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Advanced Digital Signal Processing

Office723, TEL 33669652
E-mail: jjding@ntu.edu.tw
http://djj.ee.ntu.edu.tw/ADSP.htm

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Term paper 25 scores

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abstract, conclusion,
references sectionssubsections References
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(4) Wikipedia
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Tutorial ( 17 )

(1) Kalman Filters

(2) Particle Filters
(3) Multiple Dimensional Filter Design

(4) Image Denoising

(5) Compressive Sensing
(6) Singular Value Decomposition
(7) Infrasound
(8) Underwater Acoustic Communication
(9) Face Detection and Adaboost
(10) Face Recognition
(11) Recent Development of MIMO System Analysis
(12) Corner Detection

Tutorial ( 17 )

(13) Recent Development of CDMA

(14) Superpixel for Image Segmentation
(15) Neural Network

(16) Ultrasonic Image Analysis

(17) The Optimal Way for Implementing the Convolution in Different Cases

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3/6,

5/8, HW3

3/13,

5/15,

3/20, HW1

5/22, HW3

3/27,

5/29, HW4

4/10, HW1

6/5,

4/17, HW2

6/12, HW4, HW5

4/24, Oral
5/1, HW2
7/3, HW5 term paper
2/27, 4/3, 6/19
: 3n , 3n+2

Matlab
Download:
http://comm.ntu.edu.tw/matlab/request.php

Matlab 2011.
Matlab 7 2010.

Matlab2007.
-Matlab2005.

3~5

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Question:

Why should we use the Fourier transform?

Is the Fourier transform the best choice in any condition?

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I. Introduction
Contents:

(1) Introduction

1W

(2) Digital Filter Design (A)

2W

(3) Digital Filter Design (B)

2W

(4) Homomorphic Signal Processing

1W

(5) Applications (A):

Acoustics

1W

(6) Applications (B):

Data Compression, Others

1W

(7) Fast Algorithms:

Basic, FFT, and Convolution

3W

(8) Orthogonal Transform (A): Walsh Transform

1W

(9) Orthogonal Transform (B): Number Theoretic Transform

1W

OFDM, CDMA , and Others

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(1) Digital Signal Processing

(2) Digital Signal Processing

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Part 1: Filter
Filter

digital

IIR
(aliasing)
MSE (mean square error)

filter
FIR

minimax
frequency sampling

analog
()

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IIR filter (1) easy to design
(2) (sometimes) easy to implement

FIR filter

An FIR filter is impossible to have the ideal frequency

response of

Part 2: Homomorphic Signal Processing

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convolution addition

Part 3: Applications of DSP

filter design, data compression (image, video, text), acoustics (speech, music),
image analysis (structural similarity, sharpness), 3D accelerometer

 Part 4: Fast Algorithms

Basic Implementation Techniques
Example: one complex number multiplication
= ? Real number multiplication.

Trade-off: Multiplication takes longer than addition

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FFT and Convolution

Due to the Cooley-Tukey algorithm (butterflies),
the complexity of the FFT is:

The complexity of the convolution is:

three DFTs

O( N log 2 N ) ?

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Part 5: Orthogonal Transforms

DFT :
N 1

Question: DFT

DFT x[n] x n e
n 0

Walsh Transform
(CDMA)
Number Theoretic Transform

Orthogonal Frequency-Division Multiplexing (OFDM)

Code Division Multiple Access (CDMA)

2 m n
N

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Review 1: Four Types of the Fourier Transform

(1) Fourier Transform

X f x t e

j 2 f t

x t

dt ,

X f e j 2 f t df

Alternative definitions

X x t e

j t

dt

1
x t
2

X e j t d

(2) Fourier series (suitable for period function)

X [ m] x t e
T

2 m
t
T

x t T

dt

X [ m] e

T:
m f

m
T

2 m
t
T

(3) Discrete-time Fourier transform (DSP )

Xf

x n e

j 2 f nt

1/ t

, x n t 0

X f e j 2 f n t df

t : sampling interval

x n t
2

x n e j n t

2 / t

X e j n t d

(4) Discrete Fourier transform (DFT) (DSP )

N 1

X m x n e
n 0

2 m n
N

2 m n
j
1 N 1
x n X m e N
N m 0

m f

m
m
fs
N t N

where fs = 1/t (sampling frequency)

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Fourier transforms

time domain
(1) Fourier transform continuous, aperiodic
(2) Fourier series

continuous, periodic

frequency domain
continuous, aperiodic
discrete, aperiodic

(or continuous, only the value

in a finite duration is known)
(3) discrete-time
Fourier transform
(4) discrete Fourier
transform

discrete , aperiodic

continuous, periodic

discrete, periodic

discrete, periodic

(or discrete, only the value

in a finite duration is known)

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Review 2: Normalized Frequency

(1) Definition of normalized frequency F:
F

t
f
f t
fs
2

where fs = 1/t (sampling frequency)

t : sampling interval

(2) folding frequency f0

f
f0 s
normalized frequency
2
folding frequency = 1/2

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For the discrete time Fourier transform
(1) G(f) = G(f + fs)

i.e., G(F) = G(F + 1).

