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You are on page 1of 49

DIGITAL SIGNAL

PROCESSING

Email: hktu@hcmiu.edu.vn

Cellphone: 0916656656

Digital Signal Processing

Credits: 3

Pre-requisite: Signals & Systems,

DLD

Assessment:

Assignment : 20%

Mid-term test : 20%

Final examination : 60%

Digital Signal Processing

References:

2. John G.Proakis, Dimitris

G.Manolakis, “Digital Signal

Processing: Principles, Algorithms

and Application”, Prentice Hall

1996, Upper Saddle River, New

Jersey 07458, ISBN-0133737624.

4. Lecture notes

Digital Signal Processing

Contents:

1. Sampling, Quantization and

Reconstruction

2. D/A and A/D converters

3. Discrete-time systems

4. Finite-impulse response and

convolution

5. Z-transform

Chapter 1:

Sampling, Quantization and Reconstruction

Introduction

Overview of Analog signal

Analog to digital conversion

Digital to analog conversion

1.1. Introduction

processing

A signal is defined as any physical quantity that

varies with time, space, or any other independen

variable or variables.

Mathematically, we describe a signal as a functio

of one or more independent variables.

1.1. Introduction

precisely

For example: defined

s1(t) = 9t ;

s2(t) = 20t2 ;

s3(x,y) = 3x + 2xy + 10y2

or

electroencephalogram (EEG) signals, …

1.1. Introduction

A system is defined as a physical device

that performs an operation or software

realizations of operations on a signal.

Passing a signal through a system we

have processed the signal.

This course consider the processing of signal by

digital means

1.2. Basic elements of a DSP

systems

input A/D Signal D/A output

signal Converter Processo Converter signal

r

Digital Digital

input output

signal signal

a small microprocessor; hardwired digital processo

or software

1.1.3. Advantages of Digital over

Analog Signal Processing

Flexibility in configuring

Accuracy requirements

The easy storage

Ability to implement sophisticated signal

processing algorithms

The low cost

1.2. Overview of Analog

signal

time, for example, x(t).

+∞

X ( Ω) = ∫ x ( t )e − jΩt

dt

−∞

1.2. Overview of Analog

signal

y(t)

x(t)

Linear system h(t) output

input

∞

( )( )

y ( t ) = ∫ h t − t ' x t ' dt or Y(Ω) = H(Ω).X(Ω)

−∞

1.2. Overview of Analog

signal

H(Ω) is the frequency response of the system

∞

H ( Ω ) = ∫ h( t ) e − jΩt

dt

−∞

* If input is a sinusoid

Linear system H(Ω)

input output

H(Ω) = |H(Ω)|.ejargH(Ω)

1.2. Overview of Analog

signal

The function of the Linear filter:

In time-domain:

x( t ) = e j Ωt

⇒ y( t ) = H ( Ω ) e j Ωt

= H ( Ω) e jΩt + j arg H ( Ω )

If x( t ) = A1 .e jΩ1t

+ A2 .e jΩ 2 t

+ ... + An .e jΩ n t

y ( t ) = A1 .H ( Ω ).e jΩ1t + A2 .H ( Ω ).e jΩ2t + ... + An .H ( Ω ).e jΩnt

The filter changes the magnitudes only,

not the frequencies of the signal.

1.2. Overview of Analog

signal

In frequency domain:

X ( Ω ) = 2πA1δ ( Ω − Ω1 ) + 2πA2δ ( Ω − Ω 2 ) + ... + 2πAnδ ( Ω − Ω n )

H(Ω).X(Ω)

X(Ω)

A1 A2 H(Ω) An

A1Ḥ(Ω1)

A2Ḥ̣(Ω2)

AnH(Ωn)

Ω Ω

Ω1 Ω2 Ωn Ω1 Ω2 Ωn

Y ( Ω ) = H ( Ω ). X ( Ω ) = H ( Ω ).[ 2πA1δ ( Ω − Ω1 ) + 2πA2δ ( Ω − Ω 2 ) + ... + 2πAnδ ( Ω − Ω n ) ]

Y ( Ω ) = 2πA1 H ( Ω1 )δ ( Ω − Ω1 ) + 2πA2 H ( Ω 2 )δ ( Ω − Ω 2 ) + ... + 2πAn H ( Ω n )δ ( Ω − Ω n )

1.3. Analog to digital

conversion

A/D

Converter

r

signal -time d signal signal

signal

1.3. Analog to digital

conversion

Sampling: convert a continuous-time signal into

a discrete-time signal.

x(t) → x(nT) ≡ x(n) ;

T: sampling interval

Quantization: the conversion of a discrete-time

ntinuous valued signal into a discrete-time discre

lued (digital) signal.

a b-bit binary sequence.

