Ders Notları Toplu

LECTURE NOTES ON
DIGITAL
SIGNAL
PROCESSING
Prof. Dr. A. Salim KAYHAN
Hacettepe University
Department of Electrical and Electronics Engineering
ANKARA-2014
ELE 407
Digital Signal Processing
Fall 2014
Place: E-?
Time: Wednesday 13:00-15:50
Course Outline:
W1-Sep.24
W2-Oct.01
W3-Oct.08
W4-Oct.15
Review of Discrete-time signals, systems, Fourier, Z-tr.

Sampling, Decimation, Interpolation.
Frequency Response .
Relation Between Magnitude and Phase, Flow Graph
Realization.
W5-Oct.22 Flow Graph Realization of FIR sys., Quantization.
W6-Oct.29 No Class. (Holiday)
W7-Nov.05 EXAM
W8-Nov.12 Butterworth and Chebyshev Filter Design.
W9-Nov.19 Butterworth and Chebyshev Filter, FIR Filter Design.
W10-Nov26 Discrete-time Fourier Series, DFT, Properties of DFT.
W11-Dec03 Convolution with DFT (Overlap-add/save). FFT.
W12-Dec10 Fast Fourier Transform(FFT).
W13-Dec17 EXAM.
W14-Dec24 2-D Signal Processing.
Textbook:
Discrete-time Signal Processing, Oppenheim and Schafer, Prentice-Hall.
Grading: 2Term Exams (50%), Final Exam (50%).
DIGITAL
SIGNAL
PROCESSING
Textbook: Discrete-Time Signal Processing, Oppenheim and Schafer, Prentice-Hall.
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Course Outline
Review
of Discrete-time signals, systems, Fourier tr.
Review
of Z-transform
Sampling,
Decimation, Interpolation
Time
and frequency response of systems
Flow
Graph Realization
Filter
Design
Discrete-time
Fourier Series, Discrete Fourier Transform
Fast
Fourier Transform
2-D
Signal Processing
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SIGNALS:A signal is a function of time representing a physical

variable, e.g. voltage, current, spring displacement, share market prices, number
of students asleep in the class, cash in the bank account.
Continuous time signals : x(t), x(t1,t2,...)

Speech signals, Image signals,
Video signals, Seismic signals,
Biomedical signals (ECG,EEG,...)
Discrete-time signals: x[n], x[n1,n2,...]
Inherently discrete-time (Stock market)
Discretized continuous-time(Most signals)
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SIGNALS:Deterministic or Random
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Speech data (8kHz):
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Discrete-time signal example:

Stock Market daily closing values
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Annual max. temperature and precipitation for Ankara

(data: MGM)
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Graphical Representation of Signals

Continuous-time
Signal
Discrete-time
Signal
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Reflection: x[-n]
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Time-scaling: x(at)
Time-scaling in discrete-time is not straightforward.

Discussed in Decimation and Interpolation.
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Time-shift: x[n-no]
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Basic Discrete-time Signals:

Unit Step Function :
0,
u[n]
1,
n 0
n 0
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Unit Impulse Function:

0,
[n ]
1,
n 0
n 0
[ n ] u [ n ] u [ n 1]
n
u[n]
[k ]
x [ n ] [ n ] x [ 0 ] [ n ]
Real Exponential:
x[n ] C en C a
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x [ n ] A e j o n
x [ n ] A cos( o n )
Complex Exponential:
Sinusoidal Signal:
x [ n ] Re
j ( o n )
{ Ae
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Periodicity Properties:
e
j (
2 ) n
j o n
Therefore, consider only
If periodic, then
e
(n N )
j
j o n
0 2
m
N
m and N integer.
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If periodic with fundamental period N,

then fundamental frequency is
2 / N
If there are a few exponentials in the form:
xk [n] e
jk
2
n
N
they are harmonically related

There are only N such distinct signals.
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SYSTEM:
Any process that transforms signals
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Interconnection of Systems:
Series(Cascade), Parallel, Series/Parallel.
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Properties:
Memoryless if output depends only on
input at the same time.
Example: Discrete-time systems with memory:
n
y[ n ]
x[ k ]
y[ n ] x[ n 1]
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Causality: Causal if output depends only on inputs

at the present time and in the past.
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Example: Causal system:
y[ n ] x[ n ] x[ n 1]
1
y (t )
C
x( )d
Example: Noncausal systems:
y[ n ] x[ n 1] x[ n ]
M
1
y[ n ]
x[ n k ]
2 M 1 k M
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Stability: Stable if small inputs do not lead to

diverging outputs.
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Example: Stable system:
y[ n ] x[ n ] x[ n 1]
Example: Unstable systems:

n
y[ n ]
x[ k ] ( n 1)u[ n ],
if x[ n ] u[ n ].
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Time Invariance: Time-invariant if a time shift in

input causes same time shift in output signal.
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y[ n ] nx [ n ]
Example:Time-varying:
Consider two signals
x1 [ n ],
x 2 [ n ] x1 [ n n o ]
y 2 [ n ] nx 2 [ n ]
y1 [ n ] nx1 [ n ]
shift
y 2 [ n ] nx 1[ n n o ]
y1 [ n ]
y1 [ n n o ] ( n n o ) x1 [ n n o ] y 2 [ n ]
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Linearity: Linear if posses superposition property:

Additivity and scaling (homogeneity):
1. Response to x [ n ] x [ n ]is y1[ n ] y 2 [ n ]
1
2
2. Response to
ax 1 [ n ]is
ay 1 [ n ]
Combining these two, we get:
ax1[ n ] bx 2 [ n ] ay 1[ n ] by 2 [ n ]
Example: Linear:
y[ n ] nx [ n ]
Example: Nonlinear:
y[ n ] ( x[ n ]) 2
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Linear Time-Invariant Systems:

LTI systems can be analyzed in great detail.
Many physical processess can be modeled by LTI
systems.
Unit impulse function will be used as building block
and response of LTI systems to a unit impulse will be
used to characterize such systems.
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Input/output relationship for a LTI system is given as:
y[ n ]
x[ k ]h[ n k ]
y[ n ] x[ n ] * h[ n ]
Convolution of x[n] and h[n]. h[n] is the impulse response.
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Example:
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Properties:
Commutative:
x[ n ] * h[ n ] h[ n ] * x[ n ]
x[ k ]h[ n k ]
r n k
x[ n r ]h[ r ]
Use of this property:

If it is easier; reflect and shift x[k] to obtain x[n-k] first, then
multiply with h[k] to find convolution result at time n.
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Associative:
x[ n ] * ( h1 [ n ] * h2 [ n ]) ( x[ n ] * h1 [ n ]) * h2 [ n ]
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Distributive:
x[ n ] * ( h1 [ n ] h2 [ n ]) x[ n ] * h1 [ n ] x[ n ] * h2 [ n ]
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Memory: System is memoryless if output depends only

on input at same time.Then, system is memoryless if:
h[ n ] K [ n ]
then
y[ n ] Kx [ n ].
Similarly for continuous-time, memoryless if
y ( t ) Kx ( t ).
hence
h (t ) K (t )
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Causal: Consider convolution sum :
y[ n ]
h[ k ] x[ n k ]
For a LTI system to be casual y[n] must not depend on

input x[k] for k > n.
If system is causal :
h [ n ] 0 , for n 0 .
Then,
y[ n ] h[ k ] x[ n k ]
k 0
x[ k ]h[ n k ]
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Stability: Stable if bounded inputs do not lead to

diverging outputs. Assume
x[ n ] B ,
for all n .
Then,
h[ k ] x[ n k ]
y[ n ]
y[ n ]
h[ k ] x[ n k ]
x[ n k ] B
with,
y[ n ] B
h[ k ]
h[ k ]
System is stable iff,
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Example: A system with impulse response h[n]=[nno] shifts the input
h[ n ]
[n n
] 1
Example: Consider the fallowing accumulator

n
y[ n ]
x[ k ]
this system has unit step as the impulse response
u[ n ] u[ n ] Unstable
n0
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Discrete-Time Fourier Transform (DTFT):

For a discrete-time signal x[n], DTFT is defined as
X (e
x [ n ]e
j n
the inverse DTFT (synthesis equation) is defined as

1
2
x[n ]
(e
)e
j n
Differences between continuous-time FT and DTFT:

X(ej) is periodic with 2.
Integral in synthesis eq. is over 2 interval (0 to 2 or - to
).
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x [ n ] a n u [ n ],
Example:Consider
a 1
X (e
a ne
( ae
j n
u[n]
X (e
n0
X (e
X (e
X (e
)n
1
1 ae
1
(1 a cos( )) ja sin( )
1
2
1 a 2 a cos( )
) tan
a sin( )
1 a cos( )
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Common DTFT Pairs:
x [ n ] 1 , for all n X ( e
2 ( 2 l )
x[ n ] [ n ] X ( e
) 1
x[ n ] e
j o n
X ( e
2 ( o 2 l )
x [ n ] cos( o n )
X (e
[ (
2 l ) (
2 l )]
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Convergence: Analysis equation will converge if x[n] is

absolutely summable or it has finite energy
x[ n ]
x[n ]
Example:Consider
1, n N
x[ n ]
0, n N
X (e
j n
n N
j N
1 e j ( 2 N
1 e j
1)
sin( ( N 1 / 2 ))
sin( / 2 )
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Properties of DTFT:
Linearity: DTFT is linear operator:
x [ n ] X ( e j )
y[n ] Y (e
ax [ n ] by [ n ] a X ( e
) b Y (e
Time and frequency shifting:

x[n ] X (e
x[n n o ] e
e
j o n
j n o
x[ n ] X ( e
X (e
j (
j
)
Symmetry: If x[n] is real, then

X ( e j ) X * ( e
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Real part is even, imaginary part is odd function of .

