Sampling and Signal

Processing
 Sampling Methods
• Sampling is most commonly done with two
devices, the sample-and-hold (S/H) and the
analog-to-digital-converter (ADC)
• The S/H acquires a continuous-time signal at
a point in time and holds it for later use
• The ADC converts continuous-time signal
values at discrete points in time into
numerical codes which can be stored in a
digital system
 Sampling Methods

Sample-and-Hold
During the clock c(t)
aperture time, the response
of the S/H is the same
as its excitation. At the
end of that time, the
response holds that value
until the next aperture time.
 Sampling Methods
An ADC converts its input signal into a code. The code can be
output serially or in parallel.
 Sampling Methods
Excitation-Response Relationship for an ADC
Sampling Methods
 Sampling Methods
Encoded signal samples can be converted back into a CT
signal by a digital-to-analog converter (DAC).
 Sampling
Signal to be Sampled

The fundamental
consideration in High Sampling Rate
sampling theory is
how fast to sample a
signal to be able to
reconstruct the signal Medium Sampling Rate
from the samples.

Low Sampling Rate

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 Sampling

The low sampling rate on the previous slide

might be adequate on a signal that varies more
slowly.

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 Claude Elwood Shannon

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 Pulse Amplitude Modulation
Consider an approximation to the ideal sampler, a pulse
train p ( t ) multiplying a signal x ( t ) to produce a response y ( t ) .
p ( t ) = rect ( t / w ) ∗ δ Ts ( t )
The average value of y ( t )
during each pulse is approximately
the value of x ( t ) at the time of
the center of the pulse. This is
known as pulse amplitude
modulation.

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 Pulse Amplitude Modulation

The response of the pulse modulator is

y ( t ) = x ( t ) p ( t ) = x ( t ) ⎡⎣ rect ( t / w ) ∗ δ Ts ( t ) ⎤⎦
and its CTFT is
∞
Y ( f ) = wfs ∑ sinc ( wkf ) X ( f − kf )
s s
k=−∞

where fs = 1 / Ts

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 Pulse Amplitude Modulation
The CTFT of the
response is basically
multiple replicas
of the CTFT of the
excitation with
different amplitudes,
spaced apart by
the pulse repetition
rate.

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 Pulse Amplitude Modulation

If the pulse train is modified to make the pulses have a constant

area instead of a constant height, the pulse train becomes
p ( t ) = (1 / w ) rect ( t / w ) ∗ δ Ts ( t )
and the CTFT of the modulated pulse train becomes
∞
Y ( f ) = fs ∑ sinc ( wkf ) X ( f − kf )
s s
k=−∞

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 Pulse Amplitude Modulation

As the aperture time w of

the pulses approaches zero
the pulse train approaches
a periodic impulse and the
replicas of the original signal s
spectrum all approach the
same size. This limit is
called impulse sampling.

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 Sampling vs. Impulse Sampling
If we simply acquire the values of x ( t ) at the sampling times
nTs we form a discrete-time signal x [ n ] = x ( nTs ) . This is
known as sampling, in contrast to impulse sampling in which
we form the continuous-time signal xδ ( t ) = x ( t )δ Ts ( t ) . These
are two different ways of conceiving the sampling process but
they really contain the same information about the signal x ( t ) .
The two signals, x [ n ] and xδ ( t ) , both consist only of impulses,
discrete-time in one case and continuous-time in the other case,
and the impulse strengths are the same for both at times that
correspond through t = nTs .

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 Aliasing
The CTFT of the impulse-sampled signal is
Xδ ( f ) = X ( f ) ∗ (1 / Ts )δ1/Ts ( f ) = fs X ( f ) ∗ δ fs ( f )
If the sampling rate is less
than twice the highest
frequency of the original
continuous-time signal,
the replicas, called aliases,
overlap.

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 Aliasing
If the CTFT of the original continuous-time signal is bandlimited
and the sampling rate is more than twice the highest frequency in
the signal, the aliases are separated and the original signal could
be recovered by a lowpass filter that rejects the aliases.

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 The Sampling Theorem
If a continuous-time signal is sampled for all time at a rate fs
that is more than twice the bandlimit fm of the signal, the original
continuous-time signal can be recovered exactly from the samples.

The frequency 2 fm is called the Nyquist rate. A signal sampled

at a rate less than the Nyquist rate is undersampled and a signal
sampled at a rate greater than the Nyquist rate is oversampled.

