Bài giảng Digital signal processing - Chapter 1: Sampling and Reconstruction - Nguyễn Thanh Tuấn

Chapter 1  
Sampling and Reconstruction  
Nguyen Thanh Tuan,M.Eng.
Department of Telecommunications (113B3)  
Ho Chi Minh City University of Technology  
Email: [email protected]om  
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  
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Sampling and Reconstruction  
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.  
Digital Signal Processing  
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Sampling and Reconstruction  
2. Analog to digital conversion  
Analog to digital (A/D) conversion is a three-step process.  
xQ(n)  
x(nT)≡x(n)  
11010  
x(t)  
Sampler  
t=nT  
Quantizer  
Coder  
A/D converter  
xQ(n)  
x(t)  
x(n)  
111  
110  
101  
100  
011  
010  
001  
000  
n
t
n
Digital Signal Processing  
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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  
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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  
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Sampling and Reconstruction  
Example 2  
Consider the two analog sinusoidal signals  
7
x1(t) 2cos(2t),  
8
1
x2 (t) 2cos(2t); t(s)  
8
These signals are sampled at the sampling frequency Fs=1 Hz.  
Find the discrete-time signals ?  
Solution:  
1
7 1  
7
x1(n) x1(nT) x1(n ) 2cos(2n) 2cos( n)  
F
8 1  
4
s
1
4
2cos((2)n) 2cos( n)  
4
1
11  
x2(n) x2(nT) x2(n ) 2cos(2n) 2cos( n)  
81  
1
F
4
s
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”  
Digital Signal Processing  
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Sampling and Reconstruction  
F2=1/8 Hz  
F1=7/8 Hz  
Fs=1 Hz  
Fig: Illustration of aliasing  
Digital Signal Processing  
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Sampling and Reconstruction  
4. Aliasing of Sinusoids  
In general, the sampling of a continuous-time sinusoidal signal  
at a sampling rate F =1/T results in a  
x(t) Acos(2F t )  
s
0
discrete-time signal x(n).  
xk (t) Acos(2F t )  
The sinusoids  
is sampled at Fs , resulting  
k
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  
Digital Signal Processing  
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Sampling and Reconstruction  
5. Spectrum Replication  
x(nT) x(t) x(t) (t nT) x(t)s(t)  
s(t) (t nT)  
Let  
where  
n  
n  
s(t) is periodic, thus, its Fourier series are given by  
1
1
1
S (t)ej2F ntdt (t)dt   
s(t) S ej2F nt  
s
s
where  
n
n
T   
T   
T
n  
T
T
1
s
s(t)   
ej2F nt  
Thus,  
T
n  
1
x(t)ej2nf t  
s
x(t) x(t)s(t)   
which results in  
T
n  
X (F)   
X (F nF )  
x(t)  
Taking the Fourier transform of  
yields  
s
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.  
Digital Signal Processing  
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Sampling and Reconstruction  
Fs/2 ≥ Fmax  
Fig: Spectrum replication caused by sampling  
Fs/2 < Fmax  
Fig: Aliasing caused by overlapping spectral replicas  
Digital Signal Processing  
11  
Sampling and Reconstruction  
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  
F 2Fmax  
frequency Fmax.  
s
Fs=2Fmax is called Nyquist rate; Fs/2 is called Nyquist frequency;  
[-Fs/2, Fs/2] is Nyquist interval.  
Digital Signal Processing  
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Sampling and Reconstruction  
The values of Fmax and Fs depend on the application  
Application  
Biomedical  
Speech  
Fmax  
Fs  
1 KHz  
4 KHz  
20 KHz  
4 MHz  
2 KHz  
8 KHz  
40 KHz  
8 MHz  
Audio  
Video  
Digital Signal Processing  
13  
Sampling and Reconstruction  
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 [F / 2,F / 2]  
s
s
H(F)   
An ideal reconstructor (lowpass filter)  
0
otherwise  
Then  
Xa (F) X(F)H(F) X(F)  
Digital Signal Processing  
14  
Sampling and Reconstruction  
Example 3  
The analog signal x(t)=cos(20πt) 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) ?  
Digital Signal Processing  
15  
Sampling and Reconstruction  
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) ?  
Digital Signal Processing  
16  
Sampling and Reconstruction  
Remarks: xa(t) contains only the frequency components that lie in the  
Nyquist interval (NI) [-Fs/2, Fs/2].  
sampling at Fs  
x(t), F0 NI ------------------> x(n) ----------------------> xa(t), Fa=F0  
ideal reconstructor  
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  
F F mod(F )  
a
s
Digital Signal Processing  
17  
Sampling and Reconstruction  
Example 5  
The analog signal x(t)=10sin(4πt)+6sin(16πt) is sampled at the rate 20  
Hz. Find the reconstructed signal xa(t) ?  
Digital Signal Processing  
18  
Sampling and Reconstruction  
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?  
Digital Signal Processing  
19  
Sampling and Reconstruction  
Example 7  
Digital Signal Processing  
20  
Sampling and Reconstruction  
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