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06 Tanal

The document discusses the transformation analysis of Linear Time-Invariant (LTI) systems, focusing on their frequency response, ideal filters, phase distortion, and system stability. It explains the mathematical relationships between the input and output of LTI systems in both time and z-transform domains, as well as the characteristics of ideal lowpass and highpass filters. Additionally, it covers the concepts of inverse systems, impulse responses for rational system functions, and the conditions for stability and causality in these systems.

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0% found this document useful (0 votes)
6 views21 pages

06 Tanal

The document discusses the transformation analysis of Linear Time-Invariant (LTI) systems, focusing on their frequency response, ideal filters, phase distortion, and system stability. It explains the mathematical relationships between the input and output of LTI systems in both time and z-transform domains, as well as the characteristics of ideal lowpass and highpass filters. Additionally, it covers the concepts of inverse systems, impulse responses for rational system functions, and the conditions for stability and causality in these systems.

Uploaded by

sisekelomhay9
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Transform analysis of LTI systems

Oppenheim and Schafer, Second edition pp. 240–339.


For LTI systems we can write

X
y[n] = x[n] ∗ h[n] = x[k]h[n − k].
k=−∞

Alternatively, this relationship can be expressed in the z-transform domain as

Y (z) = H(z)X(z),

where H(z) is the system function, or the z-transform of the system impulse
response.
Recall that a LTI system is completely characterised by its impulse response,
or equivalently, its system function.

1 Frequency response of LTI systems

The frequency response H(ejω ) of a system is defined as the gain that the
system applies to the complex exponential input ejωn . The Fourier transforms
of the system input and output are therefore related by

Y (ejω ) = H(ejω )X(ejω ).

In terms of magnitude and phase,

|Y (ejω )| = |H(ejω )||X(ejω )|


∢Y (ejω ) = ∢H(ejω ) + ∢X(ejω ).

In this case |H(ejω )| is referred to as the magnitude response or gain of the


system, and ∢H(ejω ) is the phase response or phase shift.

1
1.1 Ideal frequency-selective filters

Frequency components of the input are suppressed in the output if |H(ejω )| is


small at those frequencies. The ideal lowpass filter is defined as the LTI
system with frequency response

1 |ω| ≤ ωc
Hlp (e ) =

0 ωc < |ω| ≤ π.

Its magnitude and phase are

1
|H(ejω )|

−π ωc 0 ωc π
ω
π
∢H(ejω )

−π
−π ωc 0 ωc π
ω
This response, as for all discrete-time signals, is periodic with period 2π. Its
impulse response (for −∞ < n < ∞) is
Z ωc  ωc
1 1 1
hlp [n] = ejωn dω = ejωn
2π −ωc 2π jn −ωc
1 1 jωc n sin(ωc n)
= (e − e−jωc n ) = ,
πn 2j πn
which for ωc = π/4 is

2
0.3

0.2
hlp [n] 0.1

−0.1
−20 −15 −10 −5 0 5 10 15 20
n

The ideal lowpass filter is noncausal, and its impulse response extends from
−∞ < n < ∞. The system is therefore not computationally realisable. Also,
the phase response of the ideal lowpass filter is specified to be zero — this is a
problem in that causal ideal filters have nonzero phase responses.
The ideal highpass filter is

0 |ω| ≤ ωc
Hhp (ejω ) =
1 ωc < |ω| ≤ π.

Since Hhp (ejω ) = 1 − Hlp (ejω ), its frequency response is


sin(ωc n)
hhp [n] = δ[n] − hlp [n] = δ[n] − .
πn

1.2 Phase distortion and delay

Consider the ideal delay, with impulse response

hid [n] = δ[n − nd ]

and frequency response


Hid (ejω ) = e−jωnd .

3
The magnitude and phase of this response are

|Hid (ejω )| = 1,
∢Hid (ejω ) = −ωnd , |ω| < π.

