Typical Real Time DSP System

Digital Signal Processing
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A Typical DSP System
Sampling
&
A/D
Analog
Pre-filter
H
PRE
(f)
x(t) x[n] x
a
(t)
Analog
in
DSP
Filter
H
DSP
(z)
Analog
out
Analog
Post-filter
H
POST
(f)
D/A
Converter
H
DAC
(f)
y
a
(t) y(t)
y[n]
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Analog Signal Processing
x
a
(t) : an analog signal
X
a
() = x
a
(t)e
-jt
dt is its Fourier transform
The inverse Fourier transform is
x
a
(t) = (1/2) X
a
() e
jt
d
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Analog Signal Processing
Analog
Filter
H
a
()
y
a
(t)
x
a
(t)
H
a
() is the Fourier transform of the
impulse response, h
a
(t), of the filter
y
a
(t) = h
a
(t-) x
a
() d
OR, equivalently, in frequency domain
Y
a
() = H
a
() X
a
()
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A Typical DSP System A Typical DSP System
Analog
Pre-filter
H
PRE
(f)
Sampling
&
A/D
DSP
Filter
H
DSP
(z)
D/A
Converter
H
DAC
(f)
Analog
Post-filter
H
POST
(f)
x
a
(t) x(t)
x[n]
y
a
(t)
y(t) y[n]
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A digital Spectral Analyzer
Analog
Pre-filter
H
PRE
(f)
Sampling
&
A/D
FFT
Processor
Display
the
Spectrum
x
a
(t) x(t)
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Linear Time-Invariant Systems
The delta function: (t)
Linearity
Time-invariant
The Impulse Response
The Convolution Integral
The Convolution Theorem
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Sinusoidal Response of LTI System
e
jt
H
a
() e
jt
H
a
()
H
a
() = |H
a
()| e
jARG{H
a
()}
H
a
(
1
) A
1
e
j
1
t
+ H
a
(
2
) A
2
e
j
2
t
A
1
e
j
1
t
+A
2
e
j
2
t
H
a
()
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Linear Filtering
X
a
()
=2f
Y
a
()
=2f
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Sampling a Signal
x[n] = x(nT)
x(t)
T = 1/f
s
T: sampling period, (second)
f
s
: sampling frequency (sample/second)
x(t) x[n]
t n
0 1 2 3 4
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Discrete-Time Fourier Tranform (DTFT)
X() =
n
x[n] e
-jn
X() is a continuous function of
X() is periodic on with the period 2
-4
-2 0 2 4
X()

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Fourier Transform and The DTFT
If X
a
() = x
a
(t)e
-jt
dt and x[n] = x
a
(nT)
Then X() = (1/T)
k
X
a
( - k2/T)
-4f
s
-2f
s
0 2f
s
4f
s
1/T
0
X
a
()
X()

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Aliasing
-4f
s
-2f
s
0 2f
s
4f
s
X()
0
-4f
s
-2f
s
2f
s
4f
s

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Aliasing
Cos( t) = (e
jt
+ e
-jt
)
-4f
s
-2f
s
0 2f
s
4f
s

