DSP-Chapter1 Student 28062015

Chapter 1
Sampling and Reconstruction

Nguyen Thanh Tuan, Click
M.Eng.
to edit Master subtitle style
Department of Telecommunications (113B3)
Ho Chi Minh City University of Technology
Email: nttbk97@yahoo.com
Content
Sampling
Sampling theorem
Spectrum of sampling signals
Anti-aliasing pre-filter
Ideal pre-filter
Practical pre-filter
Analog reconstruction
Ideal reconstructor
Practical reconstructor
Digital Signal Processing
1. Introduction
A typical signal processing system includes 3 stages:
The analog signal is digitalized by an A/D converter

The digitalized samples are processed by a digital signal processor.
The digital processor can be programmed to perform signal processing
operations such as filtering, spectrum estimation. Digital signal processor can be
a general purpose computer, DSP chip or other digital hardware.
The resulting output samples are converted back into analog by a

D/A converter.
2. Analog to digital conversion

Analog to digital (A/D) conversion is a three-step process.
x(t)
Sampler
t=nT
x(t)
x(nT)x(n) Quantizer xQ(n) Coder

A/D converter
x(n)
11010
111 xQ(n)
110
101
100
011
010
001
000
3. Sampling
Sampling is to convert a continuous time signal into a discrete time
signal. The analog signal is periodically measured at every T seconds
x(n)x(nT)=x(t=nT), n=-2, -1, 0, 1, 2, 3

T: sampling interval or sampling period (second);
Fs=1/T: sampling rate or frequency (samples/second or Hz)
Example 1
The analog signal x(t)=2cos(2t) with t(s) is sampled at the rate Fs=4
Hz. Find the discrete-time signal x(n) ?
Solution:
x(n)x(nT)=x(n/Fs)=2cos(2n/Fs)=2cos(2n/4)=2cos(n/2)
n
x(n)
-2
Plot the signal
Example 2
Consider the two analog sinusoidal signals
7
1
x1 (t ) 2cos(2 t ), x2 (t ) 2cos(2 t ); t ( s)
8
8
These signals are sampled at the sampling frequency Fs=1 Hz.
Find the discrete-time signals ?
Solution:
1
71
7
) 2cos(2
n) 2cos( n)
Fs
81
4
1
2cos((2 ) n) 2cos( n)
4
4
1
11
1
x2 (n) x2 (nT ) x2 (n ) 2cos(2
n) 2cos( n)
Fs
81
4
x1 (n) x1 (nT ) x1 (n
Observation: x1(n)=x2(n) based on the discrete-time signals, we

cannot tell which of two signals are sampled ? These signals are
called alias
F2=1/8 Hz
F1=7/8 Hz
Fs=1 Hz
Fig: Illustration of aliasing
4. Aliasing of Sinusoids
In general, the sampling of a continuous-time sinusoidal signal
x(t ) A cos(2 F0t ) at a sampling rate Fs=1/T results in a
discrete-time signal x(n).
The sinusoids xk (t ) A cos(2 Fk t ) is sampled at Fs , resulting
in a discrete time signal xk(n).
If Fk=F0+kFs, k=0, 1, 2, ., then x(n)=xk(n) .
Proof: (in class)

Remarks: We can that the frequencies Fk=F0+kFs are
indistinguishable from the frequency F0 after sampling and hence
they are aliases of F0
5. Spectrum Replication
Let x(nT ) x (t ) x(t ) (t nT ) x(t )s(t ) where s(t )

n
(t nT )
s(t) is periodic, thus, its Fourier series are given by

s (t )
Se
j 2 Fs nt
where Sn
1
1
1
j 2 Fs nt
(
t
)
e
dt
(
t
)
dt
T T
T T
T
1 j 2 Fsnt
Thus, s(t ) e
T n
1
x (t ) x(t ) s(t ) x(t )e j 2 nf st
which results in
T n
1
Taking the Fourier transform of x (t ) yields X ( F ) X ( F nFs )
T n
Observation: The spectrum of discrete-time signal is a sum of the

original spectrum of analog signal and its periodic replication at the
interval Fs.
10
Fs/2 Fmax
Fig: Spectrum replication caused by sampling

Fig: Typical badlimited spectrum
Fs/2 < Fmax
Fig: Aliasing caused by overlapping spectral replicas

