EENG 5610: Digital Signal Processing

Class 5: Frequency-Domain Analysis of LTI Systems

Dr. Xinrong Li
Department of Electrical Engineering
University of North Texas

Outline
Frequency-Domain Characteristics of LTI Systems
Frequency Response of LTI Systems
Correlation Functions and Spectra at the Output of LTI

Systems
LTI Systems as Frequency-Selective Filters
Inverse Systems and Deconvolution

Dr. Xinrong Li

EENG 5610, UNT

Frequency-Domain Characteristics
of LTI Systems
Methodology
The basic excitation signals will be the complex exponentials and

sinusoidal functions.
Periodic signals can be represented with Fourier series in the form of

weighted sum of harmonically related complex exponentials.

Aperiodic signals can be viewed as a superposition of infinite number of
complex exponentials.
The LTI system is described by the frequency response H(w), which

is the Fourier transform of the impulse response h(n).

The frequency response function completely characterizes an LTI system

in the frequency domain.

Dr. Xinrong Li

EENG 5610, UNT

 Response to Complex Exponential and Sinusoidal Signals

x( n) Ae jwn ,
y ( n)

h(k ) x(n k ) h(k )[ Ae

jw( n k )

] A

h( k )e

jwk

jwn
jwn
e

AH
(
w
)
e

The response to the complex exponential input signal is also in the

form of complex exponential with the same frequency, but altered by

the multiplicative factor H(w).
As a result of such characteristic behavior, the exponential signal is

called an eigen function of the system. The multiplicative factor is called

an eigen value of the system.
Example
h( n) (05.1.1
.5) n u ( n),

H ( w)

x( n) Ae jn / 2 ,

h(n)e jwn

1
,
1 0.5e jw

n
H ( / 2)

y ( n) AH ( w)e jwn AH ( / 2)e jn / 2

Dr. Xinrong Li

1
2 j 26.6

e
,
1 j 0.5
5

2
Ae j (n / 2 26.6) ,
5

n .

5610, UNT
The only effect of the system on the input signalEENG
is to scale the amplitude

 The Fourier transform pair for the frequency response function:

1
jwn
H ( w) h( n)e , h(n)
H ( w)e jwn dw

2
n
For real-valued
impulse response
h(n):

H ( w) h(n)e jwn h(n) cos( wn) j h(n) sin( wn)

n

H R ( w) jH I ( w) | H ( w) | e j ( w)

HR(w) is an even function of w, HI(w) is an odd function, | H(w) | is an even

function,
is ),
anthen
odd yfunction.
If x(n)and
A(w)
cos( wn
(n) A | H ( w) | cos[ wn ( w)].
WeIf
can
derive
that:
x(also
n) easily
A sin( wn
), then
y (n) A | H ( w) | sin[ wn ( w)].

x(n) Ai cos( wi n i ),

If the input
i 1 arbitrary combination of sinusoids:
is an

H ( w) | H ( w) | e j ( w) ,

y (n) Ai | H ( wi ) | cos[ wi n i ( wi )]
i 1

Dr. Xinrong Li

EENG 5610, UNT

 Steady-State Response to Periodic Input Signals

Periodic signal can be represented with Fourier series:
N 1

x ( n ) ck e

j 2kn / N

0 n N 1;

k 0

1
ck
N

N 1

x ( n) e

j 2kn / N

0 k N 1

n0

By using the superposition property of the linear system, we can

derive the steady-state response of the LTI system H(w) to the

N 1
periodic input signal
x(n):
j 2kn / N
x ( n ) ck e

0 n N 1;

k 0

N 1

y ( n ) ck H (
k 0

2k j 2kn / N
)e
,
N

n ;

The Fourier transform of h(n), H(w), is periodic, with period 2.

2k
The response y(n)
d to
c Hthe
( periodic
), 0 signal
k Nx(n)
1 is also periodic with the
k

N series coefficients of y(n) are:

same period N. The Fourier

TheLilinear
Dr. Xinrong

EENG
5610,
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system change the shape of the periodic
input
signal
scaling the amplitude and shifting the phase of the Fourier series

 Response to Aperiodic Input Signals

Aperiodic finite-energy signals can be represented with Fourier
transforms: X ( w) x(n)e jwn , x(n) 1
X ( w)e jwn dw

2 2
n
From the convolution theorem:
y(n) = h(n)*x(n),

Y(w) = H(w)X(w)