Similarly, magnitude is even, phase is odd funtion.
Time and frequency scaling: Time scaling is not
straightforward for discrete-time signals. Consider special case;
x[an], a = -1
x [ n ] X ( e j )
x [ n / k ], if n is multiple
x1[ n ]
0, if n is not multiple
X 1 ( e j ) X ( e jk )
of k
of k
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Parsevals relation: Energy is equal on both time and

frequency domains:
2
x[n ]
1
2
X (e
Since the right hand term is the total energy in frequency

2
domain,
X ( e j ) is called energy spectral density
function (or spectrum).
Convolution: Convolution in time corresponds to
multiplication in frequency domain.
y [ n ] x[ n ] * h[ n ] Y (e
) X (e
) H (e
where,
H (e
H (e
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) is the frequency response defined as
j n
h [ n ]e
Example:Consider a LTI sytem with

Then the frequency response is
H (e
) e
h[n ] [n no ]
j n o
j
X ( eFourier
)
For any input x[n] with Fourier transform

transform of output
Y (e
) e
j n o
X (e
The output is equal to the input with a time shift

y [ n ] x [ n n o ].
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Example:Consider a LTI sytem with |a|,|b| <1
h[ n ] a n u[ n ]
H (e
x[ n ] b n u [ n ]
1
1 ae
X (e
1
1 be
then
1
1
1 ae j 1 be j
y[n] is found taking inverse Fourier transform (partial
fraction):
Y (e
y[ n ]
1
[ a n 1 b n 1 ]u[ n ].
ab
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Multiplication (Modulation): Multiplication in time

corresponds to convolution in frequency domain:
x[ n ]h[ n ]
1
2
(e
) H (e
j ( )
)d
This is periodic convolution over 2 interval.
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Z- Transform :
For a discrete-time signal x[n], z-transform is defined as
X (z)
x[n ]z n
z is a general complex variable shown in polar form as

z r e j
We get Fourier transform
as a special case when r=1
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Convergence: Z-transform exists if
| x[ n ] r
Poles-Zeros: consider X(z) as a rational function

P (z)
X (z)
Q (z)
where P(z) and Q(z) are
polynomials in z. Values of z
making X(z)=0 are called zeros of
X(z) (roots of P(z)).Values of z
making X(z)= are called poles of
X(z) (roots of Q(z)).
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x[ n ] a nu [ n ]
Example: Consider
X (z)
a u[n ]z
( az
)n
n0
For convergence
| az
X (z)
1
1 az
z
, | z | | a |
z a
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Example: Consider
x [ n ] a n u [ n 1]
X (z) 1
(a
z)n
n0
For convergence
|a
z |
X (z) 1
1
1 a
z
, | z | | a |
z a
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Example: Consider
x [ n ] a n u [ n ] b n u [ n 1 ],
a nu[n ]
b a
z
, | z | | a |
z a
b n u [ n 1]
z
, | z | | b |
z b
Then
X (z)
z
z
, | a | | z | | b |
z a
z b
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Properties of convergence:
ROC is a ring or disk centered at origin.
DTFourier transform exists if ROC includes Unit Circle
ROC can not contain any poles
If x[n] has finite duration ROC is entire plane.
If x[n] is right-sided, ROC is outward from outermost pole
If x[n] is left-sided, ROC is inward from innermost pole
If x[n] is two-sided, ROC is a ring not containing any pole.
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Example:
x [ n ] ( 0 . 4 ) n u [ n ] ( 2 ) n u [ n 1 ],
X (z)
z
z
, | 0 . 4 | | z | | 2 |
z 0 .4
z 2
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Transfer Function: Z-transform of impulse response h[n]
H (z)
h[n ]z n
If impulse response h[n] is causal then ROC will be outward.

If the system is causal and stable, all the poles will be inside the
unit circle and ROC will include the unit circle.
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Inverse Z-Transform:
Power Series Expansion: Consider
X (z) z2
remember
1
1 1
z 1
z
2
2
X (z)
x[n ]z n
x [ 2 ] z 2 x [ 1 ] z 1 x [ 0 ] z 0 x [1 ] z 1
therefore
1
1
x[ n ] [ n 2 ]
[ n 1] [ n ] [ n 1]
2
2
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Example (Long Division): |z| > |a|

1
X (z)
1 az 1 a 2 z 2 a 3 z 3
1
1 az
therefore x [ n ] a n u [ n ]
Example (Long Division): |z| < |a|
z
X (z)
a 1 z a 2 z 2 a 3 z 3
a z
x[ n ] a n u [ n 1]
therefore
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Partial fraction expansion: Consider

M
X (z)
bk z k
a k z k
k 0
N
k0
k 1
1
b o k 1 (1 c k z )
N
ao
(1 d k z 1 )
k 1
Ak
, A k (1 d k z 1 ) X ( z ) | z d k
1
1 dkz
then
x[ n ]
A k d kn u [ n ]
k 1
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Example : Consider
1
1
X (z)
, | z |
.
1 1
1 1
2
(1
z )( 1
z )
4
2
A1
A2
X (z)
1 1
1 1
(1
z )
(1
z )
4
2
1 1
A 1 (1
z ) X ( z ) | z 1 / 4 1
4
1 1
A 2 (1
z ) X ( z ) | z 1 / 2 2
2
n
n
1
1
x[ n ] 2 u [ n ] u [ n ]
2
4
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Properties of Z-Transform:
Linearity:
x[n ] X ( z )
y[n ] Y ( z )
ax [ n ] by [ n ] a X ( z ) b Y ( z )
ROC R x R y
Time shift:
x[ n ] X ( z )
y [ n ] x[ n n 0 ] Y ( z ) z no X ( z )
ROC
Rx
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Multiplication by Exponential:
x[ n ] X ( z )
z on x [ n ] X ( z / z o )
ROC
| zo | R x
Conjugation:
x *[n ] X
ROC R x
(z*)
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Time reversal:
x * [ n ] X * (1 / z * )
ROC 1 / R x
x [ n ] X (1 / z )
ROC 1 / R x
Convolution:
y [ n ] * x [ n ] X ( z )Y ( z )
ROC R x R y
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Example: x [ n ] a n u [ n ]
h[n ] u[n ]
y[ n ] x[ n ] * h[ n ]
X (z)
z
, | z | | a |
z a
Y (z) X (z)H (z)

Y (z)
H (z)
z
, | z | | 1 |
z 1
z2
, | z | | 1 |
( z a )( z 1 )
1
1
a
1 a 1 z 1
1 az
y[n] is obtained as:

y[n ]
1
(u [n ] a
1 a
n 1
u [ n ])
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Sampling:
Consider a signal x[n] which is obtained by taking samples
of a continuous time signal xc(t):
x [ n ] x c ( nT )
where T is the sampling period, its reciprocal,
the sampling frequency.
fs 1
T is
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where x [ n ] Q ( x [ n ])
has the quantized samples and
xB[n] has the coded samples (such as 2s complement).
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Frequency Domain Representation of Sampling:

Consider the sampled signal xs(t) obtained from xc(t) by
multiplying with periodic impulse train:
s (t )
( t nT )
then
x s (t ) x c (t ) s (t ) x c (t )
( t nT )
The discrete time signal is x[n]= xc(nT). The

Fourier transform of s(t) is
2
S ( )
T
( k s ),
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Then, Fourier transform of xs(t) is

1
1
X s ( )
X ( ) * S ( )
2
T
X c ( k s )
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We observe that when the sampling frequency is not

chosen high enough, copies of spectrum overlap; this is
called aliasing (distortion).
The minimum sampling frequency (rate) to avoid
aliasing is 2N (Nyquist rate) :
2
2
T
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If sampling rate is greater than Nyquist rate, then the

original signal xc(t) may be obtained without distortion
using a low-pass filter.
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x s (t )
x c ( t ) ( t nT )
Fourier Tr. of xs(t) may be written as
x c ( nT ) e
X s ( )
j Tn
On the other hand DTFT of x[n] is

X ( e j ) x [ n ]e
j n
Comparing these equations we observe that

X s ( ) X ( e j ) | T X ( e j T )
X (e
X (e
j T
1
T
1
T
X c ( k s )
X c(
2 k
)
T
T
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Example:Consider
x c ( t ) cos o t
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x r ( t ) cos( o t )
x r ( t ) cos((
o )t )
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Example: Consider
x c ( t ) cos o t ,
4000
with sampling period T=1/6000, we obtain

2
x [ n ] cos 4000 nT cos

n
3
2
s
12000 2 o 8000
T
Nyquist sampling condition is satisfied , there is no
aliasing.
Example: Consider
x c ( t ) cos o t ,
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16000
with sampling period T=1/6000, we obtain

2
s
12000 2 o 16000
T
x [ n ] cos 16000 nT
cos
2 n
cos 16000 n / 6000
4000 n / 6000
cos
n
3
Nyquist sampling condition is not satisfied , there is

aliasing.
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Reconstruction from Samples:

Use ideal LPF to recover original signal
With
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x s (t )
x c ( nT ) ( t nT )
x [ n ] ( t nT )
x r (t )
x [ n ] h ( t nT )
h ( t ) sinc
x r (t )
t T
t nT
x [ n ] sinc
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Discrete-time Processing of Cont.-time Signals:
h[n ]
with h(t) is impulse response of continuous-time system

h [ n ] T h c ( nT )
H (e
) H c(
)
T
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Changing the Sampling Rate:

Downsampling and Decimation:
Sampling rate can be reduced by further sampling in
discrete-time:
x d [ n ] x [ nM ] x c ( nMT )
Let
T ' MT
x d [ n ] x c ( nT ' )
Then
If X c ( ) 0 ,
for
and
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2
2
T '
MT
2
2 N
(N is BW of x[n])
M
Then xc(t) can be recovered without distortion.
or
Remember that
X (e
Similarly
X
X
(e
(e
)
j
1
T
X c(
2 k
)
T
T
2k
X c(
T ' k
T '
T '
2 k
)
X c(
MT k
MT
MT
)
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Let
n 0 ,
x [ n ],
x1[ n ]
otherwise
0,
1 M 1 j 2 l n / M
x1[ n ] x [ n ]
e
M l0
M , 2 M ,
, n
( [ ]=Discrete Fourier Series rep. of impulse train with period M )
x d [ n ] x 1 [ Mn ] x [ Mn ]
(e
xd [n] e
j n
x 1 [ Mn ] e
j n
k Mn n k / M
(e
j k / M
x1[ k ] e
(e
1
M
M 1
l0
M 1
1
x[n ]
M
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j 2 lk / M
l0
x [ k ]e
k
j 2 lk / M
j k / M
j k / M
Combining the exponential terms

X
(e
1
)
M
First obtain X ( e
M 1
X (e
j(
2l
)
M
M
l0
2 increments to get
)then shift by
X (e
j ( ' 2 l / M )
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If 2 2
M
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A general system for downsampling called decimator is
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Upsampling and Interpolation:

Sampling rate can be increased by obtaining intermediate
values:
x i [ n ] x c ( nT ' ), T ' T / L , L 2
given x [ n ] x c ( nT ).
x i [ n ] x [ n / L ] x c ( nT / L ),
n 0 , L , 2 L ,
Consider the following system
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The block with L inserts (L-1) zeros between samples:

x n / L , n 0 , L , 2 L
xe[n]
0,
otherwise
or
xe[n ]
x k n
kL .
In frequency domain:
X e (e
x k n
kL e
j n
X e (e
x k e
j kL
X (e
j L
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L=2
h n
sin n / L
n / L
xi[n]
x[ k ]
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sin ( n kL ) / L
( n kL ) / L
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Changing the sampling rate by noninteger factor:

Following system may be used
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51
End of Part 1
A.S.Kayhan
52
DIGITAL
SIGNAL
PROCESSING
Part 2
A.S.Kayhan
Frequency Analysis of LTI Systems:

the Input/Output (I/O) relation of a LTI system is given
by convolution:
y[ n ] x[ n ] * h[ n ]
h[ k ] x[n k ]
In z-domain
Y (z) X (z)H (z)
On the unit circle

Y (e
where H ( e
system.
) X (e
) H (e
ze
is the frequency response of the
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Magnitude is
Y (e
where H ( e
system.
) X (e
) H (e
is the magnitude response of the
Phase is Y ( e
where H ( e
system.
) X (e
) H (e
) is the phase response of the
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Discrete-time ideal filters: LPF, HPF, BPF:
These are not realizable. Why?

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Phase distortion and delay:

Assume that h id [ n ] [ n n d ]
then
or
H
H
id
id
(e
(e
j n d
) e
) 1,
id
(e
) nd
Linear phase distortion causes simple delay.

Group delay:
It measures linearity of the phase response.
Consider
x [ n ] s [ n ] cos o n
where s [ n ] is a narrowband (slowly varying) signal.
Assume around
H (e
H (e
) 1,
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) o n d
Then
y [ n ] s [ n n d ] cos o n o o n d
Thus, group delay of a system is

grd [ H ( e
where arg H ( e
)]
d
d
arg H
(e
is the continuous phase.
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Example: Consider the following filter
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Input is the following signal
Pulses are at
0 . 25 ,
0 .5 ,
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0 . 85
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Systems with Difference Equation Models:

Assume that the Input/Output (I/O) relation of a system
is given by a constant coefficient difference equation:
N
k 0
a k y[n k ] bk x[n k ]
k 0
Applying z-transform, using linearity and shifting

properties, we obtain
N
k 0
k 0
ak z
Y ( z ) bk z k X ( z )
k 0
M
a k z k Y ( z )
k 0
bk z k X ( z )
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Then, the transfer function (or system function) is

M
bk z k
Y (z)
H (z)
k N 0
X (z)
a k z k
k 0
We can write the transfer function in terms of its

poles, dk, and zeros, ck, as M
H (z)
bo
ao
(1 c k z 1 )
(1 d k z 1 )
k 1
N
k 1
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Example: Consider system function

(1 z 1 ) 2
H (z)
1 1
3 1
z )( 1
z )
2
4
1 2 z 1 z 2
Y (z)
H (z)
1 1
3 2
X (z)
1
z
z
4
8
(1
then
( 1 2 z 1 z 2 ) X ( z ) (1
1 1
3 2
z
z )Y ( z )
4
8
x[ n ] 2 x[ n 1] x[ n 2 ]
1
3
y[n ]
y [ n 1]
y[n 2]
4
8
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Stability and Causality:

A system is causal if ROC for the transfer function H(z) is
outward.
A sytem is stable if ROC for the transfer function H(z)
includes the unit circle.
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Inverse Systems :
Let H i ( z ) be the inverse system of H ( z ), then
G (z) H i(z)H (z) 1
Eq.(1)
g n h i n * h n n
then
H i(z)
1
,
H (z)
H i (e
1
H (e
Not all systems have an inverse. Ideal LPF does not have
an inverse, we can not recover high frequency components.
M
Now, consider
1
b
H (z) o
ao
(1 c k z
(1 d k z 1 )
k 1
N
k 1
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Then the inverse system has

N
H i(z)
ao
bo
(1 d k z 1 )
(1 c k z 1 )
k 1
M
k 1
Poles become zeros of the inverse system, zeros become

poles.
For Eq.(1) to hold, ROC of H i ( z ) and H ( z ) must
overlap. If H ( z ) is causal, ROC is
z max
dk
Some part of ROC of H i ( z ) must overlap with this.

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Example: Suppose
z 1 0 .5
H (z)
,
1 0 .9 z 1
z 0 .9
The inverse system is

1 0 .9 z 1
2 1 .8 z 1
H i(z)
z 1 0 .5
1 2 z 1
Two possible ROC:

z 2
h i1 n 2 n 1 u n 1 1 . 8 2 n 1 u n
which is stable but noncausal.

z 2
h i 2 n 2 n 1 u n 1 . 8 2 n 1 u n 1
unstable but causal.

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Example: Suppose
1 0 .5 z 1
,
1 0 . 9 z 1
H (z)
z 0 .9
The inverse system is

H i(z)
1 0 .9 z 1
1 0 .5 z 1
The only choice for ROC is overlaps with z 0 . 9 is

z 0 .5
then
h i n 0 . 5 u n 0 . 9
n
0 . 5 n 1 u n
which is both stable and causal.

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Impulse Response for Rational System:

Assume that a stable LTI Msystem has a rational transfer
function.
(1 c k z 1 )
bo
ao
H (z)
k 1
N
(1 d k z 1 )
k 1
M N
B r z r
H (z)
r0
0 if M N
then
h n
M N
k 1
B r n r
r0
Ak
1 d k z 1
A k d kn u n 1
k 1
Impulse response is of infinite length, called Infinite

Impulse Response (IIR) system.
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If the system has no poles, then

M
H (z)
bk z k
k 0
h n
b k n k
k 1
Impulse response is of finite length, called Finite Impulse

Response (FIR) system.
Example: Consider a causal system with
y [ n ] ay [ n 1 ] x [ n ]
If a 1 then stable and the impulse
response is
h n a n u n .
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10
Example: Consider
a n , 0 n M
h n
.
0
,
otherwise
Then
M
1 a M 1 z M 1
H ( z ) a n z n
1 az 1
n0
Zeros at
z k ae
2k
M 1
k 0 ,1 , , M .
Difference equation is
y n
a k x n k
k 1
However, we can also write

y [ n ] ay [ n 1 ] x [ n ] a
M 1
x n M 1
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Frequency Response for Rational System Functions:

Assume that a stable LTI system has a rational transfer
function. Then frequency response is obtained by
evaluating it on the unit circle:
H (e
M
b k e j k
) k N 0
a k e j k
k 0
H (e
b
) o
ao
(1 c k e
(1 d k e
k 1
N
k 1
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11
The magnitude response of the system is

M
H (e
bo
ao
1 cke
1 d ke
k 1
N
k 1
The magnitude-squared response of the system is

H (e
H (e
)H
(e
H (e
b
o
ao
1 c
1 c
*
k
d ke
1 d
*
k
k 1
N
e
e
k 1
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Log magnitude or Gain in decibels(dB) :

20 log
10
H (e
) 20 log
10
bo
ao
20 log
10
1 cke
k 1
20 log
10
1 dke
k 1
Note that :
20 log
10
Y (e
) 20 log
20 log
10
10
H (e
X (e
j
j
Attenuation in dB = - Gain in dB
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12
Phase response for rational system function:

H (e
b
) o
ao
1 cke
k 1
N
1 d ke
k 1
Group delay for rational system function:
grd H ( e
k 1
d
d
N
k 1
arg[
d
d
1 cke
arg[
1 d ke
A.S.Kayhan
Frequency Response of A single Zero:

Consider transfer function of a system as
1
H ( z ) 1 az
then with a re
H (e
) H (e
)e
j H ( e
1 re
then magnitude is
H (e
1 r
2 r cos( )
and magnitude in dB is
20 Log
10
H (e
) 20 Log
10
[1 r
2 r cos( )]
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13
then phase is
H (e
) tan
r sin( )
1 r cos( )
group delay is derivative of the (unwrapped) phase

function
grd [ H ( e
d ( H (e
)]
d
))
Example: Consider two cases : r=0.9, =0 and r=0.9, =:
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14
magnitudes of
vectors give the
magnitude response
H (z)
z a
z
H (e
ze
re
e j
v3
v1
v3
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phases of vectors give the phase response

H (e
e j re j
)
e j
v 3 v1 3 1 3
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15
Example: Consider a second order system with

H (z)
1 re
1
j
e j z 1
z 1 1 re j z 1
H (e
v3
v1 v 2
H (e
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) H (z)
ze
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16
Example: Consider a third order system with
0 . 05634 1 z 1 1 1 . 0166 z 1 z 2
H (z)
1 0 . 683 z 1 1 1 . 4461 z 1 0 . 7957 z 2
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Relation Between Magnitude and Phase:

In general, knowledge about magnitude does not provide
information about phase, and vice versa.
But, for rational system functions, with some additional
information such as number of poles and zeros, magnitude
and phase responses provide information about each other.
2
Consider
j
j
*
j
H (e
H (e
)H
H (z)H
with
(e
(1 / z * ) | z e j
H (z)
bo
ao
(1 c k z 1 )
(1 d k z 1 )
k 1
N
k 1
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17
(1 / z * )
bo
ao
(1 c k* z )
(1 d k* z )
k 1
N
k 1
Then
b
C ( z ) H ( z ) H * (1 / z * ) o
ao
(1 c k z 1 )( 1 c k* z )
(1 d k z 1 )( 1 d k* z )
k 1
N
k 1
Notice that poles and zeros of C(z) occur in conjugate

reciprocal pairs. If one pole/zero is inside the unit circle
there is another outside.
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If H(z) is causal and stable, then all poles must be inside the
unit circle, with this we can identify the poles. But zeros of
H(z) can not be uniquely identified from zeros of C(z) with
this constraint alone.
Example: Consider two stable systems with
H 1(z)
2 1 z 1 1 0 .5 z 1
1 0 . 8 e j / 4 z 1 1 0 . 8 e j
/4
z 1
j / 4
z 1
and
H 2(z)
1 0 . 8 e
z 1 1 2 z 1
/4
z 1 1 0 . 8 e
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18
Pole/zero plots are
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C 1 ( z ) H 1 ( z ) H 1* (1 / z * )
2 1 z 1 1 0 . 5 z 1 2 1 z 1 0 . 5 z
1 0 .8 e j / 4 z 1 1 0 . 8 e j / 4 z 1 1 0 . 8 e j / 4 z 1 0 . 8 e j / 4 z
C 2 ( z ) H 2 ( z ) H 2* (1 / z * )
1 z 1 2 z 1 z 1 2 z
1 0 . 8 e z 1 0 . 8 e z 1 0 . 8 e
1
1 0 .8 e
j / 4
z 1
j / 4
j / 4
j / 4
Observe that (with 4 2 z 2 z 1 )
4 1 0 . 5 z 1 1 0 . 5 z 1 2 z 1 1 2 z
then
C1(z) C 2 (z)
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19
Example: Consider pole/zero plot of C(z) for a system,

determine H(z).
For a causal
and stable
system, poles
of H(z) are: p1,
p2,p3.
For real ak, bk,
zeros/poles
occur in
complex
conjugate pairs.
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All-Pass Systems:
Consider following stable system function
ap
then
(e
ap
z 1 a *
H ap ( z )
1 az 1
j *
e j a *
)
j ( 1 ae
)
e
j
j
1 ae
1 ae
(e
) 1
, but H
ap
(e
) constant
This is called an all-pass system .

A general all-pass system has the following form
M
ap
(z) A
k 1
z 1 d k
1 d k z 1
k 1
( z 1 e k* )( z 1 e k )
(1 e k z 1 )( 1 e k* z 1 )
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20
Example: Consider pole/zero plot of a typical all-pass

system
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Example: First order all-pass system with a real pole:

z=0.9 (z=-0.9)
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21
Example: Second order all-pass system with poles:

z 0 .9 e
j / 4
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Notice that group delay for causal all-pass systems are

positive (unwrapped/continuous phase is nonpositive).
All-pass systems may be used as phase compensators.
They are also useful in transforming lowpass filters into
other frequency-selective forms.
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22
Block-Diagram Representation:
LTI systems with difference equation represetation
(rational system function) may be imlemented by
converting to an algorithm or structure that can be realized
in desired technology. These structures consists of basic
operations of addition, multiplication by a constant and
delay.In block diagram representation:
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Example: Consider the second order system :

y [ n ] a 1 y [ n 1] a 2 y [ n 2 ] b o x [ n ]
with
H (z)
bo
1 a1 z a 2 z 2
1
Block diagram representation is
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23
For a system with higher order difference equation

N
y[n ]
a k y[n k ] bk x[n k ]
k 1
k 0
M
with system function
H (z)
k 0
N
k 1
ak z k
k 1
Rewriting the equation

as
N
y[n ]
bk z k
a k y[ n k ] bk x[ n k ]
k0
a k y[n k ] v[ n ]
k 1
M
where
v[ n ]
bk x[n k ]
k 0
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Direct Form I:
N+M
Delay
element
24
Previous diagram is implementation of
H (z) H 1(z)H 2(z)
1
N
k 1
or
V (z) H 1(z)X (z)
ak z k
k 0
Y ( z ) H 2 ( z )V ( z )
k 0
bk z k
bk z k X ( z )
1
N
ak z k
k 1
V ( z )
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We can rearrange the system function
W (z) H 2(z)X (z)
Y ( z ) H 1 ( z )W ( z )
In the time domain
k 0
1
N
ak z k
k 1
X (z)
b k z k W ( z )
w[n ]
a k w [n k ] x[ n ]
bk w[n k ]
k 1
M
y[n ]
k0
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25
M=N
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Direct Form II:
Max(N,M)
Delay elements
26
Example:Consider the system with

1 2 z 1
H (z)
1 1 .5 z 1 0 .9 z 2
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Flow Graph Representation:

Similar to the block diagram representation:
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27
W1(z) W 4 (z) X (z)
W 2 (z) W1(z)
W 3(z) W 2(z) X (z)
W 4 ( z ) z 1W 3 ( z )
Y (z) W 2(z) W 4(z)
z 1
Y ( z )
1
1z
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X ( z )
z 1
H ( z )
1
1z
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28
Structures for IIR Systems:

Some considerations:
Computaional complexity(no. of multiplication, delay )
Finite precision, Ease of implementation, etc.
Direct Forms:
We have already seen direct forms.
Cascade Form:
We factor numerator and denominator polynomials of H(z)
M
H (z) A
(1 f k z 1 ) (1 g k z 1 )( 1 g k* z 1 )
(1 c k z 1 ) (1 d k z 1 )( 1 d k* z 1 )
k 1
N1
k 1
k 1
N2
k 1
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We have cascade of first or second order subsystems

N
H (z)
H k (z)
k 1
k 1
bo
k 1
b ok b 1 k z 1 b 2 k z 2
1 a 1 k z 1 a 2 k z 2
~
1 b 1 k z 1 b 2 k z 2
1 a 1k z 1 a 2 k z 2
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29

1 2 z 1 z 2
1 0 . 75 z 1 0 . 125 z 2
1 z 1 1 z 1
1 0 . 5 z 1 1 0 . 25 z 1
H (z)
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Parallel Form:
H(z) may be written as sum of subsystems
N
H (z)
Ckz
k0
k 0
N
H (z)
Nc
Ak
B k (1 e k z 1 )
1 c k z k k 0 (1 d k z 1 )( 1 d k* z 1 )
N
k0
Ckz
k 0
e ok e 1 k z 1
1 a1 k z 1 a 2 k z 2
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30
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1 2 z 1 z 2
18
25
H (z)
8
1
2
1
1 0.75 z 0.125 z
1 0 .5 z
1 0.25 z 1
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31
Transposed Forms :
Reverse the directions of all branches while keeping the
values as they were and reverse roles of input and output.
Transposed forms may be useful in finite precision
implementation.
Example:Consider the second order system with
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Structures for FIR Systems:

Consider an FIR system with following input-output
M
relation
y[n ]
bk x[ n k ]
k0
Observe that the impulse response of this system is

b , 0 n M
h[ n ] n
0 , otherwise
Direct form (or tapped delay line or transversal filter) :
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32
Transposed direct form structure:
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Cascade form:
N
H (z)
k 1
H k (z)
( b ok b 1 k z 1 b 2 k z 2 )
k 1
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33
Linear-Phase FIR Systems:

Finite impulse response (FIR) systems with linear-phase
have symmetry properties such as
h [ n ] h [ M n ], 0 n M
or
h [ n ] h [ M n ], 0 n M
For M even and odd.

