Académique Documents
Professionnel Documents
Culture Documents
COURSE ORGANIZATION
Webpage:
http://www.ece.McMaster.CA
Assessment:
Assignments: 15%
Labs: 15%
Midterm: 30%
Final exam: 40%
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COURSE ORGANIZATION
Teaching assistants:
Mohamed Saad (tutorial), CRL/B104, ext. 27904
Ramy Gohary (labs), CRL/B104, ext. 27904
Abbas Ebrahimi-Moghadam (labs), CRL/B109B, 22009
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LAB SESSIONS
There are eight possible lab session times (maximal capacity 13 persons per
session):
Friday, every other week starting from September 6
Monday, every other week starting from September 9
Tuesday, every other week starting from September 10
Wednesday, every other week starting from September 11
Friday, every other week starting from September 13
Monday, every other week starting from September 16
Tuesday, every other week starting from September 17
Wednesday, every other week starting from September 18
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LAB SESSIONS
Five labs, everybody works alone, lab report deadline is one week
Please, choose today one of six possible lab session times (use sign-up sheets
near CRL/222)!
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Introduction
Time-Domain Analysis of Discrete Signals and Systems
Frequency-Domain Analysis Using Fourier
and z-Transforms
Fast Fourier Transform (FFT) Processing
Design of Nonrecursive and Recursive Digital Filters
Random Signal Analysis
DSP Processors
Adaptive Filtering and Adaptive Signal Processing
Spectral Analysis and Signal Compression
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COURSE TEXTBOOK
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DSP operations:
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Advantages of DSP:
digital data storage and transmission is much more effective than in the
analogue form
flexibility: processing functions can be altered or adjusted
possibility of implementing much more complicated processing functions
than in analogue devices
efficient implementation of fast algorithms and matrix-based processing
speed of digital operation tends to grow rapidly with the years of technical
progress
a very high accuracy and reliability is possible to achieve
dynamic range can be increased
signal multiplexing: simultaneous (parallel) processing
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s(t)
sampling of
analog signal
t
t
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quantization levels
6
5
4
3
2
1
0
0 1 2 3 4 5 6 7 8 9 10 11
sampling instants
sample 2 2 3 5 5 4 3 3 4 2 0 3
values
binary 010 010 011 101 101 10 0 011 011 10 0 010 0 0 0 011
codes
errors
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digital
0 1 2 3 4 5 6 7
111
110
101
10 0
analog
0 11
010
0 01
000
Unit-impulse function:
(
1, n = 0
(n) =
0, n =
6 0
n
X
u(n) = (m) integration
m=
(n) = u(n) u(n 1) differentiation
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(n)
1
n
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Periodic signals:
x(t) = x(t + T ) x(n) = x(n + N )
Finite-energy signals:
Z
X
E= |x(t)|2 dt < E= |x(n)|2 <
n=
Finite-power signals:
Z T N
X
1 2 1
P = lim |x(t)| dt < P = lim |x(n)|2 <
T 2T T N 2N
n=N
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sum of
all waves second
wave
first third
wave wave
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Sinusoids:
x(t) = A sin(t + ) x(n) = A sin(n + )
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A sine wave as the projection of a complex phasor onto the imaginary axis
Im
A
Re t
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n
x(n) Ambiguity in digital sinusoids
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sin(1 t) = sin(2 t) , 1 6= 2 =
2f1 t = 2f2 t + 2m , m = . . . , 2, 1, 1, 2, . . . =
-F F f
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Remarks:
The frequency fN = 2F is referred to as the Nyquist rate
Digital signal ambiguity is often termed as the aliasing effect
Equation () represents the particular formulation of the
Shannon-Nyquist-Kotelnikov Sampling Theorem
Well study the differences between digital and analog signals further using
frequency-domain signal analysis
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2 TIME-DOMAIN ANALYSIS
2.1 Linear Time-Invariant (LTI) Systems
Definition of a system:
y(n) = T {x(n)}
where T {} is an operator that maps an input sequence x(n) into an
output sequence y(n).
Linear system: A system (or processor) is linear if it obeys the principle of
superposition.
Principle of superposition: If the input of a system contains the sum of
multiple signals then the output of this system is the sum of the system
responses to each separate signal.