The output sequence y(n) can be obtained from Y(w) using IFT. But

this method is seldom used. Instead, the z-transform is a simpler

method for determining the output sequence y(n).
2
2
2
Energy spectrum density: | Y ( w) | | H ( w) | | X ( w) |
S yy ( w) | H ( w) |2 S xx ( w)
Energy of the output signal:
1
1
Ey
S
(
w
)
dw

yy
2
2

Dr. Xinrong Li

| H ( w) |2 S xx ( w)dw

EENG 5610, UNT

Frequency Response of LTI Systems

Frequency response of the system with rational system

function, i.e., the LTI systems described in the time domain

by constant-coefficient difference equations.
If the system function H(z) converges on the unit circle, the

frequency response of the system:

When H(z) is a rational function:
M

B ( w)
H ( w)

A( w)

b e
k 0
N

1 ak e jwk
k 1

jwk

H ( w) H ( z ) z e jw

b0

(1 z e

jw

k 1
N

(1 p e
k 1

h( n)e

jwn

jw

| H ( w) |2 H ( w) H * ( w) H ( w) H ( w) H ( z ) H ( z 1 ) | z e jw

Coefficients ak and bk are real, but zk and pk may be complex valued.

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Dr. Xinrong Li

EENG 5610, UNT

Correlation Functions and Spectra at

the Output of LTI Systems
Input-Output Correlation Functions and Spectra
Input-output correlation relationships of an LTI system:
ryy (m) rhh ( m) * rxx (m)

ryx (m) h(m) * rxx ( m)

S yy ( z ) S hh ( z ) S xx ( z )

H ( z ) H ( z 1 ) S xx ( z )

S ( z ) H ( z )S ( z )
yx
xx

S yy ( w) | H ( w) |2 S xx ( z )

S yx ( z ) H ( w) S xx ( w)

H ( w) | X ( w) |2

Total energy1in the output

jwn signal:
jwm
ryy (m)
S
(
w
)
e
dw
,
S
(
w
)

r
(
m
)
e

yy
yy
yy
2
m
1
1
2
Ey
S
(
w
)
dw

|
H
(
w
)
|
S xx ( w)dw ryy (0)
yy

2
2

Dr. Xinrong Li

EENG 5610, UNT

LTI Systems as
Frequency-Selective Filters
Filter
Filter is a device that discriminates what passes through it, according

to some attributes of the objects applied at its input.

Here we use filter to describe an LTI system used to perform spectral
shaping or frequency-selective filtering.

LTI system can be viewed as a filter

Output signal spectrum: Y(w) = H(w)X(w)
By proper selection of the coefficients of H(w), LTI system can

selectively pass or attenuate certain frequency components.

Filtering is used in DSP for a wide variety of purposes:
Removal of undesired noise from desired signal,
Spectral shaping such as equalization of communication channels,
Signal/object detection in radar, sonar, and communications,
Spectral analysis of signals,
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Dr. Xinrong Li

EENG 5610, UNT

 Ideal Filter Characteristics

Classification of filters based on frequency-domain characteristics
Lowpass, highpass, bandpass, and bandstop/band-elimination filters.
Ideal filters have constant-gain, linear-phase passband and zero-gain stopband.

Group (or envelope) delay g(w):

H ( w) | H ( w) | e j ( w) ,

g ( w)

d( w)
dw

g(w) is the time delay that a signal component

of frequency w undergoes as it passes through.

When g(w) = n0 is a constant, all frequency
components of the input signal undergo the
same time delay.

A pure delay and amplitude scaling is usually

considered tolerable, and not a distortion.

Example:

H ( w)

| H ( w) | e jwn0 ,

0,

w1 w w2
otherwise

d( w)
n0
dw
Y ( w) X ( w) H ( w) CX ( w)e jwn0 , w1 w w2
( w) wn0

11

g ( w)

y (n) Cx(n n0 )

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 Ideal filters are not physically realizable but only serve as a

mathematical idealization of practical filters.

For example, ideal lowpass filter:

1,
H lp ( w)
0,

| w | wc
wc | w |

hlp ( n)

sin( wc n)
,
n

This ideal lowpass filter is non-causal and it is not absolutely summable and

therefore it is also unstable. Thus, such a filter is unrealizable.

Practical, physically realizable filters can be designed to approximate the
ideal frequency characteristics very closely.
Design of simple digital filters by the pole-zero placement method
The basic idea of such a method is to place poles near the points of the unit

circle corresponding to the frequencies to be emphasized, and to place zeros

near the frequencies to be deemphasized, under the following constraints:
All poles should be placed inside the unit circle in order for the filter to be stable.