The phase distortion of the ideal delay is therefore a linear function of ω. This
is considered to be a rather mild (and therefore acceptable) form of phase
distortion, since the only effect is to shift the sequence in time. In other words,
a filter with linear phase response can be viewed as a cascade of a zero-phase
filter, followed by a time shift or delay.
In designing approximations to ideal filters, we are therefore frequently willing
to accept linear phase distortion. The ideal lowpass filter with phase distortion
would be defined as

e−jωnd |ω| ≤ ωc

Hlp (e ) =
0 ωc < |ω| ≤ π,

with impulse response


sin(ωc (n − nd )
hlp [n] = .
π(n − nd )

A convenient measure of linearity of the phase is the group delay, which


relates to the effect of the phase on a narrowband signal. Consider the
narrowband input x[n] = s[n] cos(ω0 n), where s[n] is the envelope of the
signal. Since X(ejω ) is nonzero only around ω = ω0 , the effect of the phase of
the system can be approximated around ω = ω0 by

∢H(ejω ) ≈ −φ0 − ωnd .

Thus the response of the system to x[n] = s[n] cos(ω0 n) is approximately


y[n] = s[n − nd ] cos(ω0 n − φ0 − ω0 nd ). The time delay of the envelope s[n]
of the narrowband signal x[n] with Fourier transform centred at ω0 is therefore
given by the negative of the slope of the phase at ω0 . The group delay of a

4
system is therefore defined as
d 
τ (ω) = grd[H(ejω )] = − arg[H(ejω )] .

The deviation of the group delay away from a constant indicates the degree of
nonlinearity of the phase. Note that the phase here must be considered as a
continuous function of ω.

2 System response for LCCD systems


Ideal filters cannot be implemented with finite computation. Therefore we
need approximations to ideal filters. Systems described by LCCD equations
N
X M
X
ak y[n − k] = bk x[n − k]
k=0 k=0

are useful for providing one class of approximation.


The properties of this class of system are best developed in the z-transform
domain. The z-transform of the equation is
N
X M
X
ak z −k
Y (z) = bk z −k X(z),
k=0 k=0

or equivalently
N M
! !
X X
ak z −k Y (z) = bk z −k X(z).
k=0 k=0

The system function for a system that satisfies a difference equation of the
required form is therefore
PM   QM
Y (z) k=0 bk z b0 k=1 (1 − ck z )
−k −1
H(z) = = PN = N
.
X(z) a
Q
k=0 ak z −k 0 k=1 (1 − dk z −1 )

5
Each factor (1 − ck z −1 ) in the numerator contributes a zero at z = ck and a
pole at z = 0. Each factor (1 − dk z −1 ) contributes a zero at z = 0 and a pole
at z = dk .
The difference equation and the algebraic expression for the system function
are equivalent, as demonstrated by the next example.
Example: second-order system
Given the system function
(1 + z −1 )2
H(z) = ,
(1 − 12 z −1 )(1 + 43 z −1 )
we can find the corresponding difference equation by noting that
1 + 2z −1 + z −2 Y (z)
H(z) = = .
1 + 14 z −1 − 38 z −2 X(z)
Therefore
1 3
(1 + z −1 − z −2 )Y (z) = (1 + 2z −1 + z −2 )X(z),
4 8
and the difference equation is
1 3
y[n] + y[n − 1] − y[n − 2] = x[n] + 2x[n − 1] + x[n − 2].
4 8

2.1 Stability and causality

A difference equation does not uniquely specify the impulse response of a LTI
system. For a given system function (expressed as a ratio of polynomials),
each possible choice of ROC will lead to a different impulse response.
However, they will all correspond to the same difference equation.
If a system is causal, it follows that the impulse response is a right-sided
sequence, and the region of convergence of H(z) must be outside of the
outermost pole.

6
Alternatively, if we require that the system be stable, then we must have

X
|h[n]| < ∞.
n=−∞

For |z| = 1 this is identical to the condition



X
|h[n]z −n | < ∞,
n=−∞

so the condition for stability is equivalent to the condition that the ROC of
H(z) include the unit circle.
Example: determining the ROC
The frequency response of the LTI system with difference equation
5
y[n] − y[n − 1] + y[n − 2] = x[n]
2
is
1 1
H(z) = = .
1 − 52 z −1 + z −2 (1 − 21 z −1 )(1 − 2z −1 )
There are three choices for the ROC:
• Causal: ROC outside of outermost pole |z| > 2 (but then not stable).
1
• Stable: ROC such that 2 < |z| < 2 (but then not causal).
1
• If |z| < 2
then the system is neither causal nor stable.
For a causal and stable system the ROC must be outside the outermost pole
and include the unit circle. This is only possible if all the poles are inside the
unit circle.