-4f
s
-2f
s
0 2f
s
4f
s

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Aliasing
0 10 20 30 40 50 60 70 80 90 100
-1
-0.8
-0.6
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
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Anti-Aliasing Filter
A(f)
f
f
pass
f
s
/2 f
stop
Attenuation A(f) = -20 log {|H(f)/H(0)|}
dB per Octave or dB per Decade
A single-pole filter has 6 dB/Oct. or 20 dB/Dec.
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DAC & Analog Reconstruction
D/A
Converter
H
DAC
(f)
y[n] y
a
(t)
y
a
(t) =
n
y[n] h
DAC
(t-nT)
Y
a
() = Y() |
=
H
DAC
()
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Staircase Reconstructor
-100 -80 -60 -40 -20 0 20 40 60 80 100
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
h(t)
1
T/2 -T/2
f = 1/T = f
s
1, -T/2 t T/2
0, otherwise
h(t) = u(t) - u(t-T) = {
H(f) = (T/j) [Sin(fT)/(fT)]
t
f
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Staircase Reconstructor
-100 -80 -60 -40 -20 0 20 40 60 80 100
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
f
s
-f
s
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Anti-Image Post Filter
-100 -80 -60 -40 -20 0 20 40 60 80 100
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
f
s
/2
f
-f
s
/2
Removing high frequency images
Equalizing D/A converters in-band distortion
(may be done digitally)
-20 log[H(f
s
/2)/H(0)] ~ 3.9 dB
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Ideal Reconstructor
T
H(f)
f
-f
s
/2 f
s
/2
-100 -80 -60 -40 -20 0 20 40 60 80 100
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
h(t) = Sin(t/T)/(t/T)
t
T 2T
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A/D Converter & Quantization
R
:

f
u
l
l

s
c
a
l
e
R/2
3Q
2Q
Q
0
-Q
-2Q
-3Q
-R/2
R/Q = 2
B
Q: step size
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Quantization Noise Power
P(e)
1/Q
e
-Q/2 Q/2
E(e
2
) = (1/Q) e
2
de = Q
2
/12
or
e
rms
= Q/12
1/2
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Signal to Noise Ratio: SNR
Noise Power = Q
2
/12
Signal Power = ?
Assuming Sine Wave -> Signal Power = A
2
/2
Assuming Full Scale -> A = R/2 -> Signal Power = R
2
/8
Assuming B bits -> Q = R/2
B
Signal to Noise Power Ration = (R
2
/8) / (Q
2
/12)
SNR (dB) = 10 log(2
2B
) + 10log(12/8)
= 6.02*B + 1.76 dB
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Discrete Time Systems
{x[0], x[1], x[2], } {y[0], y[1], y[2], }
H
Example: y[n] = 2x[n] + 3x[n-1] + 4x[n-2]
Sample by Sample Processing
Block Processing
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y[n] = 2x[n] + 3x[n-1] + 4x[n-2]
y[n] = 2x[n] + 3w
1
[n] + 4w
2
[n]
w
2
[n+1] = w
1
[n]
w
1
[n+1] = x[n]
For each new input x
DO y = 2x + 3w
1
+ 4w
2
w
2
= w
1
w
1
= x
CONTINUE
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Block Processing
y[n] = 2x[n] + 3x[n-1] + 4x[n-2]
y[0] 2 0 0 0 0
y[1] 3 2 0 0 0 x[0]
y[2] 4 3 2 0 0 x[1]
y[3] = 0 4 3 2 0 x[2]
y[4] 0 0 4 3 2 x[3]
y[5] 0 0 0 4 3 x[4]
y[6] 0 0 0 0 4
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Overlap and Add
0, 1, 2, 3, 4, 5, 6,
7, 8, 9, 10, 11, 12, 13,
14, 15, 16, 17, 18, 19,
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FIR and IIR Systems
FIR: Finite Impulse Response -- non-recursive
y[n] = 2x[n] + 3x[n-1] + 4x[n-2]
IIR: Infinite Impulse Response -- recursive
y[n] = 0.5y[n-1] + 2x[n] + 3x[n-1]
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y[n] = 0.5y[n-1] + 2x[n] + 3x[n-1]
y[n] = 0.5w[n] + 3x[n] + 4v[n]
w[n+1] = y[n]
v[n+1] = x[n]
For each new input x
DO y = 0.5w + 2x + 3v
w= y
v = x
CONTINUE
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Linearity
a
1
y[n] = a
1
y
1
[n]+ a
2
y
2
[n]
+
a
2
x
1
[n]
x[n]
x
2
[n]
y[n]
H
a
1
+
a
2
x
1
[n]
x
2
[n]
H
y
1
[n]
a
1
y
1
[n]+ a
2
y
2
[n]
H
y
2
[n]
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Time-Invariance
x[n] y[n]
y
D
[n] = y[n-D]
H D y[n-D]
x[n]
D
x
D
[n]
H y
D
[n]
x[n-D]
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Impulse Response of an LTI System
[n] = 1, for n=0, and 0, otherwise
h[n]
[n]
H
n
n
For any input x[n], the output y[n] is given by
y[n] =
m
x[m]h[n-m]
=
m
h[m]x[n-m] (direct form)
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Convolution
h[n]
[n]
H
n
n
h[n-1]
[n-1]
H
n
n
h[n-2]
[n-2]
H
n
n
x[n] = x[0][n] + x[1][n-1] + x[2][n-2] +
y[n] = x[0]h[n] + x[1]h[n-1] + x[2]h[n-2] +
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FIR and IIR Filters
An FIR filter has impulse response, h[n], that extends only over a
finite time interval, say 0 n M, and is identically zero beyond that:
{h[0], h[1], h[2], h[M], 0,0,0,0,0,0..}
M is the filter order, and the length of the filter impulse response is M+1.
An IIR filter, on the other hand, has impulse response, h[n], of infinite
duration, 0 n , and is identically zero beyond that:
{h[0], h[1], h[2], h[M], h[M+1], ..}
The convolution sum is not computationally feasible, hence it is normally
represented by a constant-coefficient linear difference equation.
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I/O Difference Equation
Impulse Response: h[n]
FIR filter: convolution sum
IIR Filter: difference equation
y[n] = ay[n-1] + x[n]
h[n] = a
n
u[n]
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Causality and Stability
A causal sequence is a right-sided sequence, and an anti-causal
sequence is a left-sided sequence. A two-sided sequence is mixed.
An LTI system is causal if its impulse response, h[n], is causal.
An LTI system is stable if for any bounded input sequence,
the output sequence is also bounded.
An LTI system is stable if and only if its impulse response
is absolutely summable, i.e.,
n
|h[n]| <
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Linear Convolution: Flip & Slide
y[n] =
m
x[m]h[n-m]
If h is of order M, i.e., h = {h[0], h[1], , h[M]} is of length M+1
and x = {x[0], x[1], ,x[L-1]} is of length L, then
y[n] =
m=max(0,n-M)
min(n,L-1)
x[m]h[n-m]
and y = {y[0], y[1], ,y[L+M]} is of length (L+M)
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Linear Convolution: Flip & Slide
h
3
h
2
h
1
h
0
h
3
h
2
h
1
h
0
h
3
h
2
h
1
h
0
0 0 0 x
0
x
1
x
2
x
3
x
n-3
x
n-2
x
n-1
x
n
x
L-1
0 0 0
y
0
y
n
y
L-1+M
h = M+1
x = L
y = h*x = L M
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Transient & Steady State
y[n] =
m=0
n
h[m]x[n-m] if 0 n < M
y[n] =
m=0
M
h[m]x[n-m] if M n L-1
y[n] =
m=n-L+1
M
h[m]x[n-m] if L n < L+M
output y[n]
n
0 M L-1 L-1+M
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Convolution of Infinite Sequence
y[n] =
m=max(0,nL+1)
min(n,M)
h[m]x[n-m]
Infinite filter, finite input: M = , L <
y[n] =
m=max(0,nL+1)
n
h[m]x[n-m]
Finite filter, infinite input: M < , L =
y[n] =
m=0
min(n,M)
h[m]x[n-m]
Infinite filter, infinite input: M = , L =
y[n] =
m=0
n
h[m]x[n-m]
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Finite filter, Infinite input
Block Processing
y
0
= h*x
0
, y
1
= h*x
1
, y
2
= h*x
2
,
y
0
= L M
x = block x
0
block x
1
block x
2
y
1
= L M
y
2
= L M
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Finite filter, Infinite input
Sample Processing
y[n] =
m=max(0,nL+1)
n
h[m]x[n-m]
= h[0]x[n] + h[1]x[n-1] + . h[M]x[n-M]
FOR each input x DO
w
0
= x
y = h
0
w
0
+ h
1
w
1
+ h
2
w
2
+.. h
M
w
M
w
M
= w
M-1
w
M-1
= w
M-2
|
|
w
1
= w
0
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Circular Buffers
y
x h
0
h
1
w
0
w
1
h
2
h
3
w
2
w
3
X
MAC: Multiply & Accumulate

