Chapter 1

Sampling and Reconstruction

Nguyen Khanh Loi, M.Eng.
nkloi@hcmut.edu.vn
 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 2 Sampling and Reconstruction

 1. Introduction

❖ A typical signal processing system includes 3 stages:

❖ The analog signal is sampled and each sample is quantized to a finite

number of bits (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 an
analog reconstructor (D/A converter).
Digital Signal Processing 3 Sampling and Reconstruction
 2. Analog to digital conversion

❖ Analog to digital (A/D) conversion is a three-step process.

x(t) Sampler x(nT)≡x(n) xQ(n) 11010

Quantizer Coder
t=nT
A/D converter

x(t) x(n) 111 xQ(n)

110
101
100
011
t n 010 n
001
000

Digital Signal Processing 4 Sampling and Reconstruction

 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)

Digital Signal Processing 5 Sampling and Reconstruction

 Example 1

❖ The analog signal x(t)=2cos(2πt) 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(2πn/Fs)=2cos(2πn/4)=2cos(πn/2)
n 0 1 2 3 4
x(n) 2 0 -2 0 2

❖ Plot the signal

Digital Signal Processing 6 Sampling and Reconstruction

 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
x1 (n)  x1 (nT ) = x1 (n ) = 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
❖ 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”
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 Example 2

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 Example 2

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 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:

❖ Remarks: We can that the frequencies Fk=F0+kFs are

indistinguishable from the frequency F0 after sampling and hence
they are aliases of F0
Digital Signal Processing 10 Sampling and Reconstruction
 5. Spectrum Replication

Xc(f)
1

f
-fc fc
S(f)
1/T

f
-3fs -2fs -fs fs 2fs 3fs
Xs(f)
1/T

f
-3fs -2fs -fs -fc fc fs 2fs 3fs

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 5. Spectrum Replication

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


1 1 1
s (t ) = Se
n =−
n
j 2 Fs nt
where Sn =   (t )e
TT
− j 2 Fs nt
dt =   (t )dt =
TT T
1  j 2 Fsnt
Thus, s(t ) =  e
T n=−
1 
which results in x (t ) = x(t )s(t ) =  x(t )e j 2 nfst
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.
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❖ Fs/2 ≥ Fmax

Fig: Spectrum replication caused by sampling

Fig: Typical badlimited spectrum

❖ Fs/2 < Fmax

Fig: Aliasing caused by overlapping spectral replicas

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

Fs  2 Fmax

• Fs=2Fmax is called Nyquist rate;

• Fs/2 is called Nyquist frequency
• [-Fs/2, Fs/2] is Nyquist interval. Fig: Typical band-limited spectrum

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 ❖ 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

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 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 F  [ − Fs / 2, Fs / 2]
❖ An ideal reconstructor (lowpass filter) H ( F ) = 
0 otherwise
Then
X a ( F ) = X (F )H (F ) = X (F )

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 7. Ideal analog reconstruction
s(t)=Ʃб(t-nT)

xc(t) xs(t) xr(t)

X Hr(f)

Xc(f)
1

-fc fc f
Hr(f) Xs(f)
1/T

-fs -fc fc fs f
Xr(f)
1

-fc fc f

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 Example 3

❖ The analog signal x(t)=cos(20πt) [t:s] 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) ?

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 Example 4

❖ The analog signal x(t)=cos(100πt) 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) ?

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 ❖ 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 )

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 Example 5

❖ The analog signal x(t)=10sin(4πt)+6sin(16πt) is sampled at the rate 20

Hz. Find the reconstructed signal xa(t) ?

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 Example 6

❖ Let x(t) be the sum of sinusoidal signals

x(t)=4+3cos(πt)+2cos(2πt)+cos(3πt) 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?

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 Example 7

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 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

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 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

Digital Signal Processing 25 Sampling and Reconstruction
 ❖ The attenuation of the filter in decibels is defined as

H (F )
A( F ) = −20log10 (dB)
H ( F0 )

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

DC for a lowpass filter.
❖ α10 =A(10F)-A(F) (dB/decade): the increase in attenuation when F is
changed by a factor of ten.
❖ α2 =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 α10 =20N
(dB/decade) and α2 =6N (dB/octave)

Digital Signal Processing 26 Sampling and Reconstruction

 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:

Digital Signal Processing 27 Sampling and Reconstruction

 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 filter’s phase response is assumed to be ignored in this example.