2.2 Inverse systems

The system Hi (z) is the inverse system to H(z) if

G(z) = H(z)Hi (z) = 1,

7
which implies that
1
H(z) = .
Hi (z)
The time-domain equivalent is

g[n] = h[n] ∗ hi [n] = δ[n].

The question of which ROC to associate with Hi (z) is answered by the


convolution theorem — for the previous equation to hold the regions of
convergence of H(z) and Hi (z) must overlap.
Example: inverse system for first-order system
Let H(z) be
1 − 0.5z −1
H(z) =
1 − 0.9z −1
with ROC |z| > 0.9. Then Hi (z) is
1 − 0.9z −1
Hi (z) = .
1 − 0.5z −1
Since there is only one pole, there are only two possible ROCs. The choice of
ROC for Hi (z) that overlaps with |z| > 0.9 is |z| > 0.5. Therefore, the
impulse response of the inverse system is

hi [n] = (0.5)n u[n] − 0.9(0.5)n−1 u[n − 1].

In this case the inverse is both causal and stable.


A LTI system is stable and causal with a stable and causal inverse if and only if
both the poles and zeros of H(z) are inside the unit circle — such systems are
called minimum phase systems.
The frequency response of the inverse system, if it exists, is
1
H(ejω ) = .
Hi (ejω )
Not all systems have an inverse. For example, there is no way to recover the

8
frequency components above the cutoff frequency that were set to zero by the
action of the lowpass filter.

2.3 Impulse response for rational system functions

If a system has a rational transfer function, with at least one pole that is not
cancelled by a zero, then there will always be a term corresponding to an
infinite length sequence in the impulse response. Such systems are called
infinite impulse response (IIR) systems.
On the other hand, if a system has no poles except at z = 0 (that is, N = 0 in
the canonical LCCDE expression), then
M
X
H(z) = bk z −k .
k=0

In this case the system is determined to within a constant multiplier by its


zeros, so the impulse response has a finite length:

M b
X n 0≤n≤M
h[n] = bk δ[n − k] =
0 otherwise
k=0

In this case the impulse response is finite in length, and the system is called a
finite impulse response (FIR) system.
Example: a first-order IIR system
Given a causal system satisfying the difference equation

y[n] − ay[n − 1] = x[n],

the system function is


1 z
H(z) = = , |z| > |a|.
1 − az −1 z−a

9
The condition for stability is |a| < 1. The inverse z-transform is

h[n] = an u[n].

Example: a simple FIR system


Consider the truncated impulse response

an 0≤n≤M
h[n] =
0 otherwise.

The system function is


M
X
n −n 1 − aM +1 z −M −1
H(z) = a z = .
n=0
1 − az −1

The zeros of the numerator are at

zk = aej2πk/(M +1) , k = 0, 1, . . . , M.

With a assumed real and positive, the pole at z = a is cancelled by a zero. The
pole-zero plot for the case of M = 7 is therefore given by
z−plane Im

7th
order
1
Re

The difference equation satisfied by the input and output of the LTI system is
the convolution
M
X
y[n] = ak x[n − k].
k=0

10
The input and output also satisfy the difference equation

y[n] − ay[n − 1] = x[n] − aM +1 x[n − M − 1].

3 Frequency response for rational system


functions

If a stable LTI system has a rational system function, then its frequency
response has the form
PM
k=0 bk e
−jωk

H(e ) = PN .
k=0 ak e −jωk

We want to know the magnitude and phase associated with the frequency
response. To this end, it is useful to express H(ejω ) in terms of the poles and
zeros of H(z):
  QM
b0 k=1 (1 − ck e )
−jω

H(e ) = QN .
a0 k=1 (1 − dk e −jω )

It follows that QM
b k=1 |1 − ck e |
−jω
jω 0
|H(e )| = QN .
a0 k=1 |1 − dk e −jω |

Therefore |H(ejω )| is the product of the magnitudes of all the zero factors of
H(z) evaluated on the unit circle, divided by the product of the magnitudes of
all the pole factors evaluated on the unit circle.
The gain in dB of H(ejω ), also called the log magnitude, is given by

Gain in dB = 20 log10 |H(ejω )|,

11
which for a rational system function takes the form
M
jω b0 X
20 log10 |H(e )| = 20 log10 + 20 log10 |1 − ck e−jω |
a0
k=1
N
X
− 20 log10 |1 − dk e−jω |.
k=1

Also
Attenuation in dB = −Gain in dB.
Thus a 60dB attenuation at frequency ω corresponds to |H(ejω )| = 0.001.
Also,

20 log10 |Y (ejω )| = 20 log10 |H(ejω )| + 20 log10 |X(ejω )|.

