1497 Chapter4

Chapter 4
Sampling of Continuous-Time Signals

Der-Feng Tseng
Department of Electrical Engineering
National Taiwan University of Science and Technology
(through the courtesy of Prof. Peng-Hua Wang of National Taipei University)
February 19, 2015
Der-Feng Tseng (NTUST)
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Outline
1
4.1 Periodic Sampling
4.2 Frequency Domain representation
4.3 Reconstruction
4.4 DT Processing of Signals
4.5 CT Processing of Signals
4.6 Changing Sampling Rate
4.7 Multirate Signal Processing
4.8 A/D and D/A
4.9 Oversampling A/D and D/A

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About The figures

All the figures are from Discrete-Time Signal Processing, 2e, by
Oppenheim, Schafer, and Buck,Prentice Hall, Inc.
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4.1 Periodic Sampling
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Concept
Figure 4.1 Block Diagram of an ideal C/D converter

The sequence of samples x[n] is obtained from a continuous-time
signal xc (t) according to
x[n] = xc (nT ),
< n <
T is the sampling period. fs = 1/T is the sampling frequency.

s = 2fs = 2/T is the sampling frequency in radians per second.
The system is an ideal continuous-to-discrete-time (C/D) converter.
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Figure 4.2
(a) C/D converter = modulate by s(t) + impulse to sequence
(b) sampling by two rates
(c) output sequences

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4.2 Frequency Domain representation
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Derivation
Let s(t) be the periodic impulse train
s(t) =
(t nT ),
S(j) =
n=
2 X
( ks )
T
k=
The sampled signal xs (t) from a continuous-time signal xc (t) is

xs (t) = xc (t)s(t) = xc (t)
(t nT )
n=
xc (nT )(t nT )
n=
The Fourier transformation of xs (t) is
1 X
1
Xc (j) S(j) =
Xc (j( ks ))
Xs (j) =
2
T
k=
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Figure 4.3
Suppose xc (t) is bandlimited at = N , the replicas of Xc (j) do not

overlap if s N > N or s > 2N
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Recovery
Xc (j) can be recovered by Xs (j)Hr (j) where Hr (j) is an ideal

lowpass filter with gain T and cutoff frequency c with
N < c < s N .
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Alias
If s 2N , the copies of Xc (j) overlap, and Xc (j) cant be
recovered by lowpass filtering. The reconstructed signal by
Xr (j) = Xs (j)Hr (j) is distorted. This is refered to alias
distortion.
Nyquist Sampling Theorem. Let xc (t) be a bandlimited signal with
Xc (j) = 0 for || N . Then xc (t) is uniquely determined by its
samples x[n] = xc (nT ), n = 0, 1, 2, . . . , if
s =
2
2N .
T
N is commonly referred to as the Nyquist frequency.

The minimal sampling frequency 2N is called the Nyquist rate.
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Example
Let xc (t) = cos 0 t. With no aliasing, xr (t) = cos 0 t. With aliasing,

xr (t) = cos(0 0 )t.
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Discrete-Time FT
The sampled signal
Xs (j) =
xc (nT )ejT n
n=
Since x[n] = xc (nT ) and X(ej ) =
x[n]ejn , it follows that
n=
jT
Since X(e

Xs (j) = X(ej )=T = X(ejT )
1 X
Xc (j( ks )), we have
)=
T n=

1 X
2k
X(e ) =
Xc j
,
T n=
T
T
j
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Example 4.1
xc (t) = cos(4000t) (2000 Hz).

