EE3054

Signals and Systems

Sampling of Continuous
Time Signals

Yao Wang
Polytechnic University

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McClellan and Schafer
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4/17/2008 © 2003, JH McClellan & RW Schafer 2

LECTURE OBJECTIVES

Concept of sampling

Sampling using periodic impulse train

Frequency domain analysis

Spectrum of sampled signal

Nyquist sampling theorem

Sampling of sinusoids
 Two Processes in A/D
Conversion
Sampling Quanti-
zation
xc(t) x[n] = xc(nT) x$[ n]

Sampling Quantization
Period Interval
T Q

Sampling: take samples at time nT

T: sampling period; x[ n] = x( nT ),−∞ < n < ∞

fs = 1/T: sampling frequency

Quantization: map amplitude values into a set of discrete values ± pQ

Q: quantization interval or stepsize xˆ[ n] = Q[ x(nT )]
Analog to Digital
Conversion

1
T=0.1
Q=0.25

0.5

-0.5

-1

0 0.2 0.4 0.6 0.8 1

A2D_plot.m
How to determine T and Q?

T (or fs) depends on the signal frequency range

A fast varying signal should be sampled more frequently!

Theoretically governed by the Nyquist sampling theorem

fs > 2 fm (fm is the maximum signal frequency)

For speech: fs >= 8 KHz; For music: fs >= 44 KHz;

Q depends on the dynamic range of the signal
amplitude and perceptual sensitivity

Q and the signal range D determine bits/sample R

2R=D/Q

For speech: R = 8 bits; For music: R =16 bits;

One can trade off T (or fs) and Q (or R)

lower R -> higher fs; higher R -> lower fs

We only consider sampling in this class
SAMPLING x(t)

SAMPLING PROCESS

Convert x(t) to numbers x[n]

“n” is an integer; x[n] is a sequence of values

Think of “n” as the storage address in memory

UNIFORM SAMPLING at t = nTs

IDEAL: x[n] = x(nTs)

x(t) x[n]
C-to-D
Sampling of Sinusoid
Signals
Sampling above
Nyquist rate
ωs=3ωm>ωs0

Reconstructed
=original

Sampling under
Nyquist rate
ωs=1.5ωm<ωs0

Reconstructed
\= original

Aliasing: The reconstructed sinusoid has a lower frequency than the original!
Nyquist Sampling Theorem

Theorem:

If x(t) is bandlimited, with maximum frequency fb(or
ωb =2π fb)

and if fs =1/ Ts > 2 fb or ωs =2π / Ts >2 ωb

Then xc(t) can be reconstructed perfectly from x[n]=
x(nTs ) by using an ideal low-pass filter, with cut-off
frequency at fs/2

fs0 = 2 fb is called the Nyquist Sampling Rate

Physical interpretation:

Must have at least two samples within each cycle!
Sampling Using Periodic Impulse
Train

x[n] = x(nTs )
FOURIER
TRANSFORM
of xs(t) ???

4/17/2008 © 2003, JH McClellan & RW Schafer 10

Periodic Impulse Train

∞
p(t ) = ∑ δ (t − nT )
n = −∞
s

4/17/2008 © 2003, JH McClellan & RW Schafer 11

Impulse Train Sampling

∞ ∞
xs (t) = x(t) ∑ δ (t − nTs ) = ∑ x(t)δ (t − nTs )
n=−∞ n=−∞

∞
xs (t) = ∑ x(nTs )δ(t −nTs )
4/17/2008 n=−∞
© 2003, JH McClellan & RW Schafer 12
Illustration of Sampling
x(t)

t
∞
xs (t ) = ∑ x(nTs )δ (t − nTs )
n = −∞
x[n] = x(nTs )

n
4/17/2008 © 2003, JH McClellan & RW Schafer 13
Sampling: Freq. Domain

How is the
spectrum of xs(t)
related to that of
x(t)?
∞
= ∑ ak e jkω s t EXPECT
FREQUENCY
k = −∞ SHIFTING !!!
∞ ∞
p(t ) = ∑δ (t − nTs ) = ∑ ak e jkω s t

n = −∞ k = −∞
4/17/2008 © 2003, JH McClellan & RW Schafer 14
Fourier Series Representation
of Periodic Impulse Train

∞ ∞
2π
p (t ) = ∑δ (t − nTs ) = ∑ ak e jkω s t
ωs =
Ts
n = −∞ k = −∞
Ts / 2
1 1 Fourier Series
∫ δ (t )e
− jkω s t
ak = dt =
Ts −Ts / 2
Ts
4/17/2008 © 2003, JH McClellan & RW Schafer 15
 FT of Impulse Train
∞ ∞
1 2π
p(t ) = ∑
n = −∞
δ (t − nTs ) =
Ts ∑e
k
jkω s t
↔ P ( jω ) = ∑ T
k = −∞ s
δ (ω − kω s )