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-2 -1 0 1 2 n
x(- 2 ) (n+2)
n
x(- 1) (n+1)
n
x(0) (n)
n
x(1) (n- 1)
n
x(2) (n-2)
n
X
x(n) = x(k)(n k)
k=
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y(n) = T
{x(n)}
X
= T x(k)(n k)
k=
X
= x(k)T {(n k)}
k=
X
= x(k)h(n k)={x(n)} {h(n)} convolution sum
k=
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n n
An important property of convolution:
X
X
{x(n)} {h(n)} = x(k)h(n k) = h(k)x(n k)
k= k=
= {h(n)} {x(n)} =
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{x(n)}[{h1(n)} {h2(n)}]
= [{x(n)} {h1(n)}]{h2(n)} associativity
{x(n)}[{h1(n)} + {h2(n)}]
= {x(n)} {h1(n)} + {x(n)} {h2(n)} distributivity
x(n) y(n)
h1(n) h2 (n) = x(n) h2(n) h1(n)
y(n)
= x(n) h1(n)* h2 (n) y(n)
h1(n)
x(n) y(n) x(n) y(n)
= h1(n) +h2 (n)
h 2(n)
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x(k) N= 6
1
0 5 k
y(n)
6
0 5 10 n
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x(-7 -k ) x(k)
1 n=- 7 y =0
x(-1 -k ) x(k) k
n=- 1 y=0
x(2 -k ) x(k) k
n =2 y= 3
x(5 -k ) x(k) k
n =5 y=6
x(k) x(8 -k ) k
n =8 y= 3
x(k) x(11-k ) k
y=
n = 11 0
x(k) x(17-k ) k
n =17 y=0
k
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3
x(k)
h(k) k
3
x(-k) k
3
x(- 2 - k) k
3
x(2 - k) k
3
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h(k), x(n-k)
3
2 2
1 1 0.5
k
y(n)
8.5
5
4
2 1.5
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Proof of if: Let the input x(n) be bounded so that |x(n)| < Lx,
n [, ]. Then
X
|y(n)| = h(k)x(n k)
k=
X
|h(k)||x(n k)|
k=
X
X
Lx |h(k)| = |y(n)| < if |h(k)| <
k= k=
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then a bounded input can be found that will cause an unbounded output.
Consider
(
h(n)/|h(n)| , h(n) 6= 0
x(n) = =
0, h(n) = 0;
X
X
y(0) = x(k)h(k) = |h(k)| =
k= k=
P
if k= |h(k)| = , the output sequence is unbounded.
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Definition: A causal system is one for which the output y(n) depends on
the inputs {. . . , x(n 2), x(n 1), x(n)} only.
Result: An LTI system is causal if and only if its impulse response
h(n) = 0 for n < 0.
Proof of if: From the definition of a causal system,
X
y(n) = h(k)x(n k)
k=
X
= h(k)x(n k)
k=0
P1
Obviously, this equation is valid if k= h(k)x(n k) = 0 i.e., if
h(n) = 0 for n < 0.
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Now, it remains to prove that if h(n) 6= 0 for n < 0, than the system
can be noncausal. Let
h(n) = 0 , n < 1
h(1) 6= 0 =
X
y(n) = h(k)x(n k) + h(1)x(n + 1) =
k=0
y(n) depends on x(n + 1) = the system is noncausal.
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Example:
n
X
y(n) = x(k) accumulator
k=
n
X n1
X
y(n) y(n 1) = x(k) x(k)
k= k=
n1
X n1
X
= x(n) + x(k) x(k)
k= k=
= x(n)
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x(n) y(n)
+
one-
sample
delay
y(n- 1)
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Result: Suppose that for a given input x(n) we have found one particular
output sequence yP(n) so that a LCCD equation is satisfied. Then, the
same equation with the same input is satisfied by any output of the form
where yH(n) is any solution to the LCCD equation with zero input
x(n) = 0.
Remark: yP(n) and yH(n) are referred to as the particular and
homogeneous solutions, respectively.
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M
X N
X
b(k) a(k)
y(n) = x(n k) y(n k) forwards
a(0) a(0)
k=0 k=1
M
X N
X 1
b(k) a(k)
y(n N ) = x(n k) y(n k) backwards
a(N ) a(N )
k=0 k=0
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y(1) = ay0 + b ,
y(2) = ay(1) + 0
= a(ay0 + b) = a2y0 + ab ,
y(3) = a(a2y0 + ab) = a3y0 + a2b ,
,
y(n) = any0 + an1b
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Remark that:
Backwards recursion:
y(1) = a1(y0 0)
= a1y0 ,
y(2) = a2y0 ,
y(3) = a3y0 ,
,
y(n) = any0
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Auxiliary result: A linear system requires that the output be zero for all
time when the input is zero for all time.
Proof: Represent zero input as a product of zero constant c = 0 and
(arbitrary) non-zero signal x(n):
c x(n) = 0 x(n) = 0
Result is proven.
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whereas x(n) = 0 , n 0 =
according to Result, the system is nonlinear!
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,
y(1) = 0 ,
y(0) = 0 ,
y(1) = a 0 + b
= b,
y(2) = ab ,
,
y(n) = an1b
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3 FREQUENCY-DOMAIN ANALYSIS
3.1 Review of Continuous-Time Fourier Transform
Consider a continuous complex signal
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Proof:
Z T /2 Z T /2
1 1 2(nk)
j T t
n(t)k (t) dt = e dt
T T /2 T T /2
sin((n k))
= = (n k)
(n k)
Result is proven, i.e., we can take exponential functions
n(t) = exp{j2nt/T } as orthonormal basis = we obtain Fourier
Series!
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. . . Complex Fourier
__ t) X1 series representation
exp ( j 2 of a periodic signal
T
. . . x(t)
exp (j ___
2 n t)
T
Xn
. . .
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Fourier series can be applied for analysis of signal spectrum! Also, this
interpretation shows that periodic signals have discrete spectrum.