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 Harry Nyquist

2/7/1889 - 4/4/1976
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 Timelimited and Bandlimited
Signals

• The sampling theorem says that it is possible to

sample a bandlimited signal at a rate sufficient to
exactly reconstruct the signal from the samples.
• But it also says that the signal must be sampled for all
time. This requirement holds even for signals that are
timelimited (non-zero only for a finite time).

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Timelimited and Bandlimited Signals
A signal that is timelimited
cannot be bandlimited. Let x ( t )
be a timelimited signal. Then
⎛ t − t0 ⎞
⎛ t − t0 ⎞ rect
x ( t ) = x ( t ) rect ⎜ ⎝ Δt ⎠
⎝ Δt ⎟⎠
The CTFT of x ( t ) is
X ( f ) = X ( f ) ∗ Δt sinc ( Δtf ) e− j 2 π ft0
Since this sinc function of f is not
limited in f , anything convolved with
it will also not be limited in f and
cannot be the CTFT of a bandlimited signal.

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 Interpolation
The original continuous-time signal can be recovered
(theoretically) from samples by a lowpass filter that passes
the CTFT of the original continuous-time signal and rejects
the aliases.
X( f ) = Ts rect ( f / 2 fc ) × Xδ ( f )
! "$$#$$ % "#%
CTFT of Original Ideal Lowpass Filter CTFT of Impulse
Continuous-Time Sampled Signal
Signal

= Ts rect ( f / 2 fc ) × fs X ( f ) ∗ δ fs ( f )
Inverse transforming we get
s f s 2 fc sinc ( 2 fc t ) ∗ x ( t ) (1 / f s ) δ Ts ( t )
x ( t ) = T!
=1
"$$#$$%
∞
= (1/ fs ) ∑ x( nTs )δ ( t −nTs )
n=−∞

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 Interpolation
∞
x ( t ) = 2 ( fc / fs ) sinc ( 2 fct ) ∗ ∑ x ( nT )δ (t − nT )
s s
n=−∞
∞

(
x ( t ) = 2 ( fc / fs ) ∑ x ( nTs ) sinc 2 fc ( t − nTs ) )
n=−∞

If fc = fs / 2
∞
x (t ) = ∑ x ( nT ) sinc ((t − nT ) / T )
s s s
n=−∞

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 Practical Interpolation
Sinc-function interpolation is theoretically perfect but it
can never be done in practice because it requires samples
from the signal for all time. Therefore real interpolation
must make some compromises. Probably the simplest
realizable interpolation technique is what a DAC does.

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 Practical Interpolation
The operation of a DAC can be mathematically modeled by a
zero - order hold (ZOH), a device whose impulse response is a
rectangular pulse whose width is the same as the time between samples.
⎧1 , 0 < t < Ts ⎫ ⎛ t − Ts / 2 ⎞
h (t ) = ⎨ ⎬ = rect ⎜ ⎟⎠
⎩ 0 , otherwise ⎭ ⎝ Ts

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 Practical Interpolation
A natural idea would be to simply draw straight lines between
sample values. This cannot be done in real time because doing so
requires knowledge of the next sample value before it occurs and
that would require a non-causal system. If the reconstruction is
delayed by one sample time, then it can be done with a causal
system.
Non-Causal First- Causal First-
Order Hold Order Hold

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 Sampling Bandpass Signals
CTFT of a bandpass signal

CTFT of that bandpass signal impulse sampled at 20 kHz

Xδ ( f )

... ...
f ( kHz )
-40 -20 20 40

The original signal could be recovered by a bandpass filter

even though the sampling rate is less than twice the highest
frequency.
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 Sampling Bandpass Signals
CTFT of a bandpass signal

CTFT of that bandpass signal impulse sampled at 10 kHz

The original signal could still be recovered (barely) by an ideal

bandpass filter even though the sampling rate is half of the
highest frequency.
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 Sampling Bandpass Signals

To be able to recover the original continuous-time signal from the

samples ( k − 1) fs + ( − f L ) < f L ⇒ ( k − 1) fs < 2 f L and
kfs + ( − f H ) > f H ⇒ kfs > 2 f H . Combining and simplifying we arrive
at at the general requirement for recovering the signal as
2 fH
fs,min >
⎢⎣ f H / B ⎥⎦
where B is the bandwidth ( f H − f L ) and ⎢⎣⋅⎥⎦ means "greatest integer
less than".
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 Sampling a Sinusoid
Cosine sampled at twice its
Nyquist rate. Samples uniquely
determine the signal.

Cosine sampled at exactly its

Nyquist rate. Samples do not
uniquely determine the signal.

A different sinusoid of the same

frequency with exactly the
same samples as above.