The phase response for a rational system function is


 X
 M N
jω b0 X
∢H(e ) = ∢ + ∢[1 − ck e −jω
]− ∢[1 − dk e−jω ].
a0
k=1 k=1

The zero factors contribute with a plus sign and the pole factors with a minus.
In the above equation, the phase of each term is ambiguous, since any integer
multiple of 2π can be added at each value of ω without changing the value of
the complex number. When calculating the phase with a computer, the angle
returned will generally be the principal value ARG[H(ejω )], which lies in the
range −π to π. This phase will generally be a discontinuous function,
containing jumps of 2π radians whenever the phase wraps. Appropriate
multiples of 2π can be added or subtracted, if required, to yield the continuous
phase function arg[H(ejω )].

12
ARG[H(ejω )]
π
0 ω
π
−π

arg[H(ejω )]
π
0 ω
π
−π
−2π
−3π
−4π

3.1 Frequency response of a single pole or zero

Consider a single zero factor of the form

(1 − rejθ e−jω )

in the frequency response. The magnitude squared of this factor is

|1 − rejθ e−jω |2 = (1 − rejθ e−jω )(1 − re−jθ ejω )


= 1 + r2 − 2r cos(ω − θ),

so the log magnitude in dB is

20 log10 |1 − rejθ e−jω | = 10 log10 [1 + r2 − 2r cos(ω − θ)].

The principle value of the phase for the factor is


 
r sin(ω − θ)
ARG[1 − rejθ e−jω ] = arctan .
1 − r cos(ω − θ)
These functions are periodic in ω with period 2π.

13
The following plot shows the frequency response for r = 0.9 and three
different values of θ:
Gain (dB) 10

−10
π
θ=0 θ= 2 θ=π
−20
π 3π
0 2 π 2 2π
ω
π
2
Phase (radians)

− π2 π 3π
0 2 π 2 2π
ω
Note that
• The gain dips at ω = θ. As θ changes, the frequency at which the dip
occurs changes.
• The gain is maximised for ω − θ = π, and for r = 0.9 the magnitude of
the resulting gain is
10 log10 (1 + r2 + 2r) = 20 log10 (1 + r) = 5.57dB.

• The gain is minimised for ω = θ, and for r = 0.9 the resulting gain is
10 log10 (1 + r2 − 2r) = 20 log10 |1 − r| = −20dB.

• The phase is zero at ω = θ.


Note that if the factor (1 − rejθ ejω ) occurs in the denominator, thereby
representing a pole factor, then the entire analysis holds with the exception that
the sign of the log magnitude and the phase changes.

14
The frequency response can be sketched from the pole-zero plot using a simple
geometric construction. Note firstly that the frequency response corresponds to
an evaluation of H(z) on the unit circle. Secondly, the complex value of each
pole and zero can be represented by a vector in the z-plane from the pole or
zero to a point on the unit circle.
Take for example the case of a single zero factor
z − rejθ
H(z) = (1 − rejθ z −1 ) = r < 1,
z
which corresponds to a pole at z = 0 and a zero at z = rejθ .
Im
φ3
z−plane
v2 v3
v1
θ ω Re

If the vectors v1 , v2 , and v3 = v1 − v2 represent respectively the complex


numbers ejω , rejθ , and ejω − rejθ , then
ejω − rejθ |v3 |
|H(ejω )| = |1 − rejθ e−jω | = = = |v3 |.
ejω |v1 |
The phase is

∢H(ejω ) = ∢(1 − rejθ e−jω ) = ∢(ejω − rejθ ) − ∢(ejω )


= ∢(v3 ) − ∢(v1 ) = φ3 − φ1 = φ3 − ω.

A vector such as v3 from a zero to the unit circle is referred to as a zero


vector, and a vector from a pole to the unit circle is called a pole vector.