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Z-Transform
{x[n], n = -, }
X(z) =
n=-
x[n]z
-n
= +x[-2]z
2
+x[-1]z
1
+x[0] +x[1]z
-1
+x[2]z
-2
+
Z-transform of the impulse response h[n]
H(z) =
n=-
h[n]z
-n
is called the Transfer function of the LTI system.
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Basic Properties of Z-transform
Linearity:
Z
a
1
x
1
[n] + a
2
x
2
[n] a
1
X
1
(z) + a
2
X
2
(z)
Delay:
Z
x[n-D] z
-D
X(z)
Convolution:
Z
y[n] = h[n]*x[n] Y(z) = H(z) X(z)
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Z-transform Examples
[n]
Z
1
[n-M]
Z
z
-M
u[n]
Z
1/(1- z
-1
)
for |z|> 1
a
n
u[n]
Z
1/(1- a z
-1
)
for |z|>|a|
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ROC: Region of Convergence
ROC: Region of z C (complex plane) where X(z)
Examples:
(0.5)
n
u[n] X(z) = 1/(1-0.5z
-1
) if |z| > 0.5
-(0.5)
n
u[-n-1] X(z) = 1/(1-0.5z
-1
) if |z| < 0.5
In general
a
n
u[n] X(z) = 1/(1-a z
-1
) if |z| > |a|
-a
n
u[-n-1] X(z) = 1/(1-a z
-1
) if |z| < |a|
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Region of Convergence: Example
X(z) = 1/(1-0.8z
-1
) + 1/(1-1.25z
-1
) = (2-2.05z
-1
)/(1-2.05z
-1
+ z
-2
)
Z-plane
0.8 1.25
I
II
III
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Causal Sequence & ROC: Poles
X(z) = A
1
/(1-p
1
z
-1
) + A
2
/(1-p
2
z
-1
) + .. A
k
/(1-p
k
z
-1
) +
Z-plane
Causal ROC
ROC:
z > max|p
k
|
causal sequence = right-sided sequence = ROC is outside of a circle
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Anti-Causal Sequence & ROC
X(z) = A
1
/(1-p
1
z
-1
) + A
2
/(1-p
2
z
-1
) + .. A
k
/(1-p
k
z
-1
) +
Z-plane
Anti-Causal ROC
ROC:
z < min|p
k
|
Anti-causal sequence = left-sided sequence = ROC is inside of a circle
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Two-sided Sequence & ROC
X(z) = A
1
/(1-p
1
z
-1
) + A
2
/(1-p
2
z
-1
) + .. A
k
/(1-p
k
z
-1
) +
Z-plane
Mixed ROC
mixed sequence two-sided sequence = ROC is inside of a ring
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Stability of a Sequence and ROC
Stable Sequence = ROC contains the UNIT CIRCLE
If H is stable , i.e.,
n=-
|h[n]| <
Then on the unit circle, i.e., |z| =1
|H(z)| = |
n=-
h[n]z
-n
|
n=-
|h[n]z
-n
| =
n=-
|h[n]|<
Therefore, its ROC contain the unit circle.
If H(z)s ROC contains the unit circle
Then
n=-
h[n]z
-n
|
|z|=1
=
n=-
|h[n]| <
Therefore, H is stable.
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Stable and Causal Sequence
Stable and Causal: All poles are inside the unit circle
Z-plane
Causal ROC
unit circle
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Stable and Anti-Causal Sequence
Stable and Anti-Causal: All poles are outside the unit circle
Z-plane
Anti-Causal ROC
unit circle
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Stable and Two-sided Sequence
Stable and Two-sided:Some poles are outside the unit circle
Some poles are inside the unit circle
Z-plane
Mixed ROC
unit circle
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Stable System
An LTI system is stable if and only if
H(z)s ROC contains the unit circle.
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Inverse Z-transform
By inspection:
First order sequence/system
X(z) = 1/(1-a z
-1
) if |z| > |a| x[n] = a
n
u[n]
X(z) = 1/(1-a z
-1
) if |z| < |a| x[n] = -a
n
u[-n-1]
Second order sequence/system
X(z) =A
1
/(1-p
1
z
-1
) + A
1
*
/(1-p
1
*
z
-1
) if |z| > |p
1
|
where A
1
= B
1
e
j
1
and p
1
= R
1
e
j
1
x[n] = 2B
1
R
1
n
cos(n
1
+
1
)u[n]
X(z) = [2B
1
cos(
1
)-2B
1
R
1
cos(
1
-
1
) z
-1
]/[1-2R
1
cos(
1
) z
-1
+R
1
2
z
-2
]
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Inverse Z-transform
Partial Fraction Expansion:
X(z) = N(z)/D(z)
N(z) is of degree N & D(z) is of degree M, M<N, Real Roots
= N(z)/(1-p
1
z
-1
)(1-p
2
z
-1
).. /(1-p
M
z
-1
)
= A
1
/(1-p
1
z
-1
)+ A
2
/(1-p
2
z
-1
)+. A
M
/(1-p
M
z
-1
)
A
k
= (1-p
k
z
-1
)X(z)|
z=p
k
N(z) is of degree N & D(z) is of degree M, M<N, Complex Roots
X(z) =
k
[A
k
/(1-p
k
z
-1
) + A
k
*
/(1-p
k
*
z
-1
)]
N(z) is of degree N & D(z) is of degree M, M<N, Multiple Roots
N(z) is of degree N & D(z) is of degree M, MN, Complex Roots
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Frequency Response & Z-Transform
The DTFT (Discrete-Time Fourier Transform)
X() =
n=-
x[n]e
-jn
X(z) =
n=-
x[n]z
-n
X() = X(z)|
z=e
j
The Inverse DTFT
x[n] = (1/2)
X()e
jn
d
The Frequency Response of the LTI System
H() =
n=-
h[n]e
-jn
Y()= H()X()
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Parseval Theorem
n=-
|x[n]|
2
= (1/2)
|X()|
2
d
Digital measurement of energy of an analog signal
n=-
|x[n]|
2
= (1/2)
|X()|
2
d
= (1/2)(1/T)
|X()|
2
d
=(1/T)
|x(t)|
2
dt
Total energy = (T)
n=-
|x[n]|
2
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Frequency Response & Pole/Zero Locations
X(z) = (1-z
1
z
-1
)/(1-p
1
z
-1
)
X() = (e
j
-z
1
)/(e
j
-p
1
)
|X()| = |(e
j
-z
1
)|/|(e
j
-p
1
)|
p
1
=|p
1
|e
j
z
1
=|z
1
|e
j
|X()|
e
j
z
1