11
6. Sampling Theorem
For accurate representation of a signal x(t) by its time samples x(nT),
two conditions must be met:
1) The signal x(t) must be band-limited, i.e., its frequency spectrum must
be limited to Fmax .
Fig: Typical band-limited spectrum
2) The sampling rate Fs must be chosen at least twice the maximum

Fs 2 Fmax
frequency Fmax.
Fs=2Fmax is called Nyquist rate; Fs/2 is called Nyquist frequency;
[-Fs/2, Fs/2] is Nyquist interval.
12
The values of Fmax and Fs depend on the application

Application
Fmax
Fs
Biomedical
1 KHz
2 KHz
Speech
4 KHz
8 KHz
Audio
20 KHz
40 KHz
Video
4 MHz
8 MHz
13
7. Ideal analog reconstruction
Fig: Ideal reconstructor as a lowpass filter

An ideal reconstructor acts as a lowpass filter with cutoff frequency
equal to the Nyquist frequency Fs/2.
T
An ideal reconstructor (lowpass filter) H ( F )
0
Then
F [ Fs / 2, Fs / 2]
otherwise
X a ( F ) X ( F )H ( F ) X ( F )
14
Example 3
The analog signal x(t)=cos(20t) is sampled at the sampling
frequency Fs=40 Hz.
a) Plot the spectrum of signal x(t) ?
b) Find the discrete time signal x(n) ?
c) Plot the spectrum of signal x(n) ?
d) The signal x(n) is an input of the ideal reconstructor, find the
reconstructed signal xa(t) ?
15
Example 4
The analog signal x(t)=cos(100t) is sampled at the sampling
frequency Fs=40 Hz.
a) Plot the spectrum of signal x(t) ?
b) Find the discrete time signal x(n) ?
c) Plot the spectrum of signal x(n) ?
d) The signal x(n) is an input of the ideal reconstructor, find the
reconstructed signal xa(t) ?
16
Remarks: xa(t) contains only the frequency components that lie in the
Nyquist interval (NI) [-Fs/2, Fs/2].
sampling at Fs
ideal reconstructor
x(t), F0 NI ------------------> x(n) ----------------------> xa(t), Fa=F0
sampling at Fs
ideal reconstructor
xk(t), Fk=F0+kFs-----------------> x(n) ---------------------> xa(t), Fa=F0
The frequency Fa of reconstructed signal xa(t) is obtained by adding
to or substracting from F0 (Fk) enough multiples of Fs until it lies
within the Nyquist interval [-Fs/2, Fs/2]. That is
Fa F mod( Fs )
17
Example 5
The analog signal x(t)=10sin(4t)+6sin(16t) is sampled at the rate 20
Hz. Find the reconstructed signal xa(t) ?
18
Example 6
Let x(t) be the sum of sinusoidal signals
x(t)=4+3cos(t)+2cos(2t)+cos(3t) where t is in milliseconds.
a) Determine the minimum sampling rate that will not cause any
aliasing effects ?
b) To observe aliasing effects, suppose this signal is sampled at half its
Nyquist rate. Determine the signal xa(t) that would be aliased with
x(t) ? Plot the spectrum of signal x(n) for this sampling rate?
19
Example 7
20
8. Ideal antialiasing prefilter

The signals in practice may not band-limited, thus they must be
filtered by a lowpass filter
Fig: Ideal antialiasing prefilter
21
9. Practical antialiasing prefilter

A lowpass filter: [-Fpass, Fpass] is the frequency range of interest for
the application (Fmax=Fpass)
The Nyquist frequency Fs/2 is in the middle of transition region.
The stopband frequency Fstop and the minimum stopband
attenuation Astop dB must be chosen appropriately to minimize the
aliasing effects.
Fs Fpass Fstop
Fig: Practical antialiasing lowpass prefilter

22
The attenuation of the filter in decibels is defined as

A( F ) 20log10
H (F )
(dB)
H ( F0 )
where f0 is a convenient reference frequency, typically taken to be at