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 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  Fst

❖ It is practical to use a
staircase reconstructor

Digital Signal Processing 29 Sampling and Reconstruction

 Fig: Frequency response of staircase reconstructor

Digital Signal Processing 30 Sampling and Reconstruction

 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

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 Homework 1

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 Homework 2

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 Homework 3

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 Homework 3

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 Homework 4

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 Homework 5

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 Homework 6

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 Homework 7

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 Homework 8

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 Homework 9

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 Homework 10

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 Homework 11

❖ Cho tín hiệu ngõ vào tương tự x(t) = 3cos103πt – 4sin104πt (t: s) đi qua
hệ thống lấy mẫu và khôi phục lý tưởng với tần số lấy mẫu Fs = 8 KHz.
a) Viết biểu thức của tín hiệu sau lấy mẫu x[n]? Xác định giá trị mẫu
x[n=2] của tín hiệu sau lấy mẫu.
b) Có hay không 1 tần số lấy mẫu khác (Fsb ≠ 8 KHz) cho cùng kết quả
tín hiệu sau lấy mẫu x[n]? Nếu không, hãy chứng minh. Nếu có, hãy
chỉ ra 1 tần số lấy mẫu khác đó.
c) Vẽ phổ biên độ của tín hiệu sau lấy mẫu trong phạm vi tần số từ 0 đến
10 KHz.
d) Xác định biểu thức của tín hiệu sau khôi phục.
e) Xác định biểu thức của tín hiệu sau khôi phục trong trường hơp dùng
thêm bộ tiền lọc thông thấp thực tế có biên độ phẳng trong tầm [-4 4]
KHz và suy giảm với tốc độ -1@0dB/decade bên ngoài dải thông.

Digital Signal Processing 44 Sampling and Reconstruction

 Homework 12

❖ Cho tín hiệu ngõ vào tương tự x(t) = 2 – 4cos6πt + 8cos10πt

(t:ms) đi qua hệ thống lấy mẫu và khôi phục lý tưởng với tần số
lấy mẫu lựa chọn Fs = 7 KHz.
a) Vẽ phổ biên độ của tín hiệu ngõ vào x(t).
b) Vẽ phổ biên độ của một tín hiệu chồng lấn (aliased signal) với
x(t).
c) Vẽ phổ biên độ của tín hiệu sau lấy mẫu trong phạm vi tần số từ
0 đến 10 KHz.
d) Tìm giá trị mẫu x[n=2] của tín hiệu sau lấy mẫu.
e) Xác định biểu thức (theo thời gian) của tín hiệu sau khôi phục lý
tưởng
f) Tìm điều kiện của tần số lấy mẫu để khôi phục đúng tín hiệu ngõ
vào x(t).
Digital Signal Processing 45 Sampling and Reconstruction
 Homework 13

❖ Cho tín hiệu ngõ vào tương tự x(t) = 14sin23t + 3sin14t (t: ms) đi qua
hệ thống lấy mẫu và khôi phục lý tưởng với tần số lấy mẫu Fs = 8 KHz.
a) Tìm giá trị mẫu x[n=4] của tín hiệu sau lấy mẫu?
b) Xác định biểu thức của 1 tín hiệu chồng lấn (aliased signal) với tín
hiệu ban đầu x(t)?
c) Vẽ phổ biên độ của tín hiệu sau lấy mẫu trong phạm vi tần số từ 0 đến
8 KHz?
d) Xác định biểu thức của tín hiệu sau khôi phục?
e) Xác định biểu thức của tín hiệu sau khôi phục trong trường hơp dùng
thêm bộ tiền lọc thông thấp thực tế có biên độ phẳng trong tầm 4 KHz
và suy giảm với tốc độ -4@dB/decade bên ngoài dải thông?
f) Xác định 1 tập giá trị thích hợp (A, B, FA ≠ FB) của tín hiệu ngõ vào
x(t) = AsinFAt + BsinFBt (t: ms) để tín hiệu sau khôi phục (khi
không dùng thêm bộ tiền lọc) y(t) = 2sin2t (t: ms)?
Digital Signal Processing 46 Sampling and Reconstruction
 Homework 14

❖ Cho tín hiệu ngõ vào tương tự x(t) = 1 – 2cos6t + 3sin14t (t: ms) đi qua hệ
thống lấy mẫu và khôi phục lý tưởng với tần số lấy mẫu Fs = 8 KHz.
a) Tìm giá trị mẫu x[n=2] của tín hiệu sau lấy mẫu?
b) Xác định biểu thức (theo thời gian) của 1 tín hiệu chồng lấn (aliased signal) với
tín hiệu ban đầu x(t)?
c) Vẽ phổ biên độ của tín hiệu sau lấy mẫu trong phạm vi tần số từ 0 đến 8 KHz?
d) Xác định biểu thức (theo thời gian) của tín hiệu sau khôi phục?
e) Xác định biểu thức (theo thời gian) của tín hiệu sau khôi phục trong trường hơp
dùng thêm bộ tiền lọc thông thấp thực tế có biên độ phẳng trong tầm 4 KHz và
suy giảm với tốc độ -60dB/decade bên ngoài dải thông?
f) Tìm điều kiện của chu kỳ lấy mẫu Ts sao cho tín hiệu sau khôi phục (khi không
dùng thêm bộ tiền lọc) giống tín hiệu ban đầu x(t)?
g) Tìm tần số lấy mẫu Fs lớn nhất có thể sao cho tín hiệu sau khôi phục (khi không
dùng thêm bộ tiền lọc) là tín hiệu một chiều không đổi. Xác định giá trị một
chiều không đổi này?
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 Homework 15

Digital Signal Processing 48 Sampling and Reconstruction