p
1
YCJ
62
Transfer Function of an LTI System
Block Processing
Impulse
Response h[n]
I/O Convolution
Equation
Transfer
Function H(z)
Pole/zero
Pattern
I/O Difference
Equation
Frequency
Response
Specification
Filter
Design
Block Diagram
Realization
Sample Processing
YCJ
63
An Example
Transfer Function: H(z) = (5+2z
-1
)/(1-0.8z
-1
)
Impulse Response: H(z) = -2.5 + 7.5/(1-0.8z
-1
)
h[n] = -2.5[n] + 7.5(0.8)
n
u[n]
Difference Equation: H(z) = (5+2z
-1
)/(1-0.8z
-1
) = Y(z)/X(z)
Y(z) 0.8z
-1
Y(z) = 5X(z) +2z
-1
X(z)
y[n] = 0.8y[n-1] + 5x[n] + 2x[n-1]
Block Diagram: Y(z)/X(z) = (5+2z
-1
)/(1-0.8z
-1
)
YCJ
64
An Example - Continued
Block Diagram: Y(z)/X(z) = (5+2z
-1
)/(1-0.8z
-1
)
5
x[n]
0.8
z
-1
z
-1
+
y[n]
y[n-1]
x[n-1]
2
YCJ
65
An Example - Continued
Frequency Response: H(z) = (5+2z
-1
)/(1-0.8z
-1
)
=5(1+0.4z
-1
)/(1-0.8z
-1
)
H() =5 (1+0.4e
-j
)/(1-0.8e
-j
)
Using the identity: |1-ae
-j
| = [1-2a cos()+a
2
]
1/2
|H()| = 5 [1-0.8 cos()+0.16]
1/2
/[1-1.6 cos()+0.64]
1/2
|H()|
-0.4 0.8
35
5/3