DC for a lowpass filter.
10 =A(10F)-A(F) (dB/decade): the increase in attenuation when F is

changed by a factor of ten.
2 =A(2F)-A(F) (dB/octave): the increase in attenuation when F is
changed by a factor of two.
Analog filter with order N, |H(F)|~1/FN for large F, thus 10 =20N
(dB/decade) and 10 =6N (dB/octave)
23
Example 6
A sound wave has the form
x(t ) 2 A cos(10 t ) 2 B cos(30 t ) 2C cos(50 t )
2 D cos(60 t ) 2 E cos(90 t ) 2 F cos(125 t )
where t is in milliseconds. What is the frequency content of this

signal ? Which parts of it are audible and why ?
This signal is prefilter by an anlog prefilter H(f). Then, the output y(t)
of the prefilter is sampled at a rate of 40KHz and immediately
reconstructed by an ideal analog reconstructor, resulting into the final
analog output ya(t), as shown below:
24
Example 7
Determine the output signal y(t) and ya(t) in the following cases:
a)When there is no prefilter, that is, H(F)=1 for all F.
b)When H(F) is the ideal prefilter with cutoff Fs/2=20 KHz.
c)When H(F) is a practical prefilter with specifications as shown
below:
The filters phase response is assumed to be ignored in this example.

25
10. Practical analog reconstructors

sin( Fs t)
The ideal reconstructor has the impulse response: h(t )
which is not realizable since its impulse response is not casual Fs t
It is practical to use a
staircase reconstructor
26
Fig: Frequency response of staircase reconstructor
27
11. Anti-image postfilter

An analog lowpass postfilter whose cutoff is Nyquist frequency Fs/2
is used to remove the surviving spectral replicas.
Fig: Analog anti-image postfilter
Fig: Spectrum after postfilter

28
Review
Hot ng ca b ly mu l tng?
Hin tng chng ln?
Tnh cht lp ph?
Pht biu nh l ly mu?
Hot ng ca b khi phc l tng?
Ti sao phi dng tin lc/hu lc?
Hot ng ca b tin lc l tng/thc t?
29
Homework 1
30
Homework 2
31
Homework 3
32
Homework 3
33
Homework 4
34
Homework 5
35
Homework 6
36
Homework 7
37
Homework 8
38
Homework 9
39
Homework 10
40
Homework 11
Cho tn hiu ng vo tng t x(t) = 3cos103t 4sin104t (t: s) i qua
h thng ly mu v khi phc l tng vi tn s ly mu Fs = 8 KHz.
a) Vit biu thc ca tn hiu sau ly mu x[n]? Xc nh gi tr mu
x[n=2] ca tn hiu sau ly mu.
b) C hay khng 1 tn s ly mu khc (Fsb 8 KHz) cho cng kt qu
tn hiu sau ly mu x[n]? Nu khng, hy chng minh. Nu c, hy
ch ra 1 tn s ly mu khc .
c) V ph bin ca tn hiu sau ly mu trong phm vi tn s t 0 n
10 KHz.
d) Xc nh biu thc ca tn hiu sau khi phc.
e) Xc nh biu thc ca tn hiu sau khi phc trong trng hp dng
thm b tin lc thng thp thc t c bin phng trong tm [-4 4]
KHz v suy gim vi tc -1@0dB/decade bn ngoi di thng.
41
Homework 12
Cho tn hiu ng vo tng t x(t) = 2 4sin6t + 8cos10t (t: s)
i qua h thng ly mu v khi phc l tng vi tn s ly mu
la chn Fs = 7,@ KHz.
a) V ph bin ca tn hiu ng vo x(t).
b) V ph bin ca mt tn hiu chng ln (aliased signal) vi
x(t).
c) V ph bin ca tn hiu sau ly mu trong phm vi tn s t
0 n 10 KHz.
d) Tm gi tr mu x[n=2] ca tn hiu sau ly mu.
e) Xc nh biu thc (theo thi gian) ca tn hiu sau khi phc.
f) Tm iu kin ca tn s ly mu khi phc ng tn hiu ng
vo x(t).
42
Homework 13
Cho tn hiu ng vo tng t x(t) = 14sin23t + 3sin14t (t: ms) i qua
h thng ly mu v khi phc l tng vi tn s ly mu Fs = 8 KHz.
a) Tm gi tr mu x[n=4] ca tn hiu sau ly mu?
b) Xc nh biu thc ca 1 tn hiu chng ln (aliased signal) vi tn
hiu ban u x(t)?
c) V ph bin ca tn hiu sau ly mu trong phm vi tn s t 0 n
8 KHz?
d) Xc nh biu thc ca tn hiu sau khi phc?
e) Xc nh biu thc ca tn hiu sau khi phc trong trng hp dng
thm b tin lc thng thp thc t c bin phng trong tm 4 KHz
v suy gim vi tc -4@dB/decade bn ngoi di thng?
f) Xc nh 1 tp gi tr thch hp (A, B, FA FB) ca tn hiu ng vo
x(t) = AsinFAt + BsinFBt
(t: ms) tn hiu sau khi phc (khi
khng dng thm b tin lc) y(t) = 2sin2t (t: ms)?
43
Homework 14
Cho tn hiu ng vo tng t x(t) = 1 2cos6t + 3sin14t (t: ms) i qua h
thng ly mu v khi phc l tng vi tn s ly mu Fs = 8 KHz.
a) Tm gi tr mu x[n=2] ca tn hiu sau ly mu?
b) Xc nh biu thc (theo thi gian) ca 1 tn hiu chng ln (aliased signal) vi
tn hiu ban u x(t)?
c) V ph bin ca tn hiu sau ly mu trong phm vi tn s t 0 n 8 KHz?
d) Xc nh biu thc (theo thi gian) ca tn hiu sau khi phc?
e) Xc nh biu thc (theo thi gian) ca tn hiu sau khi phc trong trng hp
dng thm b tin lc thng thp thc t c bin phng trong tm 4 KHz v
suy gim vi tc -6@dB/decade bn ngoi di thng?
f) Tm iu kin ca chu k ly mu Ts sao cho tn hiu sau khi phc (khi khng
dng thm b tin lc) ging tn hiu ban u x(t)?
g) Tm tn s ly mu Fs ln nht c th sao cho tn hiu sau khi phc (khi khng
dng thm b tin lc) l tn hiu mt chiu khng i. Xc nh gi tr mt
chiu khng i ny?
44
Homework 15
45

