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Digital Si l Processing

P i

CSSP Lab.

L b

2010

1

773

x 31538

Office Hour: Wed. 1:30 a.m.-3:30

a.m. 3:30 p.m.

bwu@mail.nctu.edu.tw & http://cssplab.cn.nctu.edu.tw

A. V. Oppenheim and R. W. Schafer, Discrete-Time

Signal Processing, Prentice Hall: New Jersy, 1999.

:&

917 x 54428

Office Hour : Monday 7:00-9:00

7:00 9:00 p.m.

pm

40 %

30 %

2 30

%

Important Date (Tentatively)

Homework

k due

d one weekk later

l

Midterm Exam

Exam. Nov.

Nov 11,

11 2010

Final Exam. Jan. 12, 2011

Raw Grade Post Jan. 19, 2011

Final

Fi l Grade

G d Post

P t Jan.

J 20 20, 2011

3

Contents

Part

P t I:

I Review

R i off Signals

Si l andd Systems

S t ( Chapter

Ch t 2-32 3)

1. DSP Systems and Applications

2. Introduction to Discrete-Time Signals

g and Processingg

3. The Z-transform

Part II: Transform Methods for DSP ( Chapter 4-5,8-9 )

1. Sampling Theory

2. Examples of Digital Filters

3. The Discrete Fourier Transform ( DFT )

4. Fast Fourier Transform

Part III: Digital Filter Design ( Text Chapter 6-7 )

1. Type of Filters --- FIR and IIR

2. Filter Design Techniques

4

Chapter 1

Introduction

5

Introduction

Signals

Speech( e.g. sampling frequency 8KHz:64Kbps 2.4Kbps),

Biomedical signals, Audio( e.g., MP3, sampling frequency

44 1KHz) and Sound,

44.1KHz) Sound Video and Image,

Image Radar signals

signals

Digital Signal Processing

Sample an analog signals Signals in digital form.

form

Process by digital processor.

Objectives

j

Remove interference or noise from a signal.

Obtain the spectrum of data.

Transform

T f the

h signal

i l to a suitable

i bl form.

f (e.g.

( Fourier

F i Transform,

T f

Wavelet Transform, Discrete Cosine Transform )

6

Introduction

Advantages

Guaranteed Accuracy: depend on number of bits need.

Perfect Reproducibility: e.g. digital recording.

Advantages is always taken of the advances in semiconductor

technology.

Flexibility: reprogrammable.

Disadvantages

Speed & cost: fast ADC/DAC are expensive. Bandwidth of 100

MHz increased are processed by analog signal method.

Design time: DSP hardware increasingly complicated.

Finite length effects: limited-bit number of DSP processor.

7

DSP Applications

pp

Image/Video

g Processingg

Pattern recognition, robot vision, facsimile, animation, data

compression

Controls

Spectrum analysis, position and velocity control of motors, noise

reduction

Speech/Audio

Speech recognition, speech/audio synthesis, text to speech, digital

audio, quantization

Tele-communications

T l i ti

Echo cancellation, adaptive equalization, video conference

Vehicle Electronics

Telematics, Navigation, Car Safety

Biomedical signal processing

Patient monitoring

monitoring, EEG , EKG analysis,

analysis Biomedical image

8 processing

DSP Applications

pp

i

Convolution

Correlation

Filtering

Discrete Transforms

DTFTDFT FFT DCT DWT Z-Transform

Modulation

9

Overview of Real-Time Signal Processing

f max fs

fs

10

Sampling Frequency

fs : Sampling frequency

f s 2 f max

fmax: Max frequency of the input signal

Sampling theorem: f s = 2 f max

Sampling rate Nyquist frequency

No distortion (ideally)

e.g.

e g Audio signal BB.W.:

W : 22KHz CD: 44

44.1KHz

1KHz

Speech signal B.W.: 4KHz 8K or 20KHz (recommended)

11

Sampling

p g Frequency

q y

xt xtpt x t

pt

Band-limited signals

12

Aliasing

f s f c < f c f s < 2 f c = 2 f max Aliasing

Anti-Aliasing Filter

Remove the undesired input signal or noise.

noise

13

For B-bit

B bit ADC:

B quantization noise error

SNR[Signal-to-Noise-ratio] 20log[ 1.5 2 B+1 ] in dB

CD quality

B Amin

i (dB)

(d )

8 56

10 68

12 80

CD qqualityy 6

16 104

0

14

Example

Ex.

E Determine

D t i fs to

t give

i an aliasing

li i error off less

l

than 2% of the signal level in the passband.

15

Solution:

1

j 2 fC 1 1

| H( f )| = = =

R+

1 1 + j 2 fRC 1

1 + (2 fRC ) 2 2

j 2 fC

1 1

= h fC =

where = 2 KHz

KH

f 2 2 RC

1+ ( )

fC

wideband spectrum (assured

aliasing).

16

1

KH X b

P b d @ 2 KHz

Passband = 00.7071

7071

2

2

desired aliasing level < 0.7071

0 7071 = 0.01414

0 01414

100

1

< 0.01414

fa 2

1+ ( )

2

f a > 141.4

141 4 KHz

f s > f C + f a = 2 + 141

141.44 = 143.4

143 4 KHz

f s ,say,

17

Quantization & Encoding

Q g

For example ADC 0804.

0804 0~5V, 8 bits

Quantization error

V fs

LSB= .

2 B

1 1 V fs 5V

Range of quantization error = LSB = = 9

2 22 B

2

18

xt xn

e x(t ) x(n)

/2

/2 1 1 3 2

= e p (e)de = e

2 2

= Uniformly distributed

3 / 2 12

e / 2

Signal-to-Noise

Signal to Noise Ratio (SNR)

signal power A2 2 2A

SNR = 10 log10 = 10 log 2 where = B

e2 12 2

= 10log(1.5 22 B ) = 6.02B + 1.76dB

ADC bits 6 dB / bit signal quality

19

Dynamic Range:

max signal power 2B

D (in dB) = = 20log = 20log2 B = 6.02 B

min signal

g power

p 1

For CD audio quality ( > 90dB) B = 16 16-bit ADC

Ex:

E

1

| H ( f ) |=

1 + ( f fC )6

3rd order 1. Minimum stopband attenuation, Amin, for the anti-

Butterworth aliasing filter.

filter

2. Minimum f S and aliasing error/signal in passband

3. Aliasingg error/signal

g in ppassband

20

Solution:

2 V fs

RMS quantization noise : = where =

12 2 3 2B

1. For ADC, it will attenuate a signal in stopband.

max passband signal level

stopband signal level

(Half power) 2 B

=( ) /( ) = 1.5 2 B +1

3dB Bandwidth 2 2 3

1 5 2 B +1 )

A min = 20log( 1.5

= 80dB = 6.02 B + 7.78 dB

21

1

2

2. Amin = 20log

20l , f C = 4 KHz

KH

1 + ( f fC ) 6

fS

f = 86.2 KHz f S 172.4 KHz, say f S = 173 KHz

2

22

3 Aliasing Level (AL):

3.

1 3 3.64 103

AL = = 3.64 10 = 0.515%

173 4 6 0.7071

1+ ( )

4

sample

p rate AL

23

DAC: signal

g recover

yn y (t ) yt

24

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