Thus there are four types of linear phase FIR filters.
A.S.Kayhan
Example: Consider
1, 0 n 4
h[ n ]
0 , otherwise
sin( 5 / 2 )
H ( e j ) e j 2
sin( / 2 )
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34
Example: Consider
h [ n ] [ n ] 2 [ n 1] [ n 2 ]
H (e
) 1 2e
j 1
j 1
j 1
j 1
2 e
e
j 1
j 2
2 [1 cos( )].
A.S.Kayhan
Assume M is even
M
y[n ]
h [ k ]x [ n k ]
k 0
M / 2 1
h[ k ] x[ n k ] h[ M / 2 ] x[ n M / 2 ]
k0
M
h[k ] x[n k ]
k M / 2 1
With, in the last term, k=M-l, l=0M/2-1

M / 2 1
y[n ]
h[ k ] x[ n k ] h[ M / 2 ] x[ n M / 2 ]
k 0
M / 2 1
h[ M l ] x[ n M l ]
l0
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35
If
h[ n ] h[ M n ]
M / 2 1
y[n ]
h [ k ]( x [ n k ] x [ n M k ])
k 0
h[ M / 2 ]x[ n M / 2 ]
If
h[ n ] h[ M n ]
M / 2 1
y[n ]
h [ k ]( x [ n k ] x [ n M k ])
k 0
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Finite-Precision Numerical Effects:

Input Quantization:
We have seen earlier that continuous-time signals are
sampled, quantized and coded first
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36
Example:
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In twos complement binary system leftmost bit is the sign

bit. If we have (B+1)-bit twos complement fraction of the
following form
a 0 a1a 2 a B
Value is
a 0 2 0 a1 2 1 a 2 2 2 a B 2 B
Example:
Binary Code
Numeric value
0 11
3/4
1 10
1/2
1 01
1/4
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37
Quantizer step size is

With
where
2X m
2 B 1
x[ n ] X
x B [n ]
1 x B[n] 1
(two' s complement
Analysis of Quantization Errors:

We observe that the quantized samples are in general
different from the true values. The difference is the
quantization error
^
e [ n ] x [ n ] x [ n ].
and
/ 2 e[ n ] / 2
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A simplified model of quantizer is
Assumptions about e[n]:

e[n] is stationary
e[n] is uncorrelated with x[n]
e[n] is white noise
e[n] is uniformly distributed
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38
Example:
Sinusoidal
signal
3bit
quantizer
Error for
3bit
Error for
8bit
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Mean value of e[n] is zero

/2
1
de 0
/2
Variance (or power) of e[n] is

/2
2
e
/2
1
2
de
12
For (B+1)-bit quantizer with full-scale value Xm
2
e
2 2 B X
2
m
12
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39
Signal-to-Noise Ratio (SNR) is defined as the ratio of

signal variance (power) to noise variance. Expressed in dB
x2
12 2 2 B x2
SNR 10 log 10 2 10 log 10

X m2
e
6 . 02 B 10 . 8 20 log
10
m
x
If
B B 1,
x x / 2,
SNR SNR 6 dB
SNR SNR - 6dB
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Coefficient Quantization in IIR Systems:

M
Consider
bk z k
H (z)
k 0
N
ak z k
k 1
If the coefficients are quantized,

we get
M
^
b k z k
H (z)
k 0
N
a k z k
k 1
where
b k bk bk , a
ak ak
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40
Each pole or zero will be affected by all the errors in the

coefficient quantization.
If the poles (or zeros) are close to each other (clustered),
then quantization of coefficients may cause large shifts
of poles (or zeros).
Direct form structures are more sensitive to coefficient
quantization than the other forms (parallel, cascade,...)
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Example: Consider a bandpass IIR elliptic filter of order 12

implemented in cascade form of 2nd order subsystems and
direct form.
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41
Passband
cascade
unquantized
Passband
cascade
16-bit
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A.S.Kayhan
Passband
parallel
16-bit
Direct form
16-bit
42
Poles of Quantized 2nd order subsystems :

Because of robustness, parallel and cascade forms are used
more than direct forms.
We can further improve the robustness, by improving
implementation of the 2nd order subsystems.
Consider the following implementation in direct form:
A.S.Kayhan
When coefficients are quantized, a finite number of pole

possitions possible.
7bits
4bits
Circles correspond to r2,
vertical lines to 2rcos
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43
Consider the following coupled form
4bits
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7bits
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44
Coefficient Quantization in FIR Systems:

Consider FIR system with transfer function
M
H (z)
h[n ]z n
n0
If
h[ n ] h[ n ] h[n ]
then
H (z)
h[n ]z n H ( z ) H ( z )
n0
where
and
H (z)
h[n ]z n
n0
^
H (e
) H (e
) H (e
)
A.S.Kayhan
Example: Consider FIR filter of order 27
A.S.Kayhan
45
A.S.Kayhan
Effects of Round-off Noise:

Analysis of Direct Form IIR Structures:
Consider Nth-order difference equation for Direct Form I
N
y[n ]
k 1
a k y[ n k ] bk x[ n k ]
k0
Assume that all signal values and coefficients are

represented by (B+1)-bit fixed point binary numbers.
Therefore, each multiplication is followed by a quantizer,
then
^
y[n ]
k 1
M
^
Q a k y [ n k ] Q b k x [ n k ]
k 0
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46
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An alternative representation is as following
A.S.Kayhan
47
Rounding or truncation of a product bx[n] is represented by

a noise source of the form
e [ n ] Q bx [ n ] bx [ n ]
Assumptions about e[n]s:

Each e[n] is stationary
Each e[n] is uncorrelated with x[n] and other e[n]s
Each e[n] is uniformly distributed
For (B+1)-bit rounding
2 B / 2 e[ n ] 2 B / 2
A.S.Kayhan
For (B+1)-bit truncation

2 B e[ n ] 0
Mean and variance for rounding are
e 0
2 2 B
12
2
e
Mean and variance for truncation are

e
2B
2
2
e
2 2 B
12
A.S.Kayhan
48
Now, lets try to determine effect of quantization noise on

the output of the system. We can redraw the system as
e[ n ] e 0 [ n ] e1[ n ] e 2 [ n ] e 3 [ n ] e 4 [ n ]
A.S.Kayhan
Since all the noise sources are

2
e
2
e0
2
e1
2
e2
2
e3
2
e4
5
For the general Direct Form I case
2 2 B
2
e M 1 N
12
Now, we observe that
2
e0
2 2 B
5
12
f [n ]
a k f [ n k ] e[ n ]
k 1
For rounding mean of the output noise is zero.

The variance for rounding or truncation is
2
f
2 2 B
M 1 N
12
h ef [ n ] is impulse response for
h ef [ n ]
ef
(z) 1 / A(z)
A.S.Kayhan
49
Example: Consider the following system

H (z)
b
1 az
a 1.
h ef [ n ] a n u [ n ]
Then the noise variance(power) at

the output is
2
f
2 2 B
2
12
2n
2 2 B
2
12
1 a
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Analysis of Direct Form FIR systems:

Consider
y[n ]
h[ k ]x[ n k ]
k 0
f [ n ] e[ n ]
k 0
e k [ n ],
2
f
2 2 B
M 1
12
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50
End of Part 2
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51
DIGITAL
SIGNAL
PROCESSING
Part 3
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IIR Filters:
Infinite impulse response (IIR) filters have rational transfer
M
functions as
k
H (z)
bk z
ak z k
k 0
N
k 0
Some IIR filter types are: Butterworth, Chebyshev,

Elliptical.
IIR filters have to be stable.
IIR filters may not have linear-phase.
A.S.Kayhan
Filter Specifications (Low-pass):
20 log
10
( )
Design analog filter transform to Digital filter
A.S.Kayhan
Analog Butterworth Filters:

Approximate ideal LPF with magnitude-squared response
H ( )
K ,

0,
c
c
by the following rational function:

H ( )
K 1 a 1 2 a n 1 2 n 2
,
1 b1 2 b n 2 n
bn 0
|H()|2 must be even function of , and denominator

degree must be higher than degree of numerator (lowpass).
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For maximal flatness at the origin, =0, The first 2n-1

derivatives of |H()|2 must be zero. This requires
i 1, 2 , , n 1 .
a i bi ,
H ( )
K 1 b1 2 b n 1 2 n 2
,
1 b1 2 b n 2 n
bn 0
Maximal flatness also at , =, requires

i 1,2 , , n 1 .
a i bi 0,
Then, we have
2
H ( )
K
1 bn
2n
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Let |H(=0)|2 = 1. Then

H ( )
1
1 bn
2n
At the half-power frequency

H ( )
1
1
2
1 bn
2n
o
bn
2n
Then, we have the Butterworth response

H ( )
1

1
o
2n
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Consider the filter specifications:

20 log
10 log
10
/ 10
10
10
H ( ) dB .
2n
2n
dB .
To find n:
Using max and p
10
max
/ 10
10
/ 10
1/ 2n
Using min and s
p
o
2n
10
min
/ 10
s
o
2n
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Dividing these equations
10
10
min
max
/ 10
/ 10

1

1
s
p
2n
Taking logarithm, we get the filter order as:
10 min / 10 1
log
10 max / 10 1
2 log s
p
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We can normalize the frequency so that o = 1. Then

2
H ( )
1
1
2n
To find the filter poles, we let =s/j,

H (s) H ( s)
1
1
2n
1 (s / j)
1 ( 1) n s 2 n
then, set the denominator to zero

1 ( 1) n s 2 n 0
( 1) n s 2 n 1
If n is even, then
s 2n 1 e
sk e
1 2 k
j (
)
2n
j ( 2 k )
, k 0 ,1 ,..., 2 n 1 .
, k 0 ,1 ,..., 2 n 1 .
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If n is odd, then
s 2n 1 e
sk e
k
n
j 2k
, k 0 ,1 ,..., 2 n 1 .
, k 0 ,1 ,..., 2 n 1 .
Note that all the poles lie on a circle with radius 1, because
the frequency is normalized. (If we want to design analog
filter we need to correct this).
Also, choose the poles in the left-half plane to get a stable
filter.
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Example: Design specs. for a Butterworth LP filter:

At f = 2000 Hz, max= 3dB
At f = 3000 Hz, min= 10dB
Then, the filter order n is
n 2.7154 n = 3.
Poles are at
sk e
k
3
, k 0 ,1 ,..., 5 .
1
( s 1 )( s 2 s 1 )
1
.
1
3
1
3
( s 1 )( s ( j
))( s ( j
))
2
2
2
2
H (s)
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Analog Chebyshev Filters:

We can generalize Butterworth response as:
H ( )
1
1
.
n
1 F n ( ) 2
1 ( ) 2
Fn() is a function of . Now, consider:

x cos( ) cos 1 ( x )
y n cos( n ) cos( n cos
( x ))
This is the Chebyshev polynomial of order n. When

x 1 , C n ( x ) cosh( n cosh
( x ))
When x 1 , Cn(x) 0.
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Cn(x) has zeros in -1< x < 1 :

C1(x)
C2(x)
C3(x)
C4(x)
C5(x)
-1
-1
0
x
-1
1 -1
n 1
0
x
-1
-1
0
x
-1
1 -1
(x) 2 x C n (x) C
-1
1 -1
0
x
n 1
0
x
( x ),
C o ( x ) 1, C 1 ( x ) x .
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Use Cn(.) in :
H ( )
1
1 2 C n2 ( )
1 , C n ( ) cosh( n cosh
1 , C n ( ) cos( n cos
2 1 ,
( ))
( ))
is known as the ripple factor.
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Behaviour at =0:
C n2 ( ) 0 , H ( 0 )
C n2 ( ) 1 , H ( 0 )
1,
if n is odd
1
,
(1 2 )
if n is even
Behaviour at =1:
C n2 ( ) 1 ,
for all n
H ( 1)
1
(1 2 )
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The attenuation is
10 log 1 2 C n2 ( )
10
max/ 10
When 2 C n2 ( ) 1 3 . 01 dB .
This defines the half-power frequency (3dB cut-off) hp.
Then,
1
cosh( n cosh
1
1
cosh 1 ( hp )
cosh 1 ( )
n
1
1
hp cosh(
cosh 1 ( )).
n
C n (
hp
hp
))
hp
1 ).
A.S.Kayhan
The specifications for a Chebyshev filter are max, min, s

(p=1). Bandpass is between 0 1 rad/s.
To find n:
min 10 log 1 2 C n2 ( s )
10 min
With
cosh( n cosh
Finally,
cosh
n
/ 10
10
1
1 2 C n2 ( s )
max/ 10
( s ))
10
/ 10
min
cosh
10
10
1
10
min
/ 10
max
/ 10
max
1
1
/ 10
( s )
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To find the filter poles, we let =s/j,

H (s) H ( s)
then
1
1 C n2 ( s / j )
2
1 2 C n2 ( s / j ) 0 C n ( s / j ) j
Let
With
cos( x )
cos( w ) cos( u jv ) s / j
cos 1 ( s / j ) w
e
jx
e
2
jx
, cos( jx )
C n ( s / j ) cos( n cos
ex ex
cosh( x )
2
( s / j )) cos( nw )
C n ( s / j ) cos( nu ) cosh( nv ) j sin( nu ) sinh( nv )

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With
cos( nu ) cosh( nv ) 0
C n (s / j) j
then
sin( nu ) sinh( nv )
cosh(nv) can never be zero, therefore cos(nu) must be

zero. It is possible for
3 5
,
,
,
2n 2n 2n
( 2 k 1 ), k 0 ,1 , , 2 n 1 .
2n
uk
uk
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For these values of u, sin(nu) = 1, then

vk
Remember that
then
1
sinh
n
cos
1
) a.
( s / j ) w u jv

2 k 1 ja
s k j cos( w k ) j cos
2n
The poles are :

s k k j k , k 0 ,1 , , 2 n 1
2k 1
) sinh( a )
2n
2k 1
cos(
) cosh( a )
2n
sin(
Choose left
half poles.
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10
Discrete Time Filter Design(IIR):

There are 3 approches:
1-Sampling (Impulse invariance)
2-Bilinear transformation
3-Optimal procedures
Impulse invariance method:

We can sample the impulse response hc(t) of an analog filter
with desired specifications as (Td is sampling interval):
then
h [ n ] T d h c nT
H ( )
H c(
k ).
Td
Td
If
then
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/ Td
H c ( ) 0 ,
),
.
Td
Analog and digital frequencies have a linear relation:
Td .
H ( ) H c (
Consider the transfer function of a system expressed as

N
H c (s)
k 1
Ak
,
s sk
Then, the impulse response is

N
hc (t )
Ak e skt ,
t 0.
k 1
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11
The impulse response of the discrete time filter is

h n T d h c ( nT
N
d
T d A k e s k nT d u n
k 1
h n
T d ( A k e s k T d ) n u n .
k 1
Transfer (or system) function of the discrete time filter is

N
H (z)
k 1
Poles are
Td Ak
.
1 e sk T d z 1
s s k z e skTd .
If Hc(s) is stable (k < 0), then H(z) is also stable (|z| < 1).
A.S.Kayhan
Example: Transfer function of analog filter is

1
( s 1 )( s 2 s 1 )
1
1
1
1
j
j
1
2
2
12
12
.
s 1
1
3
1
3
s ( j
)
s ( j
)
2
2
2
2
H c (s)
then
Td
1 z 1e Td
1
1
1
Td ( j
)
Td ( j
2
2
12
H (z)
1 z 1 e
Td (
1
3
j
)
2
2
1 z 1e
Td (
1
)
12
.
1
3
2
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12
Bilinear Transformation:
Transformation needed to convert an analog filter to a
discrete time filter must have following properties:
1- j axis of the s-plane must be mapped onto the unit
circle of the z-plane,
2- stable analog filters must be tranformed into stable
discrete time filters (left hand plane of the s-plane must be
mapped into inside the unit circle of the z-plane).
Following BT satisfies these conditions:
1 z 1
s K
1 z 1
1 s / K
.
1 s / K
Let
s j ,
z re
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then
r
(1 / K ) 2 ( / K ) 2
.
(1 / K ) 2 ( / K ) 2
Therefore
0
( LH plane ) r 1 ( inside
( j axis ) r 1 ( the u.c. )
( RH plane ) r 1 ( outside
the u.c. )
the u.c. )
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13
Now, let
then
s j ,
z e
1 e j
e j / 2 ( e j / 2 e
j K
K j / 2
1 e j
e
( e j / 2 e
sin( / 2 )
j Kj
jK tan( / 2 ).
cos( / 2 )
or
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j / 2
j / 2
)
.
)
K tan( / 2 ).
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14
Observations:
1- According to Taylor series expansion of tan(), for small
K / 2 3 / 24 K / 2 .
2- For high frequencies, the relation is nonlinear causing a
distortion called warping effect. Therefore, BT is usually used
for the design of LPF to avoid this.
A.S.Kayhan
Discrete Butterworth Filter Design:

We convert a frequency normalized analog Butterworth filter
to a discrete filter using the BT.
Normalized half power frequecy HP=1 is mapped into the
discrete half power frequency HP. To do that, we set K to
K b ( HP 1) / tan(HP / 2) cot( HP / 2).
A.S.Kayhan
15
Then, we use
K b tan(0.5 ) tan(0.5 ) / tan(0.5HP ).
in
H
( )
1
1
2n
And get the discrete Butterworth Low Pass Filter as:

H
( )
tan( 0 . 5 )
1
tan( 0 . 5 HP )
2n
Using the design specifications, we find the filter order as
min / 10
1
log 10
max / 10
10
2 log tan( 0 . 5 s )
tan( 0 . 5 p )
A.S.Kayhan
The half-power freq. is
HP
2 tan
tan( 0 . 5
10
max
/ 10
1/ 2n
K b cot(0.5 HP ).
Example: Consider the second order analog filter

H
(s)
1
(s2
2 s 1)
Applying the BT, we get (Kb = 1)

( z 1) 2
H 2(z)
0 . 2929
( z 2 0 . 1716 )
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16
Remarks:
1- Given the specs.:
a) find the filter order n
b) obtain the analog filter H(s)(using LHP poles)
c) calculate HP and Kb
d) use biliear transformation to find H(z).
2- H(z) is BIBO stable, because H(s) is stable.
3- Applying the BT to high order filters may be tedious.
Therefore, first express H(s) as product or some of first
and second order functions, then apply the BT.
A.S.Kayhan
Example: Design specs. for a LP digital Butterworth filter:

At f = 0 Hz, 1= 18dB
At f = 2250 Hz, 2= 21dB
At f = 2500 Hz, 3= 27dB
Sampling freq. fsmp= 9000 Hz
(ej0)= 18dB
fp
max = 2 1=3dB
fs
p 2
, s 2
min = 3 1=9dB
f smp
f smp
Filter order n 5.52,n = 6, Kb = 1 and a gain G for = 18dB
H 6(z)
(z2
0 . 0037 ( z 1 ) 6
0 . 59 )( z 2 0 . 17 )( z 2 0 . 0 . 02 )
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17
Discrete Chebyshev Filter Design:

We find the constant K by transforming (normalized
passband freq.) P=1 into the discrete passband frequency
P. We get
K c 1 / tan(0.5P ) tan(0.5 ) / tan(0.5 P )
And get the discrete Chebyshev Low Pass Filter as:

H ( )
1 C
2
n
1
(tan( 0 . 5 ) / tan( 0 . 5
))
Using C1(1)=1, we let p ,
max 10 log 1
10 min
Also using s ,
We find the filter order
10
max
min
10
1
cosh 1 [tan( 0 . 5 s ) / tan( 0 . 5 p )]
cosh
/ 10
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min
/ 10
max
/ 10
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18
Example: Design specs. for a LP digital Chebyshev filter:

At f = 0 Hz, 1= 0dB
At f = 2250 Hz, max= 3dB
At f = 2500 Hz, min= 10dB
f smp
Filter order n = 3,
H 3(z)
fs
f smp
Kc = 1
(z2
0 . 09 ( z 1 ) 3
0 . 15 z 0 . 72 )( z 0 . 54 )
A.S.Kayhan

At = 2/5 rad, max= 1dB
At = /2 rad, min= 9dB
Filter order n = 3
H 3(z)
(z2
0 . 736 ( z 1 ) 3
0 . 505 z 0 . 619 )( z 0 . 472 )
Example: Design specs. for a LP digital Butterworth filter:

At = /2 rad, max= 3dB
= 0 rad, = 0dB
At = 5/9 rad, min= 10dB
Filter order n = 7,
H
(z)
z(z2
Kb = 1
0 . 01656 ( z 1 ) 7
0 . 636 )( z 2 0 . 232 )( z 2 0 . 052 )
A.S.Kayhan
19
Frequency Transformations:
The objective is to obtain transfer functions of other types of
dicrete filters from already available prototype Low Pass
Filters using tranformation functions g(z) as
H (z) H
LP
( g ( z ))

At f = 0 Hz, = 0dB
At f = 2250 Hz, 2= 21dB
At f = 2500 Hz, 3= 27dB
LPF
(z)
z3
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0 . 09 ( z 1 ) 3
0 . 691 z 2 0 . 802 z 0 . 39
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20
Now, we want a HPF with cutt-off at 3.6kHz, we use

following transformation
( z 1 0 . 5095 )
1
z
HPF
(z)
z3
(1 0 . 5095 z 1 )
0 . 0066 ( z 3 3 z 2 3 z 1 )
2 . 361 z 2 2 . 102 z 0 . 6884
A.S.Kayhan
Design of FIR Filters by Windowing:

Ideal frequency response and corresponding impulse
response functions are H ( e j ), h n .
d
Generally hd[n] is infinitely long. To obtain a causal practical

FIR filter, we can truncate it as
,0 n M
h n
h n d
, otherwise.
0
In a general form we can write it as

h n h d n w n
Where w[n] is window function.

A.S.Kayhan
21
In frequency domain, we have

H (e
1
)
2
(e
)W ( e
)d .
A.S.Kayhan
Some commonly used window functions are Rectangular,

Bartlett, Hamming, Hanning, Blackman, Kaiser.
Some important factors in choosing windows are:
Main lobe width and Peak side lobe level. For rectangular:
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22
Main lobe width and transition region; Peak side lobe level
and oscillations of filter are related.
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23
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24
Discrete Time Fourier Series (DTFS):

~
Consider a periodic sequence x n with period N:

~
x n x n N .
Which can be represented by a Fourier series as:

N 1
1
x n
N
X [ k ]e
2
nk
N
k0
The DTFS coefficients are obtained as:

N 1 ~
x n e
X [k ]
2
nk
N
n0
Let
A.S.Kayhan
2
N
The DTFS analysis and synthesis equations are:

N 1 ~
x n W
X [k ]
nk
N
n0
x n
1
N
N 1
X [k ] W
nk
N
k 0
Example: Consider the periodic impulse train:

~
x n
[ n rN ]
The DTFS
~
X [k ]
N 1
n W
nk
N
for all k .
n0
A.S.Kayhan
25
Properties:
1- Linearity: It is linear.
2- Shift of a sequence: If
x n X [ k ]
x n m W
then
Similarly,
nl
N
km
N
X [k ]
x n X [ k l ]
3- Periodic convolution: Consider two periodic

sequences with period N
~
x 1 n X 1 [ k ]
x 2 n X 2 [ k ]
Then,
~
X 3[ k ] X 1[ k ] X 2 [k ]
x 3 n
N 1 ~
x 1 [ n m ] x 2 [ m ].
m0
A.S.Kayhan
Example: Consider periodic convolution of two sequences:
A.S.Kayhan
26
Sampling the Fourier Transform:

Consider signal x[n] with DTFT X() and assume by
sampling X() , we get
~
X [ k ] X ( ) |
2

k
N
X
k
N
X [ k ] could be the sequence of DTFS coefficients
of a periodic signal which may be obtained as

1
x n
N
~
N 1
X [k ] W
nk
N
k 0
A.S.Kayhan
Substituting,
X
x m
e j m ,
X k X ( ) | 2
then,
x n x n *
n rN
x n
rN .
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27
If N is too small, then aliasing occurs in the time domain.
If there is no aliasing, we can recover x[n].
A.S.Kayhan
Discrete Fourier Tranform (DFT):

DFT is obtained by taking samples of the DTFT.
Remember DTFT is defined as:
X (e
x [ n ]e
j n
We take samples of X(ej) at uniform intervals as:

X [k ] X (e
We define:
W
2
k
N
j
, k 0 ,1 , N 1 .
2
N
A.S.Kayhan
28
Then the DFT is defined as (analysis equation):

N
X [k ]
x[ n ] W
kn
N
n0
for k=0,1,...,N-1.
The inverse DFT is defined as (synthesis equation):
x[ n ]
1
N
X [k ] W
kn
N
n0
for n=0,1,...,N-1.
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Example:
N=5
29
N=10
Properties:
1- Linearity: DFT is a linear operation.
2- Circular shift:
x n m
N , 0
n N 1 e
A.S.Kayhan
2
km
N
X [ k ].
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30
3- Circular convolution:
x 3 n x 1 n
x 2 n
N 1
x 1 [ n m
] x2[m ]
m 0
Example:
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Example:
N=L
31
N=2L
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4- Multiplication(Modulation):
x 3 n x 1 n x 2 n X
k .
Linear Convolution Using the DFT:

Since there are efficient algorithms to take DFT, like FFT,
we can use it instead of direct convolution as:
1. Compute N point DFTs, X 1 k and X 2 k .
2. Multiply them to get X 3 k X 1 k X 2 k .
3. Compute the inverse DFT, to get:
x 3 n x 1 n
x 2 n
A.S.Kayhan
32
Linear Convolution of Finite Length Signals:

Consider two sequences x1[n] of length L and x2[n] of
length P, and x3[n] = x1[n]* x2[n].
Observe that x3[n] will be of length (L+P-1).
Circular Convolution as Linear Convolution with

Aliasing :
Consider again x1[n] of length L and x2[n] of length P,
and x3[n] = x1[n]* x2[n].
X 3 (e
) X 1 (e
) X 2 (e
A.S.Kayhan
Taking N samples
X
k X 2 k .
Now, taking the inverse DFT, we have

x 3 n rN , 0 n N 1
x 3 p n r

0,
otherwise.
x 3 p n x 1 n
x 2 n
This circular convolution is identical to the linear

convolution corresponding to X1(ej) X2(ej),
if N (L+P-1).
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33
Example:
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Example:
34
A.S.Kayhan
LTI Systems Using the DFT :

Consider x[n] of length L and h[n] of length P, and y[n] =
x[n]* h[n] will be of length (L+P-1).
To obtain the result of linear convolution using the DFT,
we must use N ( (L+P-1))-point DFTs. For that x[n]
and h[n] must be augmented with zeros (zero-padding).
Overlap-Add Method :
Consider h[n] of length P, and length of x[n] is much
grater than P.
We can write x[n] as combination of length L segments:
x n rL , 0 n L 1
x r n
,
0
,
otherwise
x n
x r n rL
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35
Since the system is LTI then

y n
y r n rL ,
y r n x r n * h n
Each output segment yr[n] can be obtained using (L+P-1)

point DFT.
Example:
A.S.Kayhan
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36
Overlap-Save Method :
Consider again h[n] of length P, and length of x[n] is much
grater than P.
In this method L-point circular convolution (or DFT) is
used. Since the first (P-1) points will be incorrect,
input segments must overlap.
We can write each L sample segment of x[n] as :
x r n x n r ( L P 1 ) ( P 1 ) ,
y n
0 n L 1
y r n r ( L P 1 ) ( P 1 ) ,
y r n y rp n , P 1 n L 1
A.S.Kayhan
A.S.Kayhan
Example:
37
Efficient Computation of the DFT :

Remember the definition of the DFT and inverse DFT
N 1
x[ n ]
N 1
X [k ] W
kn
N
X [k ]
n0
x[n ] W
kn
N
n0
N2 comlex multiplications and N(N-1) additions or 4N2 real

mult. and 4N(N-2) real additions necessary:
Re{ x [ n ] } Re{ W Nkn }
kn
Im{ x [ n ] } Im{ W N }
Re{ x [ n ] } Im{ W Nkn }
j Im{ x [ n ] } Re{ W kn }
N
X [k ]
n0
A.S.Kayhan
To improve the efficiency, we can use some properties as:

1) W
2) W
k(N n)
N
k(N n)
N
W
W
kn
N
kn
N
(W
W
kn
N
)*
(k N )n
N
Example:
Re{ x [ n ] } Re{ W
kn
N
} Re{ x [ N n ] } Re{ W
Re{ x [ n ] } Re{ x [ N n ] } Re{ W
kn
N
k(N n)
N
}.
A.S.Kayhan
38
Fast Fourier Transform(FFT):