YCJ
66
Direct & Canonical Forms
H(z) = N(z)/D(z)
= (b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
)/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
)
Y(z)/X(z) = (b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
)/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
)
Y(z)= X(z)(b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
)/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
)
= X(z)(b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
)
[1/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
)] Direct Form
= X(z)/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
)
[(b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
)] Canonical Form
YCJ
67
Direct Form
Y(z)= X(z)(b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
) [1/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
)]
b
0
b
1
b
2
b
L
y[n]
y[n-1] x[n-1]
x[n]
-a
1
-a
2
-a
M
y[n-2]
x[n-2]
y[n-M]
x[n-L]
z
-1
z
-1
+
z
-1
z
-1
z
-1
z
-1
YCJ
68
Canonical Form
Y(z)= X(z)/(1+ a
1
z
-1+
a
2
z
-2
++ a
M
z
-M
) [(b
0
+ b
1
z
-1+
b
2
z
-2
++ b
L
z
-L
)]
b
0
-a
1 b
1
b
2
-a
2
-a
M
w[n-1]
w[n]
x[n]
+ +
y[n]
z
-1
z
-1
b
L
z
-1
YCJ
69
Examples
Example 6.2.1: Impulse response: h = {1, 6, 11, 6}
Difference equation: y[n] = x[n] + 6x[n-1] +11x[n-2] +6x[n-3]
Transfer Function: H(z) = 1+ 6z
-1
+11z
-2
+ 6z
-3
Pole/Zeros: H(z) = (1+z
-1
)(1+2z
-1
)(1+3z
-1
)
Frequency Response, Block Diagrams
Example 6.2.3:
(a) Difference equation: y[n] = 0.25 y[n-2] + x[n]
(b) Difference equation: y[n] = -0.25 y[n-2] + x[n]
YCJ
70
Sinusoidal Steady State Response
x[n] = e
j
o
n
DTFT: X() = 2 (
) + replicas
Y() = H()X() = H(
)2 (
) + replicas
y[n] = H(
) e
j
o
n
i.e., e
j
o
n
H
H(
) e
j
o
n
= |H(
)| e
j(
o
n+Arg[H(
o
)])
cos(n
)
H
|H(
)| cos(n
+ Arg[H(
o
)] )
sin(n
)
H
|H(
)| sin(n
+ Arg[H(
o
)] )
Linearity
Ae
j
1
n
+Be
j
2
n
H
A H(
1
) e
j
1
n
+B H(
2
) e
j
2
n
YCJ
71
Sinusoidal Transient Response
Input: x[n] = e
j
o
n
u[n] : X(z) = 1/(1- e
j
o
z
-1
)
LTI system: H(z) = N(z)/D(z) (assuming real poles)
= N(z)/(1-p
1
z
-1
)(1-p
2
z
-1
)..(1-p
M
z
-1
)
all |p
k
| < 1, k = 1, 2, ., M
Output: Y(z) = X(z)H(z) = N(z)/(1- e
j
o
z
-1
)(1-p
1
z
-1
)(1-p
2
z
-1
)..(1-p
M
z
-1
)
= H(
) /(1- e
j
o
z
-1
) + A
1
/(1-p
1
z
-1
) + A
2
/(1-p
2
z
-1
) + .. A
M
/(1-p
M
z
-1
)
y[n] = {H(
) e
j
o
n
+ A
1
p
1
n
+ A
2
p
2
n
+ .. A
M
p
M
n
}u[n]
as n y[n] H(
) e
j
o
n
YCJ
72
Sinusoidal Transient Response
Example: H(z) = (5+2z
-1
)/(1-0.8z
-1
)
Y(z) = (5+2z
-1
)/(1- e
j
o
z
-1
)(1-0.8z
-1
)
Observation:
1. Stability of the Filter
2. Effective Time Constant: n
eff
= max
k
|p
k
|,
n
eff
=
n
eff
= ln( )/ln()
Example 6.3.2
YCJ
73
Unit Step Response
Input: x[n] = e
j
o
n
u[n],
= 0 : X(z) = 1/(1- z
-1
)
LTI system: H(z) = N(z)/D(z) (assuming real poles)
= N(z)/(1-p
1
z
-1
)(1-p
2
z
-1
).. /(1-p
M
z
-1
)
all |p
k
| < 1, k = 1, 2, ., M
Output: Y(z) = X(z)H(z) = N(z)/(1-z
-1
)(1-p
1
z
-1
)(1-p
2
z
-1
)..(1-p
M
z
-1
)
= H(1) /(1- z
-1
) + A
1
/(1-p
1
z
-1
) + A
2
/(1-p
2
z
-1
) + .. A
M
/(1-p
M
z
-1
)
y[n] = {H(1) + A
1
p
1
n
+ A
2
p
2
n
+ .. A
M
p
M
n
}u[n]
as n y[n] H(1)u[n]
H(1) =
n=0
h[n] is the DC gain