However, zeros can be placed anywhere in the z-plane.

All complex zeros and poles must occur in complex-conjugate pairs in order for
the filter coefficients to be real.
M

The coefficient b0 in the system function is

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selected to normalized the frequency

response so that | H(w0) | = 1, where
w0 is a frequency in the passband.

H ( z)

b z

(1 z z

(1 p z
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k 0
N

1 ak z k
k 1

b0

k 1
N

k 1

)
)

 Lowpass, Highpass, and Bandpass Filters

Design principles:
In the design of lowpass digital filters, the poles should be placed near

the unit circle at points corresponding to low frequencies (near w = 0)

and zeros should be placed near or on the unit circle at points
corresponding to high frequencies (near w = ).
The opposite holds for highpass filters.

13

Dr. Xinrong Li

EENG 5610, UNT

 An example
1 a
1 az 1
1 a 1 z 1
H 2 ( z)
2 1 az 1
a 0.9
H1 ( z )

14

Dr. Xinrong Li

EENG 5610, UNT

 Example 5.4.1
Given the system function:

H ( z)

b0
(1 pz 1 ) 2

Determine b0 and p such that H(w) satisfy the condition:

H(0) = 1, | H(/4) |2 = 0.5.

Solution:

H ( w)

b0
,
(1 pe jw ) 2

H (0)

b0
,
(1 p ) 2

( z e jw )
b0

H( )
4
(1 pe j / 4 ) 2

b0 0.46

p 0.32

Example 5.4.2
Design a two-pole bandpass filter that has the center of its passband at w =

/2, zero frequency response at w = 0 and w = , and | H(4/9) | = 1/(2).

Solution:
j/2
1 p
2
Clearly, the filter must
z = -1.
(1 have
z 1 )(poles
1 z at
) 1,2 = re , and zeros at z1=1z and

Thus,

H ( z) G

(1 jrz 1 )(1 jrz 1 )

H ( z ) 0.15

1 0.7 z 2

The pole-zero placement method demonstrated in these examples is not

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a good method for designing digital filters. Systematic methods for

designing sophisticated digital filters for practical applications will be
discussed later.

Dr. Xinrong Li

EENG 5610, UNT

 A simple lowpass-to-highpass filter transformation

By using the frequency translation property of the Fourier transform, it is

possible to convert a lowpass filter to either a bandpass or highpass filter.

If hlp(n) is the impulse response of a lowpass filter with frequency response
Hlp(w), a highpass filter can be obtained by translating Hlp(w) by radians:
H hp ( w) H lp ( w )

hhp (n) (e j ) n hlp (n) (1) n hlp (n)

If the lowpass filter is described by the difference equation:

N

k 1

k 0

Lowpass filter : y (n) ak y (n k ) bk x(n k )

M

H lp ( w)

bk e jwk

k 0
N

1 ak e jwk

H hp ( w) H lp ( w )

k 1

(1) k bk e jwk

k 0
N

1 (1) k ak e jwk
k 1

k 1

k 0

Highpass filter : y (n) (1) k ak y (n k ) (1) k bk x(n k )

16

Dr. Xinrong Li

EENG 5610, UNT

 Digital Resonator
A digital resonator is a special two-pole bandpass filter with a pair of

complex-conjugate poles located near the unit circle.

The name resonator refers to the fact that the filter has a large magnitude

response (i.e., it resonates) in the vicinity of the pole.

The angular position of the poles determines the resonant frequency.
Digital resonator is useful in many applications, e.g., bandpass filtering,
speech generation, etc.
In the design of digital resonator with the resonant frequency at or

near w = w0, we select a pair of complex-conjugate poles and up to

jw
two zeros: p1, 2 re , 0 r 1
Case I Two zeros at origin:
0

H ( z)

b0
b0

(1 re jw0 z 1 )(1 re jw0 z 1 ) 1 2r cos(w0 ) z 1 r 2 z 2

| H ( w0 ) | 1

b0 (1 r ) 1 2r cos(2w0 ) r 2
1

1 r2
cos(w0 )
2r

Resonant frequency : wr cos

17

(1 z 1 )(1 z 1 )
1 z 2
( z) G
HCase
II Zeros
and
z=G-1:
jw0 at
jw
2
1 z = 1
0 1
1 2r cos(w0 ) z 1EENG
r 2 z 5610,
z )
Dr. Xinrong Li (1 re z )(1 re
UNT

w0

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EENG 5610, UNT

w0

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Dr. Xinrong Li

EENG 5610, UNT

 Notch Filters
A notch filter is a filter that contains one or more deep notches or,

ideally, perfect nulls in its frequency

response characteristics.
Notch filter is useful in many

applications where specific

frequency components must be
eliminated.