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x(t)
A
_
T
_
_ _T t
2 2 2 2
Xn
n
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1 1 t
4 A Xn
10
1 t
A Xn
20
1 t
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X() = lim Xn
T
Z T /2
j 2n
= lim x(t) e T t dt
T T /2
Z
= x(t) ejt dt
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x(t)
A
_ 0 _ t
2 2
X()
A
2
__ 2 4
__ __
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Sifting property:
Z
f (t) (t ) dt = f ( )
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= ejt0
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ej0t 2 ( 0)
cos(0t) [( 0) + ( + 0)]
sin(0t) [( 0) ( + 0)]
j
t 0 0 0
T= 2
0
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() = lim p()
0
where the pulselike function
(
1 ,
p() = 2 2
0, otherwise
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Z0+
2
1
= 2 lim 2
d
0
0
2
1
= 2 lim =
0
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DTFT = CTFT t
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x(t)X() Xnx(n) t T 2
we obtain DTFT!!!
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Sufficient condition:
X
|x(n)|
n=
Proof:
X
|X()| = x(n)ejn
n=
X
|x(n)| |e jn|
| {z }
n= 1
X
= |x(n)|
n=
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x(n)
A
0 n
X()
A(N+1)
2
__ 2 4__
__
N+1 N+1 N+1
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Linearity:
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Time shifting:
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Frequency shifting:
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Time reversal:
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Differentiation in frequency:
dX()
If X() = F{x(n)} then j = F{n x(n)}
d
1 1 dX()
Also if x(n) = F {X()} then n x(n) = F {j }
d
Proof:
X
X d(ejn)
F{n x(n)} = nx(n)ejn = j x(n)
d
n= n=
d n
X dX()
= j x(n) e jn =j
d d
n=
1 dX()
F {j } = F 1{F{ n x(n)}} = n x(n)
d
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Convolution theorem:
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= X()H()
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Windowing theorem:
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Remarks:
The impulse response and transfer function represent a DTFT pair =
H() is a periodic function
The transfer function shows how different input frequency components are
changed (e.g., attenuated) at system output
This function will be very useful for the consideration of signal filtering
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Interpretation of impulse
and frequency responses
(n) h(n)
LTI system
jn
e jn LTI system
e H()
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Question: How are CTFT and Fourier series related for periodic signals?
X() = 2 ( 0)
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We know that periodic signals have line equispaced spectrum. Let X()
be a linear combination of impulses equally spaced in frequency:
X
X() = 2Xn( n0) ()
n=
Using inverse CTFT, i.e., applying it to each term in the sum, we obtain
X Z
1
x(t) = 2Xn( n0)ejt d
2
n=
X
= Xnejn0t exactly Fourier series!
n=
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x(t)
t
t X()
2
__
t
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P
Proof: The impulse train k= (t k t) is a periodic signal with
period t = applying Fourier series, we obtain, that the Fourier
coefficients
Z t/2
1 j 2n t 1
Xn = (t)e t dt =
t t/2 t
>From (), we obtain
X
X() = 2Xn( n |{z}
0 )
n= 2
t
X
2 2n
=
t t
n=
Result is proven.
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Let
X
s(t) = (t nt)
n=
Introduce the modulated signal
X
xs(t) = x(t) s(t) = x(t) (t nt)
n=
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x(t)
from analog to
digital signal using
t impulse train
s(t)
x s (t) t
t
x(n) t
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Hence,
Z
1
Xs() = X()S( ) d
2
Z X
1 2 2n
= X() d
2 t t
n=
X
1 2n
= X
t t
n=
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X()
1
M 0 M
S()
. . . .
2__
0 __
2
t t
Xs()
__
1
. . t . .
M 0 M __
2
t
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X()
1
M 0 M
S()
. . . .
2__t 0 __
2 =2
t
Xs()
M
__
1
. . t . .
M 0 M 2M
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X()
1
M 0 M
S()
. . . .
2__ 0 2__
t t
1 Xs()
__
t
. . . .
0
note aliazing!!!
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Xf () =HLP()Xs()
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X()
1
M 0 M
1 X () H ()
t s , LP
t
. . . .
C 0 C
Xf ()
1
M 0 M
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Poor recovery of an analog signal from its samples using an ideal lowpass
filter
X()
1
M 0 M
t
Xs(), HLP()
1
t
. . . .
C 0 C
Xf ()
1
C 0 C
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X()
0 0 0
X () H ()
t s , LP
t
t 0
t
Xf ()
0 0 0
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X()
0 0 0
X () H ()
t s , LP
t
t 0
t
Xf ()
2 0 2
( t 0) t 0
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Result: Having a signal sampled at a rate higher than Nyquist rate and
infinite number of its discrete values, the signal can be exactly recovered as
X sin[(t n t)/t]
xf (t) = x(n)
(t n t)/t
n=
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Proof: We have seen that the signal can be reconstructed in the frequency
domain using ideal lowpass filter:
Xf () = Xs()HLP()
In time-domain:
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Antialiasing Reconstruction
x(t) filter
x(n) y(n) filter
y(t)
DSP
and ADC (DAC)
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x(n) x(nL)
L
sampling sampling
interval t interval L t
.