1.3.1. Sampling process

Ideal

sampler

x x(nT

(t) )

Analog Sample

signal d signal

x(t) x(nT)

t t

T

T: sampling period

fs = 1/T : the sampling rate

1.3.1. Sampling process

e time variables t and n of continuous-time and

screte-time signals:

n

t = nT =

fs

x(t) → x(nT)

1.3.2. The sampling

Theorem

frequency fmax, the sampling rate is selected

so that

fs ≥ 2.fmax

1.3.2. The sampling

Theorem

Example:

Consider the analog signal:

xa(t) = 3Cos50πt + 100Sin300πt – Cos100πt

What is the Nyquist rate for this signal?

1.3.2. The sampling

Theorem

n a band, they are often passed a low-pass filter

before sampling. The use of the filter here can

avoid the spectrum aliasing.

quantizer To DSP

lowpass filter

Analog Analog

signal signal

Bandlimite

d signal

1.3.2. The sampling

Theorem

There are two kind of anti-aliasing pre-filters:

ideal and practical pre-filter.

Ideal anti-aliasing pre-filter:

+ H(f) = 1 (or H(Ω) = 1) for all f ∈[-fs/2 ;

fs/2]

(or Ω ∈ [-Ω/2; Ω/2])

1.3.2. The sampling

Theorem

Practical anti-aliasing pre-filter:

⇒ There are aliases.

to minimum.

+ Consider the attenuation.

1.3.2. The sampling

Theorem

or example, from the border of folding frequency fs/2

(f) decreases A dB means:

H( f )

= 10 − A / 20

H ( f s / 2)

gnore the effect of the phase response of the filte

1.3.2. The sampling

Theorem

2.fmax?

1.3.2. The sampling

Theorem

xample:

onsider the analog signal:

a(t) = 3Cos2000πt + 5Sin6000πt + 10Cos12000π

. If fs = 5000, find the disctere-time signal

obtained after sampling?

. Find the signal ya(t) after reconstructing.

1.3.2. The sampling

Theorem

ideal re-constructor to recover the original signa

Ideal re-

constructor

x(t) x(nT) x(t)

Ideal

sampler

Analog Analog

-fs/2 fs/2

signal signal

Rate fs

1.3.2. The sampling

Theorem

s/2 ; fs/2] of signal x(nT)

s/2 ; fs/2]

1.3.2. The sampling

Theorem

From a set of frequency [ f , f ± fs , f ± 2fs , … ],

there is only one frequency fa belonging to

[-fs/2 ; fs/2].

How to find fa?

Calculate fa=f mod (fs) until fa ∈ [-fs/2 ; fs/2].

+ fa = f if and only if f ∈ [-fs/2 ; fs/2]

+ If f ∉ [-fs/2 ; fs/2] ⇒ fa ≠ f

⇒ xa(t) ≠ x(t)

although xa(nT) = x(nT)

1.3.2. The sampling

Theorem

xample:

onsider the sinusoidal xa(t) = A Cos20t [Hz].

When sampling xa(t) with fs = 14Hz, the sampled

gnal xa(nT) will cover periodic frequencies

0 + m.14Hz, but only fa=10mod14 = -4 ∈ Nyquist

terval [-7 ; 7]

f = -4 ⇒ xa(t) = ACos(-8πt) ≠ ACos20t.