FIR notch filters

To create a null in the frequency response of a filter at the w0, we can

introduce a pair of complex conjugate zeros on the unit circle at w = w0:

H ( z ) b0 (1 e jw0 z 1 )(1 e jw0 z 1 ) b0 (1 2 cos w0 z 1 z 2 )

The problem with FIR notch filter is that the notch has relatively large

bandwidth, which means that other frequency components around the

desired null are severely attenuated.
An ad-hoc, trial-and-error method: we can introduce a pair of complexconjugate poles at w = w0 to reduce the bandwidth of the notch:
(1 e jw0 z 1 )(1 e jw0 z 1 )
H ( z ) b0
(1 re jw0 z 1 )(1 re jw0 z 1 )

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Dr. Xinrong Li

EENG 5610, UNT

w0

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Dr. Xinrong Li

EENG 5610, UNT

w0

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Dr. Xinrong Li

EENG 5610, UNT

 Comb Filters
A comb filter can be viewed as a notch filter in which the nulls occur

periodically across the frequency band.

An example Moving average FIR filter:
1 M
y ( n)
x(n k ),
M 1 k 0

1 M k
1 1 z ( M 1)
H ( z)
z M 1
M 1 k 0
1 z 1

e jwM / 2 sin( w( M 1) / 2)
H ( w)
M 1
sin( w / 2)

The filter has zeros on the unit circles: z = ej2k/(M + 1), 1 k M. The

pole at z = 1 is actually canceled by the zero at z = 1.

23

Dr. Xinrong Li

EENG 5610, UNT

 All-Pass Filters
An all-pass filter has a constant magnitude response for all

frequencies, that is | H(w) | = 1, 0 w .

Pure delay system H(z) = z-k is an all-pass filter.
More interesting all-pass filter:
N
1
N
a z N k

N A( z )
k
k 0 k
H ( z)

z
,
a

1
,
A
(
z
)

a
z

0
k
N
A( z )
k 0
k 0 ak z k
| H ( w) |2 H ( z ) H ( z 1 ) | z e jw 1
If z0 is a pole of H(z), then

1/z0 is a zero of H(z), that is,

the poles and zeros of H(z)
are reciprocals of one another.

All-pass filters are used as phase equalizers.

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When
Dr. Xinrong
Li

placed in series with a system, a phase equalizer

is designed
to
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compensate for the poor phase characteristics of the system and therefore

25

Dr. Xinrong Li

EENG 5610, UNT

 Digital Sinusoidal Oscillators

A digital sinusoidal oscillator is a limiting form of a two-pole

resonator with the complex-conjugate poles lie on the unit circle.

Two-pole resonator:
H ( z)

b0
b0
b0

(1 re jw0 z 1 )(1 re jw0 z 1 ) 1 2r cos(w0 ) z 1 r 2 z 2 1 a1 z 1 a2 z 2

y ( n) a1 y (n 1) y ( n 2) b0 ( n)
b0 r n
h ( n)
sin[( n 1) w0 ]u (n)
sin( w0 )

Digital sinusoidal oscillator:

Poles are located on the unit circle, that is, r = 1. If we choose

b0 = Asin(w0), then the impulse

response of the system is
a sinusoid:
h(n) = Asin[(n+1)w0]u(n).
Digital sinusoidal oscillator is

26

a basic component of digital

frequency synthesizer.
The impulse at n = 0 serves the purpose
of beginning the sinusoidal oscillation;

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 Coupled-form Oscillator
In some practical applications involving modulation of two sinusoidal

carriers in phase quadrature, there is a need to generate the sinusoids

Asin(w0n) and Acos(w0n).
The coupled-form oscillator can be implemented as following:

Coupled difference equations :

yc (n) cos(w0 ) yc (n 1) sin( w0 ) y s (n 1)

y s (n) sin( w0 ) yc (n 1) cos(w0 ) y s (n 1)

Initial conditions :
yc (1) A cos(w0 ),

y s ( 1) A sin( w0 )

Trigonomet ric formulas used :

27

cos( ) cos cos sin sin

sin( ) sin cos cos sin

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