To avoid aliasing, the signal x(n) should be filtered with the cutoff
frequency M < /t = for decimated signal, L times lower
cutoff frequency d,M < /Lt is required, so that
M = L d,M
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X()
M 0 M 2
Xd()
0 L M 2
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The effect of frequency scaling for the decimated sequence: the aliasing
occurs because LM >
X()
M 0 M 2
Xd()
0 LM 2
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Upsampling system
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X()
x(n)
n Xp()
xp(n)
n Xi ()
x i (n)
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4 z-TRANSFORM
4.1 Definition and the Regions of Convergence
Consider a continuous-time LTI system with x(t) = est:
Z
y(t) = h( )x(t ) d
Z
= h( )es(t ) d
Z
= est h( )es d = H(s)est =
| {z }
H(s)
Z
H(s) = h(t)est dt , s = + j Laplace transform
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1
Re
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Recall that
(
X 1 , |c| < 1
|c|n = 1|c| =
n=0
, |c| 1
the ROC is determined by |z| > |a|
Im
a 1 Re
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1
a Re
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Im
Re
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N2
X
X(z) = x(n)z n finite duration signal
n=N1
Particular cases:
if N1 < 0 and N2 > 0 then the ROC does not include z = 0 and z =
if N1 0 then the ROC includes z = , but does not include z = 0
if N2 0 then the ROC includes z = 0, but does not include z =
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X
X(z) = x(n)z n right sided sequence
n=N1
Particular cases:
if N1 < 0 then the ROC does not include z =
if N1 0 then the ROC includes z =
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N2
X
X(z) = x(n)z n left sided sequence
n=
Particular cases:
if N2 > 0 then the ROC does not include z = 0
if N2 0 then the ROC includes z = 0
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X
X(z) = x(n)z n two sided sequence
n=
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Re Re
a) b)
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Im Im Im
Re Re
U
Re
=
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Im Im Im
1 Re 1 Re 1 Re
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Examples:
1
anu(n) 1
, |z| > |a|
1 az
(n m) z m , z 6= 0 if m > 0 , z 6= if m < 0
[sin ]z 1
sin(n) u(n) 1 2
, |z| > 1
1 [2 cos ]z + z
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Example:
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Linearity:
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Time shifting:
addition/deletion of z = 0 or z =
Differentiation of X(z):
dX(z)
If X(z) = Z{x(n)} then z = Z{n x(n)}
dz
1 1 dX(z)
Also if x(n) = Z {X(z)} then n x(n) = Z { z }
dz
with ROC = ROCx
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Time reversal:
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x(n) = anu(n)
X(z) = Y (1/z)
1
= , |z| < |a1|
1 az
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Convolution of sequences:
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z2
Z{{x(n)} {y(n)}} = X(z)Y (z) =
(z a)(z 1)
1 1 a
= 1
1
, |z| > 1
1a 1z 1 az
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Using the linearity property and the standard z-transform pairs, we obtain
that
1
{x(n)} {y(n)} = u(n) an+1u(n) , |z| > 1
1a
a 1 Re
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Result: A discrete-time LTI system is causal if and only if the ROC of its
transfer function is the exterior of a circle including infinity.
Proof: follows from the ROC properties and from the fact that h(n) is a
right-sided sequence.
Result: A discrete-time LTI system is stable if and only if the ROC of its
transfer function H(z) includes the unit circle |z| = 1.
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Example:
1
H(z) =
1 [2r cos ]z 1 + r2z 2
Im Im
r
r 1 Re
1 Re
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X(z) __
1 Y(z)
B(z) A(z)
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N
X 1
j2 kn
X(k) = x(n) e N DFT
n=0
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Proof:
1 X X
N 1 N 1
km
j2 N j2 kn
x(n) = x(m) e e N
N
k=0 m=0
N
X 1 1 NX
1 k(mn)
= x(m) ej2 N = x(n)
N
m=0 k=0
| {z }
(mn)
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Alternative formulation:
N
X 1
j 2
X(k) = x(n) W kn W = e N
n=0
N
X 1
1
x(n) = X(k) W kn
N
k=0
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k0 Schematic
W =1 representation
x(0) of DFT
. . . X(k)
k(N-1)
W
x(N-1 )
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N
X 1
j2 kn
X(k) = e N = N (k) =
n=0
the rectangular pulse is interpreted by the DFT as a spectral line at the
frequency = 0
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01 4
X(k) , X()
5
0 5 10
0 2 4
