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Evans

Dept. of Electrical and Computer Engineering

The University of Texas at Austin

Lecture 4 http://courses.utexas.edu/

EE 445S Real-Time Digital Signal Processing Lab Fall 2012

Sampling and Aliasing

4 - 2

Outline

Sampling

Time and frequency domains

Sampling theorem

Aliasing and folding

Bandpass sampling

Conclusion

7 - 3

Data Conversion

Analog-to-Digital Conversion

Lowpass filter has

stopband frequency

less than f

s

to reduce

aliasing due to sampling

(enforce sampling theorem)

Digital-to-Analog Conversion

Discrete-to-continuous

conversion could be as

simple as sample and hold

Lowpass filter has stopband

frequency less than f

s

reduce artificial high frequencies

Analog

Lowpass

Filter

Discrete to

Continuous

Conversion

f

s

Lecture 7

Analog

Lowpass

Filter

Quantizer

Sampler at

sampling

rate of f

s

Lecture 8 Lecture 4

4 - 4

| | ( )

s

T k f k f =

Sampling: Time Domain

Many signals originate in continuous-time

Talking on cell phone, or playing acoustic music

By sampling a continuous-time signal at

isolated, equally-spaced points in time, we

obtain a sequence of numbers

k e {, -2, -1, 0, 1, 2,}

T

s

is the sampling period.

Sampled analog waveform

( ) ( )

=

=

k

s sampled

T k t t f t f ) ( o

impulse train

f(t)

t

T

s

T

s

( ) t f

sampled

Review

4 - 5

Sampling: Frequency Domain

Sampling replicates spectrum of continuous-time

signal at integer multiples of sampling frequency

Fourier series of impulse train where e

s

= 2 t f

s

( ) ( ) ) (2 cos 2 ) ( cos 2 1

1

) ( . . . + + + = =

=

t t

T

T k t t

s s

s

k

s T

s

e e o o

( ) ) (2 cos ) ( 2 ) ( cos ) ( 2 ) (

1

) ( ) ( ) ( . . . + + + = = t t f t t f t f

T

t t f t g

s s

s

T

s

e e o

e

G(e)

e

s

2e

s

2e

s

e

s

e

F(e)

2tf

max

-2tf

max

max max max

2 2 2 2 if only and if gap f f f f f

s s

> < t t t

Modulation

by cos(2 e

s

t)

Modulation

by cos(e

s

t)

Review

How to

recover

F(e)?

4 - 6

Sampling Theorem

Continuous-time signal x(t) with frequencies no

higher than f

max

can be reconstructed from its

samples x(k T

s

) if samples taken at rate f

s

> 2 f

max

Nyquist rate = 2 f

max

Nyquist frequency = f

s

/ 2

Example: Sampling audio signals

Normal human hearing is from about 20 Hz to 20 kHz

Apply lowpass filter before sampling to pass low

frequencies up to 20 kHz and reject high frequencies

Lowpass filter needs 10% of maximum passband frequency

to roll off to zero (2 kHz rolloff in this case)

What happens

if f

s

= 2 f

max

?

Review

4 - 7

Sampling Theorem

Assumption

Continuous-time signal has

absolutely no frequency

content above f

max

Sampling time is exactly the

same between any two

samples

Sequence of numbers

obtained by sampling is

represented in exact

precision

Conversion of sequence to

continuous time is ideal

In Practice

4 - 8

Sampling and Oversampling

As sampling rate increases above Nyquist rate,

sampled waveform looks more like original

Zero crossings: frequency content of a sinusoid

Distance between two zero crossings: one half period

With sampling theorem satisfied, sampled sinusoid

crosses zero right number of times per period

In some applications, frequency content matters not time-

domain waveform shape

DSP First, Ch. 4, Sampling and Interpolation

demo http://www.ece.gatech.edu/research/DSP/DSPFirstCD/

4 - 9

Aliasing

Continuous-time

sinusoid

x(t) = A cos(2t f

0

t

+ |)

Sample at T

s

= 1/f

s

x[n] = x(T

s

n) =

A cos(2t f

0

T

s

n + |)

Keeping the sampling

period same, sample

y(t) = A cos(2t (f

0

+ l f

s

) t + |)

where l is an integer

y[n] = y(T

s

n)

= A cos(2t(f

0

+ lf

s

)T

s

n + |)

= A cos(2tf

0

T

s

n + 2tlf

s

T

s

n + |)

= A cos(2tf

0

T

s

n + 2tln + |)

= A cos(2tf

0

T

s

n + |)

= x[n]

Here, f

s

T

s

= 1

Since l is an integer,

cos(x + 2 t l) = cos(x)

y[n] indistinguishable

from x[n]

4 - 10

Aliasing

Since l is any integer, a countable but infinite

number of sinusoids give same sampled sequence

Frequencies f

0

+ l f

s

for l = 0

Called aliases of frequency f

0

with respect to f

s

All aliased frequencies appear same as f

0

due to sampling

Signal Processing First, Continuous to Discrete

Sampling demo (con2dis)

users.ece.gatech.edu/mcclella/SPFirst/Updates/SPFirstMATLAB.html

4 - 11

Aliasing

Sinusoid sin(2 t f

input

t) sampled at f

s

= 2000

samples/s with f

input

varied

Mirror image effect about f

input

= f

s

gives rise

to name of folding

A

p

p

a

r

e

n

t

f

r

e

q

u

e

n

c

y

(

H

z

)

Input frequency, f

input

(Hz)

1000

1000 2000 3000 4000

f

s

= 2000 samples/s

4 - 12

Bandpass Sampling

Reduce sampling rate

Bandwidth: f

2

f

1

Sampling rate f

s

must

be greater than analog

bandwidth f

s

> f

2

f

1

For replica to be centered

at origin after sampling

f

center

= (f

1

+ f

2

) = k f

s

Practical issues

Sampling clock tolerance: f

center

= k f

s

Effects of noise

Ideal Bandpass Spectrum

f

1

f

2

f

f

2

f

1

Sample at f

s

Sampled Ideal Bandpass Spectrum

f

1

f

2

f

f

2

f

1

Lowpass filter to

extract baseband

4 - 13

Sampling for Up/Downconversion

Upconversion method

Sampling plus bandpass

filtering to extract

intermediate frequency

(IF) band with f

IF

= k

IF

f

s

Downconversion method

Bandpass sampling plus

bandpass filtering to extract

intermediate frequency (IF)

band with f

IF

= k

IF

f

s

f

f

max

-f

max

f f

s

f

IF

f

IF

f

s

f

1

f

2

f

f

2

f

1

Sample

at f

s

f

f

2

f

1

-f

IF

f

IF

4 - 14

Conclusion

Sampling replicates spectrum of continuous-time

signal at offsets that are integer multiples of

sampling frequency

Sampling theorem gives necessary condition to

reconstruct the continuous-time signal from its

samples, but does not say how to do it

Aliasing occurs due to sampling

Noise present at all frequencies

A/D converter design tradeoffs to control impact of aliasing

Bandpass sampling significantly reduces

sampling rate by using aliasing to our benefit

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