YCJ
74
Pole/Zero Designs
First Order Filter
H(z) = G(1+bz
-1
)/(1-az
-1
)
H(0), H(), and speed of response n
eff
H(0) =G(1+b)/(1-a), H() = G(1-b)/(1+a)
H() / H(0) = (1-b)(1-a)/(1+b)(1+a)
a =
1/n
eff
Example
H(z) = (5+2z
-1
)/(1-0.8z
-1
)
= 5(1+0.4z
-1
)/(1-0.8z
-1
)
YCJ
75
Second-Order Filter & Resonator
Re
j
o
o
Re
-j
o
Z-Plane
Unit circle
H(z) = G/(1- Re
j
o
z
-1
)(1- Re
-j
o
z
-1
)
= G/(1-2Rcos(
o
)z
-1
+ R
2
z
-2
)
H() = G/(1- Re
j
o
e
-j
)(1- Re
-j
o
e
-j
)
G: normalize H(
o
) = 1
YCJ
76
|H()|
2
= G/(1- 2Rcos(
o
) + R
2
)(1- 2Rcos(+
o
) + R
2
)
|H()|
2
2(1-R)
(3 dB bandwidth)
1
1/2
o

YCJ
77
Transfer Function: H(z) = G/(1- 2Rcos(
o
) z
-1
+ R
2
z
-2
)
Impulse Response: h[n] = [G/sin(
o
)] R
n
sin(n
o
)
Difference Equation: y[n] = 2Rcos(
o
)y[n-1] - R
2
y[n-2] Gx[n]
G
-a
1
-a
2
y[n]
x[n]
z
-1
+
y[n-1]
z
-1
a
1
= -2Rcos(
o
)
a
2
= R
2
y[n-2]
YCJ
78
Example:
Design a 2-pole resonator filter with peak frequency
f
o
= 500 Hz, and 3-dB bandwidth f = 32 Hz.
The sampling frequency is f
s
= 10KHz.
R = 0.99, G = 0.0062,
a
1
= -1.8831 and a
2
= 0.9801
H(z) = 0.0062/(1 - 1.8831 z
-1
+ 0.9801 z
-2
)
YCJ
79
Parametric Equalizer
Re
j
o
o
Re
-j
o
Z-Plane
Unit circle
p
1
=R e
j
o
p
1
*
=R e
-j
o
z
1
=r e
j
o
z
1
*
=r e
-j
o
H(z) = (1- re
j
o
z
-1
)(1- re
-j
o
z
-1
) /(1- Re
j
o
z
-1
)(1- Re
-j
o
z
-1
)
= (1 + b
1
z
-1
+ b
2
z
-2
)/(1 + a
1
z
-1
+ a
2
z
-2
)
a
1
= -2Rcos(
o
), a
2
= R
2
b
1
= -2rcos(
o
), b
2
= r
2
H(z) = (1 + b
1
z
-1
+ b
2
z
-2
)/(1 + (R/r)b
1
z
-1
+ (R/r)
2
b
2
z
-2
)
YCJ
80
Parametric Equalizer
H(z) = (1 + b
1
z
-1
+ b
2
z
-2
)/(1 + (R/r)b
1
z
-1
+ (R/r)
2
b
2
z
-2
)
b
1
= -2rcos(
o
), b
2
= r
2
|H()|
2
= (1- 2rcos(
o
) + r
2
)(1- 2rcos(+
o
) + r
2
)
/(1- 2Rcos(
o
) + R
2
)(1- 2Rcos(+
o
) + R
2
)
|H()|
2

o
YCJ
81
Notch Filter
Re
j
o
o
Re
-j
o
Z-Plane
Unit circle
Notch Filter:
Zeros on the unit circle
r = 1
H(z) = (1 + b
1
z
-1
+ z
-2
)/(1 + (R)b
1
z
-1
+ R
2
z
-2
)
b
1
= -2cos(
o
),
YCJ
82
Notch Filter
Notch polynomial N(z) =
k
(1-e
j
k
z
-1
)
Real Coefficient N(z) =
k
(1-2cos
k
z
-1
+z
-2
)
Notch Filter = N(z)/N(
-1
z)
= (1 + b
1
z
-1
+ b
2
z
-2
++ b
2
z
-(M-1)
+ b
1
z
-M
)/
(1 + b
1
z
-1
+
2
b
2
z
-2
++
-(M-1)
b
2
z
-(M-1)
+
M
b
1
z
-M
)
Examples 6.4.3, 6.4.4
YCJ
83
Comb Filter
From notch filter moves the zeros inside the poles
H(z) = (1 + rb
1
z
-1
+ r
2
b
2
z
-2
++ r
-(M-1)
b
2
z
-(M-1)
+ r
M
b
1
z
-M
)/
(1 + b
1
z
-1
+
2
b
2
z
-2
++
-(M-1)
b
2
z
-(M-1)
+
M
b
1
z
-M
)
r and r <
YCJ
84
DTFT, DFT, and FFT
Fourier Transform X
a
() = x
c
(t)e
-jt
dt
Discrete Time Fourier Transform X() =
n
x[n] e
-jn
x[n] = x
c
(nT) X( T) =(1/T)X
a
()
Discrete Fourier Transform X [k] =
n=0
(N-1)
x[n] e
-jnk2/
If only N points X[k] = X()|
=k2/N
Fast Fourier Transform
YCJ
85
Digital Spectral Analysis
Frequency Resolution
Mainlobe
Sidelobe
Windowing
Physical Resolution
Computational Resolution
YCJ
86
The Fast Fourier Transform
Decimation in Time (DIT) Algorithm
DIT Butterfly
Decimation in Frequency (DIF) Algorithm
DIF Butterfly
N log(N) Algorithm
Bits Reversal Indexing

Typical Real Time DSP System

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Typical Real Time DSP System

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Digital Signal Processing

Digital Signal Processing

Digital Signal Processing

Digital Signal Processing

h[n] is the DC gain

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