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x(n) zero-padded
signal for DFT (N=10)
1
01 4 9
Y(k) , X()
5
0 5 10
0 2
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X = Wx
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N=10
10
|DFT(x)|
0
25 20 15 10 5 0 5 10 15 20 25
FREQUENCY (HZ)
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DFT with N = 15 and zero-padding to 512 points. The signals are still
unresolved because f2 f1 = 2 Hz < 1/(N t) ' 3.3 Hz
16
N=15
14
12
10
|DFT(x)|
0
25 20 15 10 5 0 5 10 15 20 25
FREQUENCY (HZ)
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N=30
35
30
25
|DFT(x)|
20
15
10
0
25 20 15 10 5 0 5 10 15 20 25
FREQUENCY (HZ)
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DFT with N = 100 and zero-padding to 512 points. The signals are well
resolved because f2 f1 = 2 Hz > 1/(N t) = 0.5 Hz
110
N=100
100
90
80
70
60
|DFT(x)|
50
40
30
20
10
0
25 20 15 10 5 0 5 10 15 20 25
FREQUENCY (HZ)
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DFT with N = 300 and zero-padding to 512 points. The signals are fine
resolved because f2 f1 = 2 Hz 1/(N t) ' 0.17 Hz
N=300
300
250
200
|DFT(x)|
150
100
50
0
25 20 15 10 5 0 5 10 15 20 25
FREQUENCY (HZ)
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300 N=300
250
200
|DFT(x)|
150
100
50
0
25 20 15 10 5 0 5 10 15 20 25
FREQUENCY (HZ)
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{x(n)} = {. . . , x(0),
| . . . {z
, x(N 1)}, x(0),
| . . . {z
, x(N 1)}, . . .}
{x(n)} {x(n)}
where X(k) = N Xk
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Remarking that
(k+mN )n
j2 kn
W kn = e N = ej2 N = W (k+mN )n
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Proof:
1 N
pn
N
X 1 N
X 1
X
j2 kn 1 j2 N j2 kn
x(n) e N = X(p) e e N
N
n=0 n=0 p=0
N
X 1 1 NX
1 (pk)n
= X(p) ej2 N = X(k)
N
p=0 n=0
| {z }
(pk)
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Linearity:
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If X(k) = DF T {x(n)}
j2 km
then X(k)e N = DF T {x ((n m)mod N ) }
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n
n
0 N n
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N
X 1 N
X 1
x((n m)mod N ) W kn = x ((n m)modN ) W k(nm+m)
n=0 n=0
NX1
= W km x((n m)modN )W k(nm
n=0
NX1
= W km x((n m)modN )
n=0
W k(nm)mod N = W kmX(k)
where we use the facts that W k(l mod N ) = W kl and that the order of
summation in DFT does not change its result
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If X(k) = DF T {x(n)}
j2 mn
then X((k m)mod N ) = DFT {x(n)e N }
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Parseval theorem:
N
X 1 N
X 1
1
x(n)y (n) = X(k)Y (k) general form
N
n=0 k=0
NX1 N
X 1
1
|x(n)|2 = |X(k)|2 specific form
N
n=0 k=0
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Conjugation:
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Circular convolution:
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{x(n)} = {1, 1, 1, 1, 1, 0, 1, 2}
{y(n)} = {5, 4, 3, 2, 1, 1, 0, 1}
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Example (continued):
+ x(5)y(7)+x(6)y(6)+x(7)y(5) = 5
z(5) = x(0)y(5)+x(1)y(4)+x(2)y(3)+x(3)y(2)+ x(4)y(1)
+ x(5)y(0)+x(6)y(7)+x(7)y(6) = -8
z(6) = x(0)y(6)+x(1)y(5)+x(2)y(4)+x(3)y(3)+ x(4)y(2)
+ x(5)y(1)+x(6)y(0)+x(7)y(7) = 4
z(7) = x(0)y(7)+x(1)y(6)+x(2)y(5)+x(3)y(4)+ x(4)y(3)
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+ x(5)y(2)+x(6)y(1)+x(7)y(0) = 7
CoE 4TL4, Hamilton 200
12 3
x(m)
0 0 0 0
2 2 2
3 3 3
5 5 5
6 6
6 y((-m) modN)
7 4 7 4 7 4
(i.e., n= 0)
1 1 1
0 0 0
2 2 2
3 3 3
5 5 5
6 6 6
7 4 7 4 7 4
y((1-m) modN)
(i.e., n=1)
1 1 1 1
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N
X 1 N
X 1
DFT {{x(n)}
{y(n)}} = x(m)y((n m)mod N )W kn
n=0 m=0
| {z }
{x(n)}
{y(n)}
N
X 1 N
X 1
= y((n m)mod N )W knx(m)
m=0 n=0
| {z }
Y (k)W km
N
X 1
= Y (k) x(m)W km = X(k)Y (k)
|m=0 {z }
X(k)
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Multiplication:
1
then X(k)
Y (k) = DF T {x(n)y(n)}
N
Also if x(n) = DF T 1{X(k)} and y(n) = DF T 1{y(k)}
1 1
then x(n)y(n) = DF T { X(k)
Y (k)}
N
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In a short notation:
kH
XN (k) = GN/2(k) + WN N/2(k)
where GN/2(k) and HN/2(k) are the N/2-point DFTs involving x(n)
with even and odd n, respectively
x(0) GN/2 (0) XN (0)
x(2) GN/2 (1) WN0 XN (1)
x(4) N point DFT 1
GN/2 (2) WN XN (2)
2
x(6) GN/2 (3) WN2 XN (3)
N=8 3
WN
x(1) HN/2 (0) XN (4)
x(3) HN/2 (1) WN4 XN (5)
x(5) N point DFT WN5
HN/2 (2) XN (6)
2 WN6
x(7) HN/2 (3) XN (7)
WN7
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Corollary: any N -point DFT with even N can be computed via two
N/2-point DFTs. In turn, if N/2 is even then each of these N/2-point
DFTs can be computed via two N/4-point DFTs and so on. In the case
N = 2r , all N, N/2, N/4 . . . are even and such a process of splitting
ends up with 2-point DFTs!