Sampling period T = 1/6000, s = 12000 (6000 Hz)
x[n] = cos(4000T n) = cos 0 n where 0 = 2/3.
Xc (j) = ( 4000) + ( + 4000)
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Example 4.2
xc (t) = cos(16000t) (8000 Hz).

x[n] = cos(16000T n) = cos 0 n where 0 = 8/3 = 2/3.
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Example 4.3
xc (t) = cos(4000t) (2000 Hz).

x[n] = cos(4000T n) = cos 0 n where 0 = 2/3.
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4.3 Reconstruction
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Derivation
Sampled sequence
xs (t) =
x[n](t nT )
n=
Reconstruction by a lowpass filter with cutoff frequency

N < c < s N .
xr (t) =
x[n]hr (t nT )
n=
A common choice c = s /2 = /T
sin(t/T )
t/T
X
sin[(n T )/T ]
x[n]
xr (t) =
(t nT )/T
n=
hr (t) =
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Illustration
xr (t) =
x[n]hr (t nT )
n=
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Illustration
sin(t/T )
t/T
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hr (t) =
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Interpolation
hr (nT ) = 0, xr (mT ) = x(mT )

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Bandlimited Reconstruction
Ideal discrete-to-continuous-tine (D/C) converter
Xr (j) =
x[n]Hr (j)ejT n = Hr (j)X(jT )
n=
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4.4 DT Processing of Signals
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Concept
x[n] = xc (nT )

1 X
2k
j
X(e ) =
Xc j
T
T
T
k=
sin[(n T )/T ]
(t nT )/T
n=
(
T Y (jT ),
Yr (j) = Hr (j)Y (jT ) =
0,
yr (t) =
y[n]
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|| < /T,
otherwise.
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LTI System
If the discrete-time system is LTI, we have Y (ej ) = H(ej )X(ej )
The spectrum of the output signal
Yr (j) = Hr (j)H(ejT )X(ejT )

X
2k
jT 1
)
= Hr (j)H(e
Xc j
T
T
T
k=
If Xc (j) = 0 for || /T , then

(
H(ejT )Xc (j),
Yr (j) =
0,
|| < /T,
|| /T.
We have Yr (j) = Heff (j)Xc (j) where

(
H(ejT ), || < /T,
Heff (j) =
0,
|| /T.
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Illustration
(
H(ejT ),
Heff (j) =
0,
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|| < /T,
|| /T.
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Example 4.4
(
1,
H(ej ) =
0,
|| < c ,
c < || .
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Example 4.5
The ideal continuous-time differentiator yc (t) =
d
dt xc (t)
The corresponding frequency response Hc (j) = j

The effective frequency response Heff (j) = j for || < /T .
The discrete-time have frequency response
H(ej ) =
j
,
T
|| <
The corresponding impulse response

n cos n sin n
=
h[n] =
n2 T
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(
0,
n=0
cos n
, n=
6 0.
nT
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Example 4.5
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Example 4.6
xc (t) = cos(0 t) x[n] = cos(0 n) where 0 = 0 T
( 0 ) + ( + 0 ),
T
T
X(ej ) = ( 0 ) + ( + 0 )
X(ejT ) =
for || /T
The output of the digital differentiator

j
[( 0 ) + ( + 0 )]
T
j0
j0
=
( 0 )
( + 0 )]
T
T
Y (ej ) = H(ej )X(ej ) =
The output of the D/C converter

Yr (j) = Hr (j)Y (ejT ) = T Y (ejT )

j0
j0
(T 0 T )
(T + 0 T )
=T
T
T
= j0 ( 0 ) j0 ( + 0 )
yr (t) = j0 12 ej0 t j0 12 ej0 t = 0 sin(0 t)

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Impulse Invariance
If h[n] = hc (nT ) H(ej ) =
T Hc (j T ),
thus
h[n] = T hc (nT ) H(ej ) = Hc (j