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Chapter 1

Sampling and Reconstruction

Digital Signal Processing

Sampling and Reconstruction

The analog signal is digitalized by an A/D converter

The resulting output samples are converted back into analog by a

Sampling and Reconstruction

2. Analog to digital conversion

x(nT)x(n) Quantizer xQ(n) Coder

Digital Signal Processing

Sampling and Reconstruction

x(n)x(nT)=x(t=nT), n=-2, -1, 0, 1, 2, 3

Sampling and Reconstruction

Plot the signal

Digital Signal Processing

Sampling and Reconstruction

Observation: x1(n)=x2(n) based on the discrete-time signals, we

Sampling and Reconstruction

Fig: Illustration of aliasing

Digital Signal Processing

Sampling and Reconstruction

Proof: (in class)

Sampling and Reconstruction

Let x(nT ) x (t ) x(t ) (t nT ) x(t )s(t ) where s(t )

s(t) is periodic, thus, its Fourier series are given by

Observation: The spectrum of discrete-time signal is a sum of the

Sampling and Reconstruction

Fig: Spectrum replication caused by sampling

Fig: Aliasing caused by overlapping spectral replicas

Sampling and Reconstruction

Fig: Typical band-limited spectrum

2) The sampling rate Fs must be chosen at least twice the maximum

Sampling and Reconstruction

The values of Fmax and Fs depend on the application

Digital Signal Processing

Sampling and Reconstruction

7. Ideal analog reconstruction

Fig: Ideal reconstructor as a lowpass filter

Digital Signal Processing

Sampling and Reconstruction

Digital Signal Processing

Sampling and Reconstruction

Digital Signal Processing

Sampling and Reconstruction

Sampling and Reconstruction

Digital Signal Processing

Sampling and Reconstruction

Digital Signal Processing

Sampling and Reconstruction

Digital Signal Processing

Sampling and Reconstruction

8. Ideal antialiasing prefilter

Fig: Ideal antialiasing prefilter

Digital Signal Processing

Sampling and Reconstruction

9. Practical antialiasing prefilter

Fig: Practical antialiasing lowpass prefilter

Sampling and Reconstruction

The attenuation of the filter in decibels is defined as

where f0 is a convenient reference frequency, typically taken to be at

10 =A(10F)-A(F) (dB/decade): the increase in attenuation when F is

Sampling and Reconstruction

where t is in milliseconds. What is the frequency content of this

Digital Signal Processing