2-point DFT
4-point DFT
FFT (Cooley and Tukey)
2-point DFT
8-point DFT
2-point DFT
4-point DFT
2-point DFT
16-point DFT
2-point DFT
4-point DFT
2-point DFT
8-point DFT
2-point DFT
4-point DFT
2-point DFT
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WN(l+N/2)
Computational complexity: each stage has N complex multiplications and
N complex additions and there are log2N stages = the total
complexity (additions and multiplications) is
CFFT = N log2N
CDFT = N 2
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6 DIGITAL FILTERS
6.1 What is Filtering
Definition: Digital filtering is just changing the frequency-domain
characteristics of a given discrete-time signal
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Y () = H()X()
x(n) y(n)
H()
h(n) h(n)
FIR IIR
n n
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A very general form of digital filter can be obtained from the familiar
equation (recall LTI-systems)
N
X M
X
a(k)y(n k) = b(k)x(n k) ARMA
k=0 k=0
where {x(n)} is the filter input signal and {y(n)} is the filter output
signal. As obtained before, the transfer function corresponding to this
equation is the following rational function
PM k
k=0 b(k)z B(z)
H(z) = PN =
k A(z)
k=0 a(k)z
If N = 0, the filter becomes a FIR (nonrecursive) one
If N > 0, the filter becomes a IIR (recursive) one
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lowpass filters (to pass low frequencies from zero to a certain cut-off
frequency C and to block higher frequencies)
highpass filters (to pass high frequencies from a certain cut-off frequency
C to and to block lower frequencies)
bandpass filters (to pass a certain frequency range [min, max], which
does not include zero, and to block other frequencies)
bandstop filters (to block a certain frequency range [min, max], which
does not include zero, and to pass other frequencies)
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ideal ideal
H() lowpass H() highpass
0 C 0 C
ideal ideal
H() bandpass H() bandstop
passband
region
stopband
region
3
P S
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passband passband
region 1 passband region
region 2
stopband stopband
stopband region 1 region 2
region
3
P1 S1 S 2 P2 S1 P P S
1 2 2
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H() = |H()|ej()
where
= angle{H()} phase response
Definition: If a signal is transmitted through a system (filter) then this
system is said to provide a distortionless transmission if the signal form
remains unaffected, i.e., if the output signal is a delayed and scaled replica
of the input signal.
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1.5
1
1
0.5
0.5
0
0.5
0.5
1
1
2 2
0 100 200 300 400 500 600 700 800 900 1000 0 100 200 300 400 500 600 700 800 900 1000
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() = ()
() = = const
H() = |H()| ej
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or, in terms of f ,
(
ejf N , 0 f 0.125
H(f ) =
0, otherwise
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1.4
0.25
1.2
0.2
0.1
0.05 0.8
0.6
0.05
0.1
0.4
0.15
0.2
0.2
0.25
0
0 1 2 3 4 5 6 7 8 9 10 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
TIME INDEX FREQUENCY
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1.4
0.25
1.2
0.2
0.1
0.05 0.8
0.6
0.05
0.1
0.4
0.15
0.2
0.2
0.25
0
0 5 10 15 20 25 30 35 40 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
TIME INDEX FREQUENCY
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1.4
0.25
1.2
0.2
0.1
0.05 0.8
0
0.6
0.05
0.1
0.4
0.15
0.2
0.2
0.25
0
0 20 40 60 80 100 120 140 160 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
TIME INDEX FREQUENCY
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-0.09
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Window properties:
windows are always real and symmetric, i.e., they must satisfy
for either even or odd N . If the ideal FIR filter has linear phase, symmetric
windows do not affect this property.
windows must be positive to avoid any changes of sequence polarities
a good window must satisfy the tradeoff between the width of the
mainlobe and the magnitudes of the sidelobes in the frequency domain
a good window must have a smooth transition at the edges of sequence
in order to reduce the truncation effect
the window length must correspond to the sequence length
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Classical window types (have been derived based on intuition and educated
guesses):
Rectangular window (corresponds to the natural truncation like in the
impulse response truncation method!)