If we choose fs = 22Hz ≥2.f = 20Hz

we will have the reconstructed signal with 10Hz

1.3.2. The sampling

Theorem

Example:

Consider the signal:

x(t) = 4 + 3Cosπt + 2Cos2πt + Cos3πt t[ms]

a. Find fs so that there is no alias.

b. Supposing that x(t) is sampled with fs equal a

half of Nyquist rate, find xa(t) which is alias of x(t

1.3.2. The sampling

Theorem

onsider the following sound wave, where t is in millisecond

x(t) = Sin(20πt) + Sin(30πt) + Sin(80πt).

his is pre-filtered by an analog anti-aliasing pre-filter H(f)

nd then sampled at frequency rate fs = 40KHz. The resultin

amples are immediately reconstructed using an ideal

e-constructor. Determine the output ya(t) of the re-constru

the following cases and compare it with the original x(t).

1. When there is no pre-filter (H(f) = 1).

2. When H(f) is an ideal pre-filter with cut off of

20KHz.

3. When H(f) is a practical pre-filter that has a flat

pass-band up to 20KHz and attenuates at a rate

of 48dB/octave beyond 20KHz. Ignore the effects

1.3.3. Quantization

mplitude signal into a digital signal by expressing

ch sample value as a finite (instead of an infinite

mber of digits, is called quantization.

xq(n) = Q[x(n)]

output of the quantizer.

eq(n) = xq(n) – x(n) : quantization error.

1.3.3. Quantization

Example:

Consider the analog exponential signal

xa(t) = (0.9)t , t ≥ 0.

Sampling xa(t) at the sampling frequency fs = 1H

we have: T = 1/fs = 1.

⇒ x(nT) = x(n) = (0.9)n ; n ≥ 0.

1.3.3. Quantization

1.3.3. Quantization

Consider the first 10 samples of x(n).

Numerical Illustration of Quantization with one significant digit using

truncation or rounding

n

Discrete-time signal (Truncation) (Rounding) (Rounding)

0 1 1.0 1.0 0.0

1 0.9 0.9 0.9 0.0

2 0.81 0.8 0.8 -0.01

3 0.729 0.7 0.7 -0.029

4 0.6561 0.6 0.7 0.0439

5 0.59049 0.5 0.6 0.00951

6 0.531441 0.5 0.5 -0.031441

7 0.4782969 0.4 0.5 0.0217031

8 0.43046721 0.4 0.4 -0.03046721

9 0.387420489 0.3 0.4 0.012579511

1.3.3. Quantization

1.3.3. Quantization

the quantization levels.

+ The distance ∆ between two successive

quantization levels is called the quantization

step size or resolution.

+ The quantization error eq(n)

∆ ∆

− ≤ eq ( n ) ≤

2 2

1.3.3. Quantization

Consider an analog sinusoid signal

xa(t) = ACosΩot ⇒x(n) = xa(nT)

1.3.3. Quantization

1.3.3. Quantization

1.3.3. Quantization

τ τ

1 1 2

Pq = ∫ eq ( t ) dt = ∫ eq ( t ) dt

2

2τ −τ τ0

quantization levels.

τ 2

∆ 1 ∆ 2 ∆2

Since eq ( t ) = t , − τ ≤ t ≤ τ ⇒ Pq = ∫ t dt =

2τ τ 0 2τ 12

1.3.3. Quantization

The root mean-square error erms is:

∆

erms =

12

If the quantizer has b bits of accuracy and the

quantizer covers the entire range 2A,

the quantization step is:

2A

∆= b

2

2

A 1

⇒ Pq = . 2b

3 2

1.3.3. Quantization

Tp

1 A2

Px =

Tp ∫0 ( ACosΩ o t )dt = 2

1.3.3. Quantization

s usually measured by the signal-to-quantization

noise ratio (SQNR)

Px 3 2b

SQNR = = .2

Pq 2

Expressed in decibels (dB)

1.3.3. Quantization

he full scale range R of a A/D converter is divided

ually to 2b quantization levels. The quantization step

R R

∆= b =2 b

2 ∆

R

20 log10 = 20 log10 ( 2b ) = b.20 log10 2 = 6.b

∆

1.3.4. Coding of Quantized

samples

a unique binary number to each quantization leve

binary numbers. With a word length of b bits,

we can create 2b different binary numbers.

⇒ 2b ≥ L or b ≥ log2L.

1.4. Digital to analog

conversion

an analog signal.

signal by performing some kind of interpolation

whose accuracy depends on the quality of the

D/A conversion process.

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