(2) If g[n] is real

G(F) = G*(F) (* means conjugation)

G(F) for 0 F ( 0 < f < f0)

G(F)
(3) If g[n] = g[n] (even)
g[n] = g[n] (odd)

Analog
filter: H(f)

G(F) = G(F),
G(F) = G(F)

 Discrete time Fourier transform of the lowpass, highpass, and band pass filters
low pass filter (pass band fS )

fs (F = 1)

0 (F = 0)

fs (F = 1)

high pass filter

fs (F = 1)

F = 0.5

0 (F = 0)

F = 0.5

fs (F = 1)

band pass filter

fs (F = 1)

F = 0.5

0 (F = 0)

F = 0.5

fs (F = 1)

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Review 3: Z Transform and Laplace Transform

Z-Transform
suitable for discrete signals

G z

g n z

Compared with the discrete time Fourier transform:

G f

g n e

j 2 f n t

z e j 2 f t

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Laplace Transform
suitable for continuous signals

One-sided form

G s g (t )e st dt
0

Two-sided form

G s g (t )e st dt

Compared with the Fourier transform:

G f g t e j 2 f t dt

s j 2 f

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Review 4: IIR Filter Design

Two types of digital filter:

(1) IIR filter (infinite impulse response filter)

(2) FIR filer (finite impulse response filer)

There are 3 popular methods to design the IIR filter.

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Method 1: Impulse Invariance
sampling

analog filter ha(t)

digital filter h[n]

h n ha nt

Advantage :

Simple

Disadvantage : (1) infinite

(2)

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Method 2: Step Invariance
step function response sampling
analog filter ha(t)

digital filter h[n]

step function (continuous form)

1
u(t)
0
t=0
step function (discrete form)
1 1 1
u[n]
0

0
n=0

Laplace transform of u(t):

1
s
Fourier transform of u(t):
1
j 2 f
1

Z transform of u[n]:
1
1 z 1

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Step 1 Calculate the convolution of ha(t) and u(t)

ha ,u t ha t u t ha u t d ha d
H a ,u ( f )

Ha ( f )
j 2 f

( ha(t) )

Step 2 Perform sampling for ha,u(t)

hu n ha ,u nt

Step 3 Calculate h[n] from h n hu n hu n 1

Note: Since

so

hu n h n u n

Hu z

1 H z
1 z 1
H z 1 z 1 H u z

h n hu n hu n 1

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Advantages of the step invariance method:

Disadvantages of the step invariance method:

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Method 3: Bilinear Transform
Suppose that we have known an analog filter ha(t) whose frequency
response is Ha(f).
To design the digital filter h[n] with the frequency response H(f),
H f new H a f old

fold (, )
fnew (fs/2, fs/2)
fs = 1/t (sampling frequency)

The relation between fnew and fold is determined by the mapping function
1 z 1
sc
1 z 1

s: index of the Laplace transform

z: index of the Z transform

c: some constant

1 z 1
sc
1 z 1

j 2 f old

s j 2 fold

ze

j 2 f new t

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1 e j 2 fnew t
e j fnew t e j fnew t
c
c j fnew t
j 2 f new t
1 e
e
e j fnew t
j sin f new t
c
cos f new t

2 f old c tan f new t

f new

1
2
f
2

atan
f old s atan
f old
t
c

c

Suppose that the Laplace transform of the analog filter ha(t) is Ha,L(s)
The Z transform of the digital filter h[n] is Hz(z)
H z z H a,L

1
1

z
c
1 z 1

f new

atan
fold

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fs

fold

fnew
0.5

c = 2

fnew/fs

-0.5
-8

fold
-6

-4

-2

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analog filter
Ha(f)

fc

fc

digital filter
H(f)
fs/2

fc,1

fc,1

fc ,1
Advantage of the bilinear transform

Disadvantage of the bilinear transform

fs/2

2
atan
fc

fs

 DSP
(1) Concepts:
(2) Comparison:

(3) Advantages:
(3-1) Why?
(4) Disadvantages:
(4-1) Why?
(5) Applications:
(6) Innovations:

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