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),
T
||
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Example 4.7
Ideal lowpass filter
(
1,
Hc (j) =
0,
|| < c
|| c .
The impulse response of the continuous-time system

hc (t) =
sin(c t)
t
The impulse response of the discrete-time system by impulse invariant

is
sin(c n)
sin(c nT )
=
h[n] = T hc (nT ) = T
nT
n
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Example 4.8
The ideal impulse response of the continuous-time system
hc (t) = Aes0 t u(t)
The impulse response of the discrete-time system by impulse invariant
is
h[n] = T hc (nT ) = T Aes0 nT u(nT ) = AT es0 nT u[n]
The z-transform is
H(z) =
AT
1 es0 n z 1
Alias occurs!
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4.5 CT Processing of Signals
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Concept
Suppose
xc (t) =
yc (t) =
n=
n=
We have
x[n]
sin[(t nT )/T ]
(t nT )/T
y[n]
sin[(t nT )/T ]
(t nT )/T
Xc (j) = T X(ejT ), Yc (j) = Hc (j)Xc (j), Y (ej ) =
1
Yc (j )
T
T
Hc (j) = H(ejT )
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Example 4.9
The desired system

H(ej ) = ej
for implementing y[n] = x[n ] (no formal meaning).
The continuous system
Hc (j) = H(ejT ) = ejT
and
y(t) = xc (t T )
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Example 4.10
The desired system

H(ej ) =
1 sin[(M + 1)/2] jM/2

e
M +1
sin(/2)
M is even y[n] = w[n M/2]

M is odd y[n] = w[n] followed by a continuous-time delay of
M T /2.
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4.6 Changing Sampling Rate
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Motivation
Given x[n] = xc (nT ), we want to find x [n] = xc (nT ) where T 6= T .
One approach is to reconstruct xc (t) from x[n], then sample xc (t) by
T .
Not desirable, because reconstruction, A/D and D/A
are not ideal.
We are interested in the method that involve only discrete-time
operations.
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Reduction by M
If T = M T , we have the (sampling rate) compressor

xd [n] = x[nM ] = xc (nM T )
If Xc (j) = 0 for || N , then xd [n] is an exact representation of
xc (t) if /T = /(M T ) N .
The sampling rate can be reduced by a factor of M
if the original sampling rate was at least M times the Nyquist rate, or
if the bandwidth of the sequence is first reduced by a factor of M by
discrete-time filtering.
In general, we the operation of reducing the sampling rate (including

any prefiltering) will be called the downsampling.
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Reduction by M
What is the Fourier transform of xd [n] ?
x[n] = xc (nT )

2k
1 X
Xc j
X(e ) =
T
T
j
k=
xd [n] = xc (nM T ), r = i + kM

1 X
2r
Xd (e ) =
Xc j
M T r=
MT
"
#

M1
2i 2k
1 X 1 X
Xc j
=
M i=0 T
MT
T
j
k=
1
M
M1
X
X(ej(/M2i/M) )
i=0
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Reduction by M
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Reduction by M
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Reduction by M
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Reduction by M
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Downsampling
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Increase by L
T = T /L, we want to obtain

xi [n] = xc (nT )
from x[n] = xc (nT ).
This is called the upsampling.

xi [n] = x[n/L] = xc (nT /L) for n = 0, L, 2L, . . .
(Sampling rate) expander
xe [n] =
Xe (ej ) =
X
x[n/L], n = 0, L, 2L, . . .
=
x[k][n kL]
0,
otherwise.
k=
x[k]ejLk = X(ejL )
k=
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Increase by L
T = T /L, we want to obtain

xi [n] = xc (nT )
from x[n] = xc (nT ).
This is called the upsampling.

xi [n] = x[n/L] = xc (nT /L) for n = 0, L, 2L, . . .
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Increase by L
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Increase by L
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Interpolator
Ideal lowpass filter, gain L, cutoff frequency /L
sin(n/L)
n/L
X
sin[(n kL)/L]
xi [n] =
x[k]
(n kL)/L
hi [n] =
k=
We can use other interpolator instead of the ideal lowpass filter