(
1, 0 n N 1
w(n) =
0, otherwise
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0.9
0.8
0.7
50
BARTLETT WINDOW
0.6
W(f) (DB)
0.5
0.4
100
0.3
0.2
0.1
0 150
0 5 10 15 20 25 30 35 40 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
SAMPLE INDEX FREQUENCY
n o
W (f )[dB] = 10 log10 |W (f )|2 = 20 log10 {|W (f )|}
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Hamming window
(
0.54 0.46 cos[2n/(N 1)] , 0 n N 1
w(n) =
0, otherwise
Blackman window
( h i h i
0.42 0.5 cos N2n
1 + 0.08 4n , 0 n N 1
N 1
w(n) =
0, otherwise
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1
0
0.9
0.8
0.7
50
HANNING WINDOW
0.6
W(f) (DB)
0.5
0.4
100
0.3
0.2
0.1
0 150
0 5 10 15 20 25 30 35 40 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
TIME SAMPLE INDEX FREQUENCY
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1
0
0.9
0.8
0.7
50
HAMMING WINDOW
0.6
W(f) (DB)
0.5
0.4
100
0.3
0.2
0.1
0 150
0 5 10 15 20 25 30 35 40 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
TIME SAMPLE INDEX FREQUENCY
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1
0
0.9
0.8
0.7
50
BLACKMAN WINDOW
0.6
W(f) (DB)
0.5
0.4
100
0.3
0.2
0.1
0 150
0 5 10 15 20 25 30 35 40 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
TIME SAMPLE INDEX FREQUENCY
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1
0
0.9
0.8
DOLPHCHEBYSHOV WINDOW
0.7
50
0.6
W(f) (DB)
0.5
0.4
100
0.3
0.2
0.1
0 150
0 5 10 15 20 25 30 35 40 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
TIME SAMPLE INDEX FREQUENCY
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1
0
0.9
0.8
DOLPHCHEBYSHEV WINDOW
0.7
50
0.6
W(f) (DB)
0.5
0.4
100
0.3
0.2
0.1
0 150
0 5 10 15 20 25 30 35 40 0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
TIME SAMPLE INDEX FREQUENCY
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Now, let us take Dolph-Chebyshev window and design the lowpass filter
using the window method with N = 11, N = 41, and N = 161.
Remember that
sin((n 0.5N )/4)
hid(n) =
(n 0.5N )
We take the window length equal to the length of hid(n).
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1.4
0.25
1.2
TRUNCATED AND WINDOWED INPULSE RESPONSE
0.2
0.1
0.05 0.8
0.6
0.05
0.1
0.4
0.15
0.2
0.2
0.25
0
0 1 2 3 4 5 6 7 8 9 10 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
TIME SAMPLE INDEX FREQUENCY
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1.4
0.25
1.2
TRUNCATED AND WINDOWED INPULSE RESPONSE
0.2
0.1
0.05 0.8
0.6
0.05
0.1
0.4
0.15
0.2
0.2
0.25
0
0 5 10 15 20 25 30 35 40 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
TIME SAMPLE INDEX FREQUENCY
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0.25
1.2
TRUNCATED AND WINDOWED INPULSE RESPONSE
0.2
0.1
0.05 0.8
0
0.6
0.05
0.1
0.4
0.15
0.2
0.2
0.25
0
0 20 40 60 80 100 120 140 160 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
TIME SAMPLE INDEX FREQUENCY
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0
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2
POLES OF H(s)
Im (s) Im (s)
N=3 N=4
0
Re (s) Re (s)
POLES OF H(s)
N=3 Im (s) N=4 Im (s)
Re (s) Re (s)
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p s p s
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the s-plane poles and the transfer function of Chebyshev filter can be
obtained in the way similar to that for Butterworth filter
other analog filter structures are known, such as Bessel and elliptic filters
an important question is what kind of transformation can be used to
obtain digital IIR filters from the corresponding analog filters
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= 2arctan(a t 2)
d
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Another method of IIR filter design does not require analog filter design
and exploits the ARMA model
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Minimizing
Z PM jk 2
2 k=0 b(k)e
|| = Hid() PN d
jk
k=0 a(k)e
we obtain desired filter parameters a(k), k = 1, . . . , N and b(k),
k = 1, . . . , M
Efficient numerical optimization methods can be formulated to minimize
this function
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unit delay
x(n) y(n)=x(n- 1)
t
BASIC
adder/summator BUILDING
x1(n) BLOCKS
x2(n) y(n) = x1(n)+x2(n)+x3(n)
+
x3(n)
multiplier by a constant
x(n) a y(n) = a x(n)
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t
b(1)
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x(n) y(n)
+
t
- a(1)
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t
b(M- 1)
t
b(M)
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t
b(1) -a(1)
+
t
b(M) + -a(M )
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PX (x) = Probability{X x}
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Since PX () = 1, we obtain
Z
pX (x) dx = 1
Simple interpretation:
Probability{x /2 X x + /2}
pX (x) = lim0
p (x) SURFACE =Probability{x
X 1< X< x 2}
x1 x2 x
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Sample variance:
N
X 1
2 1
X = (x(n) mX )2
N 1
n=0
The weighting 1/N 1 is used instead of 1/N to make the sample
variance unbiased:
2 } = 2
E{X X
For the complex random variable:
2 = E{(X E{X})(X E{X})}
var{X} = X
= E{|X|2} |E{X}|2
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R = E{XXT }
R = E{XXH }
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w1 w2 w3 wN
y(n)
The weight vector
W = (w1, w2, . . . , wN )T
is tuned according to some adaptation criterion
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Observations vector
Rn = E{n(n)nH (n)}
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Wopt = R1
n aS Wiener filter
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Adaptive algorithm:
where we assume that the observation Xk does not contain the desired
signal (this assumption can be relaxed later)
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Vk+1 = [I Rn] Vk
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x(n) = exp(j2f
| {z S n)} + 100
| exp(j2f
{z I n)
} + (n)
|{z}
desired signal interference noise
15
10 OPTIMAL SNR
5 MU3
MU2
0
SNR (DB)
MU1
5
10
15
20
25
30
0 1000 2000 3000 4000 5000 6000 7000
NUMBER OF ITERATION
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10
30
40
50
60
70
80
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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1 2
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Periodogram:
1 N
X 1 2
Pp() = x(n)ejn
N
n=0
Example:
x(n) = A exp(j2fS n) + (n)
where fS = 0.3 and is zero-mean unit-variance complex noise.