15
Consider now the pole zero system depicted below:
Im

z−plane

v3
v1
v1
v3 φ3 ω
v2 Re

The frequency response for the single zero at different values of r and θ = π is
10
Gain (dB)

0
r=0.5
−10 r=0.7
r=0.9
−20 r=1
π 3π
0 2 π 2 2π
ω
π
2
Phase (radians)

− π2 π 3π
0 2 π 2 2π
ω
Note that the log magnitude dips more sharply as r approaches 1 (and at ω = π
tends to −∞ as r tends towards 1). The phase function has positive slope
around ω = θ. This slope increases as r approaches 1, and becomes infinite for
r = 1. In this case the phase function is discontinuous, with a jump of π

16
radians at ω = θ.
If r increases still further, to lie outside of the unit circle,
Im

z−plane

v1
v3
φ3
φ3 v2
Re

then the frequency response becomes

10
Gain (dB)

−10 r=1/0.9
r=1.25
−20 r=2.0
π 3π
0 2 π 2 2π
ω
π
Phase (radians)

−π π 3π
0 2 π 2 2π
ω

17
3.2 Frequency response with multiple poles and zeros

In general, the z-transform of a LTI system can be factorised as


  QM   QM
b0 (1 − c k z −1
) b0 k=1 (z − ck )
H(z) = QN
k=1
= z N −M QN .
a0 k=1 (1 − d k z −1 ) a 0 k=1 (z − dk )
Depending on whether N is greater than or less than M , the factor z N −M
represents either N − M zeros at the origin, or M − N poles at the origin. In
either case, the z-transform can be written in the form
QM0
k=0(z − zk )
H(z) = K QN
k=0(z − pk )
0

where z1 , . . . , zM0 are the zeros, and p1 , . . . , pN0 the poles of H(z). This
representation could also be obtained by merely factorising H(z) in terms of z
rather than z −1 .
The frequency response of this system is
QM0 jω
i=0 (e − zk )
H(ejω ) = K QN .
0
(e jω − p )
i=0 k

The magnitude is therefore


QM0jω
jω i=0 |e − zk |
|H(e )| = |K| QN0 ,

i=0 |e − pk |
and the phase is
M0
X N0
X
jω jω
∢H(e ) = ∢(e − zk ) − ∢(ejω − pk ).
i=0 i=0

In the z-plane, (ejω − zk ) is simply the vector from the zero zk to the point on
the unit circle. The term |ejω − zk | is the length of this vector, and
∢(ejω − zk ) is the angle that it makes with the positive real axis. Similarly, the
term (ejω − pk ) corresponds to the vector from the pole pk to the point on the
unit circle.

18
It follows then that the the magnitude response is the product of the lengths of
the zero vectors, divided by the product of the lengths of the pole vectors. The
phase response is the sum of the angles of the zero vectors, minus the sum of
the angles of the pole vectors. Thus, for the two pole, two zero system
Im
φ1 V1
z−plane
p1
U1
U2 θ1
θ2
z2 V2 z1 Re
p2 φ2

the frequency response is


U1 U2
|H(ejω )| = |K|
V1 V2
∢H(ejω ) = θ1 + θ2 − (φ1 + φ2 ).
Here K is a constant factor which cannot be determined from the pole zero
diagram alone, but only serves to scale the magnitude.

4 Realisation structures for digital filters


The difference equation, the impulse response, and the system function are all
equivalent characterisations of the input-output relation for a LTI discrete-time
system. For implementation purposes, systems described by LCCDEs can be
implemented by structures consisting of an interconnection of the basic
operations of addition, multiplication by a constant, and delay. The desired
form for the interconnections depends on the technology to be used.

19
Discrete-time filters are often represented in the form of block or signal flow
diagrams, which are convenient for representing the difference equations or
transfer functions.
For example, the system with the difference equation

y[n] = x[n − 1] − b1 y[n − 1] + b2 y[n − 2] + b3 y[n − 3]

can be represented in a block diagram form as


x[n] x[n−1]
z −1
y[n]
z −1
−b1
y[n−1]
z −1
b2
y[n−2]
z −1
b3
y[n−3]

The symbol z −1 represents a delay of one unit of time, and the arrows
represent multipliers (with the constant multiplication factors next to them).
The equivalent signal flow diagram is
z −1
x[n] y[n]
z −1
−b1
y[n−1]
z −1
b2
y[n−2]
z −1
b3
y[n−3]

The relationship between the diagrams and the difference equation is clear.

20
Many alternative filter structures can be developed, and they differ mainly with
respect to their numerical stability and the effects of quantisation on their
performance. A discussion of these effects can be found in many DSP texts.

21

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