linear interpolation
(
1 |n|/L, |n| L
0,
otherwise

1 sin(L/2)
Hlin (ej ) =
L sin(/2)
hlin [n] =
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Linear Interpolator
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Noninteger Factor
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Example 4.11
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Example 4.11
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4.7 Multirate Signal Processing
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Introduction
Multirate signal processing refers in general to utilizing upsampling,
down sampling, compressor, and expanders to increase the efficiency
of signal-processing systems.
For example, if T = 1.01T , we can first interpolate by L = 100 using
a lowpass filter, that cutoff at c = /101. then decimate by
M = 101.
Require large amounts of computation for each output
sample.
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Downsampling Identities
These two systems are equivalent
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Downsampling Identities
In (b), Xb (ej ) = H(ejM )X(ej ) and
M 1
1 X
Y (e ) =
Xb (ej(/M 2i/M ) )
M
j
1
M
i=0
M
1
X
H(ej(2i) )X(ej(/M 2i/M ) )
i=0
M 1
1 X
X(ej(/M 2i/M ) )
= H(e )
M
j
i=0
j
= H(e )Xa (e )
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Upsampling Identities
These two systems are equivalent
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Upsampling Identities
In Fig.(a),
Y (ej ) = Xa (ejL ) = X(ejL )H(ejL )
In Fig.(b), Xb (ej ) = X(ejL ), thus
Y (ej ) = H(ejL )Xb (ej )
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Polyphase Decom of Signal
Decompose h[n] into

h[n] =
M
1
X
hk [n k],
k=0
(
h[n + k], n = integer multiple of M,
where hk [n] =
0,
otherwise.
ek [n] = h[nM + k] = hk [nM ]
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Polyphase Decom of Signal
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Polyphase Decom of Filter
H(z) =
M
1
X
Ek (z M )z k
i=0
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Polyphase Decimation
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Polyphase Decimation
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Polyphase Interpolation
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Polyphase Interpolation
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4.8 A/D and D/A
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Introduction
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Prefiltering: Ideal
Antialiasing filter
Haa (j) =
1, || < c < /T,

0, || > c .
Heff (j) Haa (j)H(ejT )

Need sharp-cutoff antialiasing filters. Drawbacks
May account for a major part of the cost.
Difficult and expensive to implement.
Highly nonlinear phase response.
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Prefiltering: Oversampling
Simple antialiasing filter Haa (j) has a gradual cutoff with significant
attenuation at M N .
Oversampling at M N .
Downsampling by M with sharp cutoff antialiasing digital filter.
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Prefiltering: Oversampling
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A/D Conversion
Sampling-and-hold
Analog-to-digital: quantization
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A/D: Sample-and-hold
The output of the ideal sample-and-hold system is

x0 (t) =
x[n]h0 (t nT ) = h0 (t)
x[n](t nT )
n=
n=
h0 (t) is the impulse response of the zero-order-hold system

(
1, 0 < t < T,
h0 (t) =
0, otherwise.
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A/D: Concept
Practical A/D Converter
Conceptual A/D Converter
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A/D: Quantizer
The quantizer is a nonlinear system whose purpose is to transform the
input sample x[n] into one of a finite set of prescribed values.
x
[n] = Q(x[n])
Uniform quantizers: quantization levels are spaced uniformly.
Linear quantizers: quantization levels are of linear progression.
Bipolar quantizers: both positive and negative samples can be
quantized.
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A/D: Quantization
Uniform, linear, bipolar

The input sample is rounded to the nearest quantization level.
8 (even) quantization levels
No quantization level at zero amplitude.
An equal number of positive and negative quantization
level.
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A/D: Quantization
Let Xm be the full-scale level of the A/D converter. Each

quantization level is coded by B + 1 bits.
For a bipolar quantizer, the step size of the quantizer is
2Xm
Xm
= B+1 = B
2
2
If the signal amplitude exceeds the full-scale value, some samples are
clipped.
Der-Feng
Tseng (NTUST)
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A/D: Quantization Error
Quantization error defined as

e[n] = x
[n] x[n]
We can analyze the quantization error statistically by assuming
e[n]
e[n]
e[n]
The
is a stationary random process.

and x[n] are uncorrelated.
is a white-noise process.
pdf of e[n] is uniformly.
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Example 4.12
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/2 < e[n] < /2

Variance of e[n]: e2 =
2
12
2
22B Xm
12
Let x2 be the rms value of the signal. The signal-to-noise ratio (SNR)
SNR = 10 log10
x2
Xm
= 6.02B + 10.8 20 log10
e2
x
Increase B by 1 bit, SNR increases by 6 dB.

