Chapter 3

Sampling

Introduction

Digital signal processing system

Most signals in nature are in analog form
Necessitating an analog-to digital conversion process
Sampling
Acquisition of a continuous signal at discrete time intervals
Conversion of continuous signal to discrete signal

A-D convertor

Digital system

Fig. 3-1.

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Sampling theory

Sampling with interval T

Sampling with a periodic impulse train by conversion to a discretetime sequence

Analog signal

Discrete signal
Sampler

Fig. 3-2.
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 Interval, T > 0

Fig. 3-3.

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Mathematical representation of Sampling theorem

Multiplying band-limited and continuous signal to impulse train

Fig. 3-4.
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 Sampling using impulse train in time domain

xs (t ) = x(t ) s (t )
=
s (t )

(3-1)

(t nT )

(3-2)

n =

Similar with sampling signal value at impulse signal

x(t ) (t =
t0 ) x(t0 ) (t t0 )
Applying to Eq.(3-1)
=
xs (t )

x(nT ) (t nT )

(3-3)

n =

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 Using multiplying and convolution properties of Fourier transform

X s ( j )
=

1
[ X ( j ) S ( j )]
2

(3-4)

By example 2-8
2
=
S ( j )
T

( k )

k =

(3-5)

Convolution of a signal and impulse

Shifting the signal to the position of impulse
X ( j ) ( 0=
) X ( j ( 0 ))
1
=
X s ( j )
X ( j ( ks ))

T k =

(3-6)

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 Sampling in frequency domain

Repeating the spectrum of sampled signal at period of s
Scaled to 1/ T
If s 2max , all information of signal is preserved
If s < 2max , the signal is indistinguishable from overlap region

Fig. 3-5.
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 Sampling of sinusoidal signal with frequency, f

Varying sampling interval

Fig. 3-6.
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 Sampling frequency to reconstruct original signal

s > 2max

(3-7)

Nyquist frequency
=
N

fN =

1
[ Hz ]
2T

=
[ rad / sec]
2 T

(3-8)

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Reconstruction of continuous
signal from samples

Reconstruction of band-limited signal

X r ( j ) = X s ( j ) H ( j )
= X ( j )

Using low-pass filter in time domain

x=
xs (t ) h(t )
r (t )
By Eq.(3-3)
=
xr (t )

x(nT )h(t nT )

(3-9)

n =

Interpolation with ideal low-pass filter

Impulse response
h(t ) =

cT sin(c t )
c t

(3-10)

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 Reconstructed continuous signal

xr (t ) =

n =

x(nT )

cT sin(c (t nT ))

c (t nT )

(3-11)

Fig. 3-7.
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c =
/T
Reconstructed signal with=
s /2

xr (t ) =

sin (t nT )
T

x(nT )

n =

T
Using convolution of sinc function
h(t ) =

(3-12)

(t nT )

sin( t / T )
t / T

(3-13)

Fig. 3-8.
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Zero-order hold interpolation

Nearest neighbor interpolation
Transfer function with properties of sinc function
2sin(T / 2)
H 0 ( j ) = e jT /2

Fig. 3-9.

(3-14)

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 Properties in frequency domain

Inducing distortion of reconstructed signal
Holding effect of rectangle function in time domain
Passing spectrums over Nyquist frequency

Ideal low-pass filter

Zero-order hold interpolation function

Fig. 3-10.

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 Applying low-pass filtering for additional smoothing

Smoothing
filter

Fig. 3-11.

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First-order hold interpolation

Linear interpolation
Transfer function
1 sin(T / 2)
H1 ( j ) =
T / 2

(3-15)

Fig. 3-12.
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Comparison of interpolation functions

Ideal interpolation function

Zero-order hold interpolation function

First-order hold interpolation function

Fig. 3-13.

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Analog to digital conversion

Digital signal process

Anti-aliasing filter
To reduce aliasing and bandlimit analog input signal

A/D converter
Converting the analog input signal into digital form

D/A converter
Converting processed signal back into analog form

Smoothing filter
Smoothing the reconstructed signal and removing unwanted high
frequency components
Anti-aliasing filter

A/D

Digital

D/A

Smoothing

(LPF)

converter

processors

converter

filter

Fig. 3-14.

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Analog to digital conversion process

Sampling
Converting the analog signal into a discrete-time continuous signal

Quantizer
Each signal sample is quantized into one of 2 B levels

Encoder
Encoding the discrete levels into distinct binary word each of length B bits

Lowpass

Sample and hold

Quantizer

Encoder

filter

Fig. 3-15.

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Quantization

Quantization error
e=
(n) x(n) x (n)

(3-16)

x(t) and Quantization

Quantization error

Fig. 3-16.
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 Max. quantization error

If q is least-significant bit(LSB)

q/2

Quantization error using probability density function

Average and variance
me = 0

e2 = q 2 /12

Fig. 3-17.

(3-17)

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 Quantization interval
2Xm Xm
=
2B
2 B 1

=
q

(3-18)

Variance of quantization error

22 B + 2 X m2
=
12
2
e

(3-19)

Signal-to-noise ratio
x2

x2
=
=
SNR 10
log10 2 10 log10 2 B + 2 2

X
2
/12

m
e

(3-20)

3 22 B x2
Xm
= 10 log10
= 6.02 B + 4.77 20 log10

2
X
m

x
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Example 3-1
1
,
p (e) = q
0,

q
q
e
2
2

otherwise

Mean-square quantization error(MSQE)

e2 = e 2 p(e) de

q
2
q

1 2
e de
q

q2
=
12

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