Fast Fourier Transform algorithms are used to implement
DFT efficiently to reduce the number of multiplication
and addition operations.
Two main algorithms are:
Decimation in time and Decimation in frequency. Both
these algorithms require that N=2m.
The number of operations using either of these will
require (N/2)log2N complex multiplications and Nlog2N
additions.
Direct implementation of DFT requires N2 complex
multiplications and additions. For N=1024= 210
N2=1048576
but
Nlog2N=10240.
A.S.Kayhan
Decimation in time FFT Algorithm:

Assume N=2m
N 1
X [k ]
kn
N
x[n ] W
, k 0 ,1 , , N 1
n0
Which can be written as

X [k ]
x[ n ] W
kn
N
n even
x[n ] W
kn
N
n odd
With n = 2r (even) and n = 2r+1 (odd)

N / 2 1
X [k ]
N / 2 1
x [ 2 r ] (W
2
N
) rk W
k
N
r0
x [ 2 r ] (W
rk
N / 2
r0
G k
2
N
) rk
r0
N / 2 1
x [ 2 r 1 ] (W
N / 2 1
) W
k
N
x[ 2 r 1] W
rk
N / 2
r0
k
N
k .
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39
X [ k ] G k
k
N
k .
Requires N+2(N/2)2 multiplications

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Since G[k] and H[k] requires 2(m-1)-point DFTs, each one can
be similarly decomposed until we reach 2-point DFTs.
A.S.Kayhan
40
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A.S.Kayhan
Butterfly:
(in place comp.)
Bit reversed
order
41
Decimation in frequency FFT Algorithm:

Again assume N=2m
N 1
X [k ]
x[n ] W
kn
N
, k 0 ,1 , , N 1
n0
then
N 1
X [2 r ]
x[ n ] W
2 rn
N
r 0 ,1 , , ( N / 2 1 )
n0
N / 2 1
X [2 r ]
N 1
x[ n ] W
2 rn
N
n0
2 rn
N
N / 2 1
x[ n ] W
2 rn
N
n0
N / 2 1
X [2r ]
x[ n ] W
nN /2
N / 2 1
X [2 r ]
x[ n N / 2 ] W
2r(n N /2)
N
n0
N / 2 1
( x [ n ] x [ n N / 2 ]) W
rn
N /2
n0
rn
N /2
n0
Similarly for
g [n ] W
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0 r ( N / 2 1)
N / 2 1
X [ 2 r 1]
( x [ n ] x [ n N / 2 ]) W
n
N
rn
N /2
n0
N / 2 1
h [ n ] W Nn W
rn
N /2
n0
A.S.Kayhan
42
Procedure is repeated until 2-point DFTs
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Computation of Inverse DFT:

x[ n ]
1
N
X [ k ]W
kn
N
, n 0 ,1 , , N 1
n0
X [k ]
x[ n ] W
kn
N
, k 0 ,1 , , N 1
n0
1 N
1
x [n]
X * [ k ]W Nkn
DFT
N n0
N
*
1
x[ n ]
DFT X * [ k ]
N
*
[k ]
A.S.Kayhan
43
End of Part 3
A.S.Kayhan
44
DIGITAL
SIGNAL
PROCESSING
Part 4
J.S. Lim, Two Dimensional Signal Processing, in Advanced Topics In Signal Processing, Prentice-Hall.
A.S.Kayhan
Signals:
Impulses: The 2-D impulse is,
1, n1 n 2 0
n1 , n 2
0, otherwise
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xn1 , n2 may be represented as a linear

Any sequence
combination of shifted impulses:
xn1 , n2
x[k , k ] [n
1
k1 , n2 k 2 ].
k1 k 2
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Line impulses are:

1, n1 0
xn1 , n2 T n1
0, otherwise
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or
1, n2 0
xn1 , n2 T n2
0, otherwise
or
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1, n1 n2
xn1 , n2 T n1 n2
0, otherwise
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Step sequence:
n1 , n2 0
otherwise
1,
un1 , n2
0,
un1 , n2
n1
n2
[k , k ].
1
k1 k 2
or
n1 , n2 un1 , n2 un1 1, n2 un1 , n2 1 un1 1, n2 1
A.S.Kayhan
Separable sequences:
A separable sequence can be written as
xn1 , n2 f1 n1 f 2 n2
The impulse is separable
n1 , n2 n1 n2
Periodic sequences:
~
x n1 , n2 is periodic with period N1xN2 if
~
x n1 , n2 ~
x n1 N1 , n2 ~
x n1 , n2 N 2
Example: Periodic with 2x4
~
x n1 , n2 cos n1 ( / 2)n2
A.S.Kayhan
Systems:
The relation between the input
of the system is given by
x n 1 , n 2 and the response
y n 1 , n 2 F x n 1 , n 2 .
Linearity: A system is linear if

F a x 1 n 1 , n 2 b x 2 n 1 , n 2
where
a y 1 n 1 , n 2 b y 2 n 1 , n 2 .
y i n 1 , n 2 F x i n 1 , n 2
, i 1, 2 .
Time Invariance: A system is time-invariant if

F x n 1 m 1 , n 2 m
y n 1
m 1 , n 2 m 2 .
Convolution:
For a LSI system with impulse response,
input/output relation is given by
A.S.Kayhan
h n 1 , n 2 ,the
yn1 , n2 xn1 , n2 * hn1 , n2
x[k , k ]h[n
1
k1 , n2 k 2 ].
k1 k 2
Where * denotes 2-D convolution.

Convolution with a delayed impulse :
xn1 , n2 * n1 m1 , n2 m2 x[n1 m1 , n2 m2 ].
A.S.Kayhan
Example:
y n1 , n 2
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x [ k 1 , k 2 ] h [ n 1 k 1 , n 2 k 2 ].
k 1 k 2
A.S.Kayhan
If the impulse response is separable

hn1 , n2 h1 n1 h2 n2
then
yn1 , n2
x[k , k ]h [n
1
k1 ]h2 [n2 k 2 ]
k1 k 2
h [n
1
k1
h [n
1
k1 ] x[k1 , k 2 ]h2 [n2 k 2 ]

k 2
k1 ] f k1 , n2
k1
A.S.Kayhan
Stability:
A system is BIBO stable iff a bounded input leads to a
bounded output. If
h[n , n ]
1
n1 n2
then the system is BIBO stable.
A.S.Kayhan
Discrete Space Fourier Transform (2D F.T.):

2D Fourier Transform and its inverse are defined as
X (1 , 2 )
xn , n e
1
j1n1 j 2 n2
n1 n2
xn1 , n2
1
(2 ) 2
X (1 , 2 ) e j1n1 e j2 n2 d1d 2 .
X (1 , 2 ) is periodic with 2 in both variables.

X (1 , 2 ) X (1 2 , 2 ) X (1 , 2 2 ).
A.S.Kayhan
A.S.Kayhan
Example:
Frequency Response:
If the input to Linear Shift Invariant (LSI) system is
x[n1 , n2 ] e j1n1 e j2 n2
then
y[ n1 , n2 ] H (1 , 2 ) e j1n1 e j2 n2
where
H (1 , 2 )
hk , k e
1
j1k1 j2 k 2
k1 k 2
is the frequency response function of the system.
Example:
Given LPF
A.S.Kayhan
1, 1 a and 2 b
H (1 , 2 )
otherwise
0,
This function can be written as
H (1 , 2 ) H1 (1 ) H 2 ( 2 ).
A.S.Kayhan
then
h[n1 , n2 ] h1[n1 ] h2 [n2 ]
sin(an1 ) sin(bn2 )
.
n1
n2
A.S.Kayhan
Example:
Given impulse response and magnitude response of a 2D LPF
as
A.S.Kayhan
10
Original
LP-Filtered
HP-Filtered
A.S.Kayhan
Example:
Given frequency response of a 2D LPF as
1,
H (1 , 2 )
0,
12 22 c
12 22 c and 1 , 2
A.S.Kayhan
11
then the impulse response sequence is

c
h[n1 , n2 ]
J 1 c n12 n22
2 n12 n22
where J1(x) is the Bessel function.
A.S.Kayhan
2D Z-Transform:
2D Z-Transform is defined as
X ( z1 , z2 )
xn , n z
1
n1
z2
n2
n1 n2
where z1 and z2 are complex variables. The space

represented by (z1, z2) is four-dimesional (4-D).
A.S.Kayhan
12
Example:
Given 2D sequence
x[n1 , n2 ] a n1 b n2 un1 , n2
then
X ( z1 , z2 )
n1
b n2 un1 , n2 z1 n1 z 2 n2
n1 n2
a

n1 0 z1
n1
n2
b
1
1

, z1 a and z 2 b
1 az11 1 bz 21
n2 0 z 2
A.S.Kayhan
Inverse Z-Transform:
2-D polynomials, in general, can not be factored as a product
of lower order polynomials, therefore the partial fraction
expansion is not a general procedure for 2-D signals.
Linear, Constant Coefficient Difference Eq.:

LCCDE is given in the following form
ak , k yn
1
( k1 , k 2 ) R A
k1 , n2 k 2
bk , k xn
1
k1 , n2 k 2
( k1 , k 2 ) RB
where a and b are known, with boundary conditions.

A.S.Kayhan
13
Example:
Given an IIR filter with a first quadrant impulse response as
H ( z1 , z 2 )
1
1 0.5 z11 z21
then
H ( z1 , z 2 )
Y ( z1 , z 2 )
1
X ( z1 , z 2 ) 1 0.5 z11 z 21
Y ( z1 , z 2 ) 0.5 Y ( z1 , z 2 ) z11 z 21 X ( z1 , z 2 )
y[n1 , n2 ] 0.5 y[n1 1, n2 1] x[n1 , n2 ]
A.S.Kayhan
End of Part 4
A.S.Kayhan
14

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LECTURE NOTES ON

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