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# International University of Ho Chi Minh City

DIGITAL SIGNAL
PROCESSING

## Instructor: HUYNH KHA TU, MEng.

Email: hktu@hcmiu.edu.vn
Cellphone: 0916656656
Digital Signal Processing

## Code of module: IT07IU

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
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

## 1.1.1. Signals, systems and Digital signal

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

## image, speech, electrocardiogram (ECG),

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

## Digital signal processing

1.2. Basic elements of a DSP
systems

## Analog Digital Analog

input A/D Signal D/A output
signal Converter Processo Converter signal
r

Digital Digital
input output
signal signal

## A large programmable digital computer;

a small microprocessor; hardwired digital processo
or software
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

## n analog signal is represented in term of a functio

time, for example, x(t).

## he frequency spectrum of x(t) is:

+∞
X ( Ω) = ∫ x ( t )e − jΩt
dt
−∞

1.2. Overview of Analog
signal

## Consider the response of a linear system

y(t)
x(t)
Linear system h(t) output
input

## is system is characterized by the impulse response h(t)

( )( )
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

## x(t) = e jΩt y(t) = H(Ω).ejΩt

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

## After passing through the filter

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

## Analog Discrete Quantize Digital

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.

## Coding: each discrete value xq(n) is represented

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

## riodic sampling establishes a relationship betwee

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

## With an analog signal x(t) having maximum

frequency fmax, the sampling rate is selected
so that
fs ≥ 2.fmax

## fs/2 : Nyquist frequency or folding frequency.

1.3.2. The sampling
Theorem

Example:
Consider the analog signal:
xa(t) = 3Cos50πt + 100Sin300πt – Cos100πt
What is the Nyquist rate for this signal?

## Nyquist rate is: 300Hz.

1.3.2. The sampling
Theorem

## However, in practical, most signals is not limited

n a band, they are often passed a low-pass filter
before sampling. The use of the filter here can
avoid the spectrum aliasing.

## x(t) Analog x(t) Sampler and x(nT)

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:

## + Omit all frequencies beyond fs/2.

+ 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:

## + Can not omit all frequencies beyond fs/2.

⇒ There are aliases.

## ⇒ Design a suitable filter can reduce the aliase

to minimum.
+ Consider the attenuation.
1.3.2. The sampling
Theorem

## he attenuation A dB means H(f) decreases 10-A/20

or example, from the border of folding frequency fs/2
(f) decreases A dB means:

H( f )
= 10 − A / 20
H ( f s / 2)

## We often assume that |H(fs/2)| = 1 and

gnore the effect of the phase response of the filte
1.3.2. The sampling
Theorem

## What happens if fs <

2.fmax?
1.3.2. The sampling
Theorem

xample:
onsider the analog signal:
a(t) = 3Cos2000πt + 5Sin6000πt + 10Cos12000π

## . What is the Nyquist rate for this signal?

. If fs = 5000, find the disctere-time signal
obtained after sampling?
. Find the signal ya(t) after reconstructing.
1.3.2. The sampling
Theorem

## sed on the sampling theory, we can use

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

## Cancel the frequency components outside

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

## Keep only the frequency components belonging

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

## e process of converting a discrete-time continuo

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)]

## xq(n): the sequence of quantized samples at the

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

## x(n) xq(n) xq(n) eq(n) = xq(n) – x(n)

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 value allowed in the digital signal are called

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

## Quantization of sinusoidal signals

Consider an analog sinusoid signal
xa(t) = ACosΩot ⇒x(n) = xa(nT)
1.3.3. Quantization
1.3.3. Quantization
1.3.3. Quantization

## The mean-square error power Pq is:

τ τ
1 1 2
Pq = ∫ eq ( t ) dt = ∫ eq ( t ) dt
2

2τ −τ τ0

## with τ: the time that xa(t) stays within the

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

## The average power of the signal xa(t) is:

Tp
1 A2
Px =
Tp ∫0 ( ACosΩ o t )dt = 2
1.3.3. Quantization

## The quality of the output of the A/D converter

s usually measured by the signal-to-quantization
noise ratio (SQNR)
Px 3 2b
SQNR = = .2
Pq 2
Expressed in decibels (dB)

## SQNR (dB ) = 10 log10 SQNR = 1.76 + 6.02b

1.3.3. Quantization

## he dynamic range of the quantizer (in dB)

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
∆

## 6.b is the dynamic range in dB

1.3.4. Coding of Quantized
samples

## The coding process in an A/D converter assigns

a unique binary number to each quantization leve

## If we have L levels, we need at least L different

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

## This is the process converting a digital signal in

an analog signal.

## All D/A converters connect the dots in a digital

signal by performing some kind of interpolation
whose accuracy depends on the quality of the
D/A conversion process.