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Periodogram (A = 1, N = 100)
150
100
PERIODOGRAM
50
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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Periodogram (A = 1, N = 1000)
1000
900
800
700
600
PERIODOGRAM
500
400
300
200
100
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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Periodogram (A = 1, N = 10000)
250
200
150
PERIODOGRAM
100
50
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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6
PERIODOGRAM
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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12
10
PERIODOGRAM
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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3.5
2.5
PERIODOGRAM
1.5
0.5
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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1 M
X 1 2
Pl () = xl (n)ejn
M
n=0
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10
AVERAGED PERIODOGRAM
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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Remark: There are many other advanced spectral analysis methods which
perform significantly better than the periodogram methods
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3.5
3
WELCH PERIODOGRAM
2.5
1.5
0.5
0
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
FREQUENCY
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Useful property:
r(k) = r(k)
The two definitions of the power sprectral density are identical if r(k)
decays sufficiently fast.
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Correlogram:
N
X 1
Pc()= r(k)ejk
k=N +1
where we can use either unbiased or biased estimates of r(k).
Unbiased estimate:
(
1 PN 1 x(i)x(i k) , k 0
r(k)= N k i=k
r(k) , k<0
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Unbiased estimate:
1
r(0) = [x(0)x(0) + x(1)x(1) + x(2)x(2) + x(3)x(3)]
4
1
r(1) = [x(1)x(0) + x(2)x(1) + x(3)x(2)]
3
1
r(2) = [x(2)x(0) + x(3)x(1)]
2
r(3) = x(3)x(0)
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Biased estimate:
(
1 PN 1 x(i)x(i k) , k 0
r(k)= N i=k
r (k) , k<0
Prof of bias:
N
X 1 N
X 1
1 1 N k
E{r(k)} = E{x(i) x(i k)} = r(k) = r(k)
N N N
i=k i=k
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Biased estimate:
1
r(0) = [x(0)x(0) + x(1)x(1) + x(2)x(2) + x(3)x(3)]
4
1
r(1) = [x(1)x(0) + x(2)x(1) + x(3)x(2)]
4
1
r(2) = [x(2)x(0) + x(3)x(1)]
4
1
r(3) = x(3)x(0)
4
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The signal y(m) can be viewed as the output of the filter with the transfer
function
N 1
1 X
X() = x(k)ejk
N k=0
Relationship between filter input and output PSDs:
X
Py () = |X()|2P()= |X()|2 r(k)ejk
k=
X
= |X()|2 (k)ejk = |X()|2
k=
2
NX1
1 jn
=Pp()
= x(n)e
N
n=0
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Summarizing, we have proven the following two equalities for the spectrum
of the auxiliary signal y(n):
Py () = Pp() , Py () = Pc() =
Pp() = Pc()
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r(k) is a poor estimate for higher lags k. Hence, let us truncate it (use
M N points)
Let us use some lag window
M
X 1
PBT () = w(k)r(k)ejk
k=M +1
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compressed orthonormal
signal basis original
signal
=
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Original signal:
Compression:
where
M N
=
L Nc
and T is the M L transformation matrix. The signal X can be
interpreted as a vector of coefficients of expansion using orthonormal basis
T:
x = TX
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Signal reconstruction:
orthonormal
recovered basis
signal compressed
= signal
transmission
x(n) or storage ~
x(n)
compression reconstruction
TH of T
compressed
data
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SVD of a N N matrix
N
X
R= iUiViH 1 2 . . . N
i=1
Compressed representation
M
X
R' iUiViH , M <N
i=1
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1.5
1
ORIGINAL SIGNAL
0.5
0.5
1.5
2.5
100 200 300 400 500 600 700 800 900 1000
DISCRETE TIME
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1.5
1
RECONSTRUCTED SIGNAL
0.5
0.5
1.5
2.5
100 200 300 400 500 600 700 800 900 1000
DISCRETE TIME
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1.5
1
RECONSTRUCTED SIGNAL
0.5
0.5
1.5
2.5
100 200 300 400 500 600 700 800 900 1000
DISCRETE TIME
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CONCLUSIONS
many of you will work on DSP problems in future
when working with digital signals, do not forget their sampling rate!
we considered only the most general topics of DSP
to broaden and deepen your knowledge in specific DSP topics, an
advanced DSP course might be needed
DSP makes fun!
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