Halve x , SNR decreases by 6 dB.
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If the signal amplitude has a Gaussian distribution, only 0.064% of
the samples would have an amplitude greater than 4x .
We may set Xm = 4x , then
SNR 6B 1.25dB
The signal amplitude should be carefully matched to the full-scale
amplitude of the A/D converter.
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D/A Conversion
Ideal reconstruction
xr (t) =
n=
Practical reconstruction
xDA (t) =
=
n=
x
[n]
sin[(t nT )/T ]
(t nT )/T
x[n]h0 (t nT )
x[n]h0 (t nT ) +
n=
e[n]h0 (t nT )
n=
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D/A Conversion
The spectrum of x0 (t) is
"
#
1 X
X0 (j) =
Xa (j( 2k/T )) H0 (j)
T
k=
Compensated reconstruction filter

r (j) = Hr (j)
H
H0 (j)
The frequency response of the zero0order hold filter is
2 sin(T /2) jT /2
e
The compensated reconstruction filter is

( T /2
jT /2 , || < /T
r (j) = sin(T /2) e
H
0,
|| > /T.
H0 (j) =
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D/A Conversion
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4.9 Oversampling A/D and D/A
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Oversampled A/D
Assume that xa (t) is a zero-mean, WSS random process with PSD

xa xa (j) and autocorrelation function xa xa ( ). xa xa (j) is
bandlimited to N .
Sampling rate 2/T = M 2N . M is called the oversampling
ratio.
The decimation filter is an ideal lowpass filter with c = /M .
We want to analyze the quantization error.
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Oversampled A/D
Additive noise model.

We want to determine the ratio of signal power E{x2da [n]} to
quantization-noise power E{x2de [n]}. The ratio is a function of
quantization step and the oversampling ratio M .
Original signal xa (t), PSD is xa xa (j), power is
E{x2a (t)}
1
=
2
xa xa (j)d
N
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Oversampled A/D
Analysis of signal power
After C/D, x[n] = xa (nT ), PSD and power are
(
1
x x (j/T ), || < /M
j
xx (e ) = T a a
0,
/M <
Z
1
xx (ej )d = E{x2a (t)}
E{x2 [n]} =
2
After downsampling by M , PSD and power are
1
xx (ej/M )
M Z
1
E{x2da [n]} =
x x (ej )d
2 da da
Z
1
1
xx (ej/M )d = E{x2 [n]} = E{x2a (t)}
=
2 M
xda xda (ej ) =
We have E{x2da [n]} = E{x2 [n]} = E{x2a (t)}

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Oversampled A/D
Analysis of noise power
Since variance of e[n] is e2 = 2 /12, we have ee [m] = e2 [m] and
ee (ej ) = 2 for || < .
After ideal lowpass filter, the noise power is
E{e2 [n]} =
1
2
/M
e2 d =
/M
e2
M
After downsampling, the noise power does not changed.

E{x2de [n]}
2
2
1
= e =
=
M
12M
12M
Xm
2B
2
Let Pde = E{x2de [n]}, we have

1
1
1
B = log2 M log2 12 log2 Pde + log2 Xm
2
2
2
For example, M = 4, one less bit to achieve a desired accuracy.
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Oversampled A/D
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Noise Shaping A/D
If we want to reduce the number of bits from 16 to 12, we need

M = 44 = 256.
The concept of noise shaping is to modify the A/D conversion
procedure so that the PSD of the quantization noise is not uniform,
but rather, is shaped such that most of the noise power is outside
|| < /M .
DSP Chapter 4
February 19, 2015
93 / 100
1st-Order Noise Shaping A/D
Denote the transfer function from x[n] to y[n] as Hz (z)

