39 19 58 = . The values of n for which [ ] w n can be nonzero are thus
10 28 and 40 58 n n

(b) From Problem 16, the 40-point circular convolution of [ ] x n and [ ] y n is related to
their linear convolution by
[ ] [ 40 ] [ ] [ 40], 0 39
r
g n w n r w n w n n
=
= =

where [ ] w n is nonzero for 0 58 n . Therefore, the points of [ ] w n for
40 58 n are time-aliased to the points in the range 0 18 n , and the
YH Leung (2005, 2006, 2007, 2008)
remaining points of [ ] w n in the range 19 39 n are free of aliasing terms. In
other words,
[ ] [ ] for 19 39 g n w n n =
Now, from Part (a), we see that [ ] 0 w n = for 0 9 n . Therefore, we have the
further result
[ ] [ 40] for 0 9 g n w n n =

18. (Made up)
Result follows since both sides of given equation equals N-point circular convolution
of [ ] x n with [ ] y n . Regardless, we shall attempt a direct proof.
Now,

1 1 1
0 0 0
[ ] [ ] [ ] [ ] [ ] [ ]
N N N
m m r r m
x m y n m x m y n m rN x m y n m rN

= = = = =
' '
1 1
1 1
= =
! !
1 1
1 1 + +

Consider next the linear convolution of [ ] x n and [ ] y n

1
0
[ ] [ ] [ ]
N
m
w n x m y n m
=
=

Hence

1 1
0 0
[ ] [ ] [( ) ] [ ] [ ]
N N
m m
w n rN x m y n rN m x m y n m rN

= =
= =

and
1
0
[ ] [ ] [ ]
N
m r
x m y n m w n rN

= =
=

(a) From pp. 7-24 of the Lecture Notes, segment length L is given by
1 100 1 256 L P L = =
i.e. 157 L=
Therefore, from Eq. (44), number of segments required is
10000 157 63.69 64 R Q L l l l = = = =
l l l

and number of DFTs required is 64 1 65 = where extra DFT is used to compute
DFT of the impulse response of the FIR filter. Number of IDFTs required is 64.
YH Leung (2005, 2006, 2007, 2008)
(b) From Section 8.2 of the Lecture Notes, number of segments is given by
( 1) ( 1) (10000 256 1) (256 100 1) 65 R Q L L P l l = = =
l l

Number of DFTs required is thus 65 1 66 = where extra DFT is used to
compute DFT of the impulse response of the FIR filter. Number of IDFTs
required is 65.
(c) Overlap-Add
Number of complex operations to compute FFTs and IFFTs:
Complex multiplications ( )
2
(65 64) 256 2 log 256 132096 =
Complex additions
2
(65 64) 256log 256 264192 =
Therefore, number of real operations required to compute FFTs and IFFTs:
Real multiplications 4 132096 528384 =
Real additions 2 132096 2 264192 792576 =
Number of times output segments overlap is 1 63 Q = . (Final points of last
output segment do not overlap with any other output segment.) Number of
overlapping points is 1 100 1 99 P = = . Therefore, to add overlapping output
segment points, we require 63 99 6237 = real additions.
Summarising, overlap-add method requires: 528,384 real multiplications
798,813 real additions
Overlap-Save
Number of complex operations to compute FFTs and IFFTs:
Complex multiplications ( )
2
(66 65) 256 2 log 256 134144 =
Complex additions
2
(66 65) 256log 256 268288 =
Therefore, number of real operations required to compute FFTs and IFFTs:
Real multiplications 4 134144 536576 =
Real additions 2 134144 2 268288 804864 =
Unlike overlap-add, overlap-save method does not require any post-FFT/IFFT
arithmetic operations.
Therefore, overlap-save method requires: 536,576 real multiplications
Direct Convolution
For direct convolution, output sequence has length 10000 100 1 10099 = . Each
output point requires 100 real multiplications and 99 real additions.
Therefore, direct convolution requires: 1,009,900 real multiplications

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Curtin University of Technology, Australia 3-15

YH Leung (2005, 2006, 2007)

= , determine the frequency response of the continuous-time filter

= , determine the frequency response of the continuous-time filter

to produce a third system with system function ( ) H z and impulse

. Therefore, from Eqs. (77) and (76)

= . Therefore, second summation is non-zero only when 0 k k

. Accordingly, the discrete-time frequency response is given by

' 1 1'' 1 1'

is the linear convolution of

= , then by the time shifting property of DFTs

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