2π
ωs =
Ts

4/17/2008 © 2003, JH McClellan & RW Schafer 16

Frequency-Domain Analysis:
Using Fourier Series

x s (t ) = x(t ) p (t )
∞
1
p(t ) = ∑
n = −∞
δ (t − nTs ) =
Ts ∑
k
e jkω s t

∞ ∞
1 jkω st 1 jkω st
xs (t) = x(t) ∑ e = ∑ x(t)e
Ts
k =−∞ Ts k=−∞

1 ∞
Xs ( jω ) = ∑ X( j(ω − kω s ))
Ts
k =−∞ 2π
ωs =
Ts
 Frequency-Domain Analysis:
Using Multiplication-
Convolution duality
∞ ∞
1 2π
p(t ) = ∑
n = −∞
δ (t − nTs ) =
Ts ∑e
k
jkω s t
↔ P ( jω ) = ∑ T
k = −∞ s
δ (ω − kω s )

1
x(t)p(t) ⇔ X( jω )∗ P( jω )
2π
∞
1 1 2π
X s( jω ) =
2π
X ( jω ) * P( jω ) =
2π ∑ T
k = −∞ s
X ( jω ) * δ (ω − kω s )

∞
1
=
Ts ∑ X ( j(ω − kω ))
k = −∞
s
Frequency-Domain
Representation of Sampling
“Typical”
bandlimited signal

1 ∞
Xs ( jω ) = ∑
Ts k=−∞
X( j(ω − kω s ))

4/17/2008 © 2003, JH McClellan & RW Schafer 19

Aliasing Distortion
“Typical”
bandlimited signal

If ωs < 2ωb , the copies of X(jω) overlap,

and we have aliasing distortion.

4/17/2008 © 2003, JH McClellan & RW Schafer 20

 Frequency Domain
Interpretation of Sampling
Original signal

Sampling
impulse train The spectrum of the
sampled signal includes
the original spectrum and
its aliases (copies) shifted
Sampled signal to k fs , k=+/- 1,2,3,…
ωs>2 ωm The reconstructed signal
from samples has the
frequency components
upto fs /2.
Sampled signal
ωs<2 ωm When fs< 2fm , aliasing
(Aliasing effect) occur.
Reconstruction: Frequency-Domain

If ω s > 2ωb , the copies of

H r ( jω ) X ( jω ) do not overlap, so
X r ( jω ) = H r ( jω ) X s ( jω )

4/17/2008 © 2003, JH McClellan & RW Schafer 22

Nyquist Sampling Theorem

Theorem:

If x(t) is bandlimited, with maximum frequency fb(or
ωb =2π fb)

and if fs =1/ Ts > 2 fb or ωs =2π / Ts >2 ωb

Then xc(t) can be reconstructed perfectly from x[n]=
x(nTs ) by using an ideal low-pass filter, with cut-off
frequency at fs/2

fs0 = 2 fb is called the Nyquist Sampling Rate

Physical interpretation:

Must have at least two samples within each cycle!
Sampling of Sinusoid
Signals: Temporal domain
Sampling above
Nyquist rate
ωs=3ωm>ωs0

Reconstructed
=original

Sampling under
Nyquist rate
ωs=1.5ωm<ωs0

Reconstructed
\= original

Aliasing: The reconstructed sinusoid has a lower frequency than the original!
Sampling of Sinusoid:
Frequency Domain
Spectrum of
cos(2πf0t)
-f0 0 f0

No aliasing
fs >2f0
fs -f0 >f0
Reconstructed
-fs -f0 -fs -fs+f0 -f0 0 f0 fs-f0 fs fs+f0
signal: f0
-fs/2 fs/2

With aliasing
f0<fs <2f0 (folding) 0
fs -f0 <f0
Reconstructed signal: fs -f0 -fs -f0 -fs -f0 -fs+f0 fs-f0 f0 fs fs+f0

With aliasing
fs <f0 (aliasing) -fs
0 fs
f0-fs <f0
Reconstructed signal: fs -f0 -fs -f0 -f0 -f0+fs f0-fs f0 fs+f0
More examples with
Sinusoids
SAMPLING GUI (con2dis)

4/17/2008 © 2003, JH McClellan & RW Schafer 27

Strobe Movie

From SP First, Chapter 4, Demo on

“Strobe Movie”
How to determine the necessary
sampling frequency from a signal
waveform?

Given the waveform, find the shortest ripple, there
should be at least two samples in the shortest ripple

The inverse of its length is approximately the highest
frequency of the signal

Fmax=1/Tmin

Tmin

Need at least two

samples in this
interval, in order not
to miss the rise and
fall pattern.
Sampling with Pre-Filtering
Pre-Filter Periodic
H (f) Sampling
x(t) x’(t) xd(n)

Sampling
period T

• If fs < 2fb, aliasing will occur in sampled signal

• To prevent aliasing, pre-filter the continuous signal so that fb<fs/2
• Ideal filter is a low-pass filter with cutoff frequency at fs/2
(corresponding to sync functions in time)
•Common practical pre-filter: averaging within one sampling interval
Summary

Sampling as multiplication with the periodic impulse train

FT of sampled signal: original spectrum plus shifted
versions (aliases) at multiples of sampling freq.

Sampling theorem and Nyquist sampling rate

Sampling of sinusoid signals

Can illustrate what is happening in both temporal and freq.
domain. Can determine the reconstructed signal from the
sampled signal.

Need for prefilter

Next lecture: how to recover continuous signal from
samples, ideal and practical approaches
Readings

Textbook: Sec. 12.3.1-12.3.2, 4.1-4.3

Oppenheim and Willsky, Signals and
Systems, Chap. 7.

Optional reading (More depth in frequency
domain interpretation)