1
X(z) z 1 Yx (z)
1
1z
Yx (z) = X(z), Hx (z) = 1
Yx (z) =
Denote the transfer function from e[n] to y[n] as He (z)

Ye (z) = E(z) +
1
1 z 1

z 1 Ye (z)
Ye (z) = (1 z 1 )E(z), He (z) = 1 z 1

DSP Chapter 4
February 19, 2015
94 / 100
e[n] = e[n] e[n 1]

ee(ej ) = e2 |He (ej )|2 = e2 [2 sin(/2)]2
Pda = E{x2da [n]} = E{x2 [n]} = E{x2a (t)}
IfM is sufficient large such that sin(/2M ) /2M , the
quantization-noise power is
Pde =
B=
23
1 2
2 12M
[2 sin(/2M )]2 d
1 2 2
36 M 3
log2 M + log2 (/6) 12 log2 Pde + log2 Xm
DSP Chapter 4
February 19, 2015
95 / 100
DSP Chapter 4
February 19, 2015
96 / 100
Equivalent saving in quantizer bits relative to M = 1

Direct quantization
1
1
1
B = log2 M log2 12 log2 Pde + log2 Xm
2
2
2
First-order noise shaping
1
3
B = log2 M + log2 (/6) log2 Pde + log2 Xm
2
2
DSP Chapter 4
February 19, 2015
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2nd-Order Noise Shaping A/D
He (z) = (1 z 1 )2
ee(ej ) = e2 [2 sin(/2)]4
DSP Chapter 4
February 19, 2015
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Oversampling D/A
DSP Chapter 4
February 19, 2015
99 / 100
Oversampling D/A
DSP Chapter 4
February 19, 2015
100 / 100

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Chapter 4

Sampling of Continuous-Time Signals

February 19, 2015

Der-Feng Tseng (NTUST)

February 19, 2015

4.1 Periodic Sampling

4.2 Frequency Domain representation

4.4 DT Processing of Signals

4.5 CT Processing of Signals

4.6 Changing Sampling Rate

4.7 Multirate Signal Processing

4.8 A/D and D/A

4.9 Oversampling A/D and D/A

February 19, 2015

About The figures

Der-Feng Tseng (NTUST)

February 19, 2015

4.1 Periodic Sampling

Der-Feng Tseng (NTUST)

February 19, 2015

Figure 4.1 Block Diagram of an ideal C/D converter

T is the sampling period. fs = 1/T is the sampling frequency.

February 19, 2015

(a) C/D converter = modulate by s(t) + impulse to sequence

(b) sampling by two rates

(c) output sequences

February 19, 2015

4.2 Frequency Domain representation

Der-Feng Tseng (NTUST)

February 19, 2015

The sampled signal xs (t) from a continuous-time signal xc (t) is

The Fourier transformation of xs (t) is

Der-Feng Tseng (NTUST)

February 19, 2015

Suppose xc (t) is bandlimited at = N , the replicas of Xc (j) do not

February 19, 2015

Xc (j) can be recovered by Xs (j)Hr (j) where Hr (j) is an ideal

Der-Feng Tseng (NTUST)

February 19, 2015

N is commonly referred to as the Nyquist frequency.

Der-Feng Tseng (NTUST)

February 19, 2015

Let xc (t) = cos 0 t. With no aliasing, xr (t) = cos 0 t. With aliasing,

Der-Feng Tseng (NTUST)

February 19, 2015

Since x[n] = xc (nT ) and X(ej ) =

x[n]ejn , it follows that

Der-Feng Tseng (NTUST)

February 19, 2015

xc (t) = cos(4000t) (2000 Hz).

February 19, 2015

xc (t) = cos(16000t) (8000 Hz).

Der-Feng Tseng (NTUST)

February 19, 2015

xc (t) = cos(4000t) (2000 Hz).

Der-Feng Tseng (NTUST)

February 19, 2015

Der-Feng Tseng (NTUST)

February 19, 2015

Reconstruction by a lowpass filter with cutoff frequency

Der-Feng Tseng (NTUST)

February 19, 2015

Der-Feng Tseng (NTUST)