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 Sampling a Sinusoid
Sine sampled at its Nyquist rate.
All the samples are zero.

Adding a sine at the

Nyquist frequency
(half the sampling
rate) to any signal
does not change the
samples.

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 Sampling a Sinusoid
Sine sampled slightly above
its Nyquist rate

Two different sinusoids

sampled at the same rate
with the same samples

It can be shown that the samples from two sinusoids

x1 ( t ) = A cos ( 2π f0t + θ ) (
x 2 ( t ) = A cos 2π ( f0 + kfs ) t + θ )
taken at the rate fs are the same for any integer value of k.
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 Bandlimited Periodic Signals
• If a signal is bandlimited it can be properly
sampled according to the sampling theorem.
• If that signal is also periodic its CTFT
consists only of impulses.
• Since it is bandlimited, there is a finite
number of (non-zero) impulses.
• Therefore the signal can be exactly
represented by a finite set of numbers, the
impulse strengths.

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 Bandlimited Periodic Signals
• If a bandlimited periodic signal is sampled above
the Nyquist rate and at a rate which is an integer
multiple of its fundamental frequency over exactly
one fundamental period, that set of numbers is
sufficient to completely describe it
• If the sampling continued, these same samples
would be repeated in every fundamental period
• So the number of numbers needed to completely
describe the signal is finite in both the time and
frequency domains

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 Bandlimited Periodic Signals

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 CTFT-DFT Relationship
The relation between
the CTFT of a continuous-
time signal and the DFT
of samples taken from it
will be illustrated in the
next few slides. Let an
original continuous-
time signal x ( t ) be
sampled N times at
a rate fs .

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 CTFT-DTFT Relationship
Let x ( t ) be a continuous-time signal and let
∞
xδ ( t ) = x ( t )δ Ts ( t ) = ∑ x ( nT )δ (t − nT ).
s s Also let x s [ n ] = x ( nTs ) .
n=−∞
∞
Then Xδ ( f ) = X ( f ) ∗ fsδ fs ( f ) = ∑ x ( nTs ) e− j 2 π fnTs
n=−∞
∞ ∞
and Xδ ( fs F ) = fs ∑ X ( fs ( F − k )) = ∑ s x [ n ] e − j 2 π nF

k=−∞
!##"##$
n=−∞
= Xs ( F )

Summarizing, if xδ ( t ) = x ( t )δ Ts ( t ) and x s [ n ] = x ( nTs ) then

∞
X s ( F ) = Xδ ( fs F ) , Xδ ( f ) = X s ( f / fs ) and X s ( F ) = fs ∑ X ( f ( F − k ))
s
k=−∞
∞
Xs e( ) = X ( f Ω / 2π ), X ( f ) = X ( f / f ) and X ( e ) = f ∑ X ( f (Ω / 2π − k ))
jΩ
δ s δ s s s
jΩ
s s
k=−∞

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 CTFT-DTFT Relationship
Sampling in time corresponds to periodic repetition in frequency.

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 CTFT-DFT Relationship

The sampled signal is

x s [ n ] = x ( nTs )
and its DTFT is
∞
X s ( F ) = fs ∑ X ( f ( F − n ))
s
n=−∞

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 CTFT-DFT Relationship
Only N samples are
taken. If the first sample
is taken at time t = 0 (the
usual assumption) that is
equivalent to multiplying
the sampled signal by the
window function
⎧1 , 0 ≤ n < N
w[n] = ⎨
⎩0 , otherwise

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 CTFT-DFT Relationship
The DTFT of x sw [ n ] is the periodic convolution of X s ( F ) with W ( F ) .
X sw ( F ) = W ( F ) ! X s ( F ) , W ( F ) = e− jπ F ( N −1) N drcl ( F, N )
X sw ( F ) = fs ⎡⎣ e− jπ F ( N −1) N drcl ( F, N ) ⎤⎦ ∗ X ( fs F )

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 Sampling in Frequency
Let x [ n ] be an aperiodic function with DTFT X ( F ) and let
x p [ n ] be a periodic extension of x [ n ] with period N p such
∞
that x p [ n ] = ∑ x ⎡⎣ n − mN p
⎤⎦ = x [ n ] ∗ δ N p [ n ]. Then
m = −∞
∞

( )
X p ( F ) = X ( F ) 1 / N p δ1/ N p ( F ) = 1 / N p ( ) ∑ X ( k / N )δ ( F − k / N )
k = −∞
p p

( )
and X p [ k ] = X k / N p . Now let x swp [ n ] = ∑ x [ n − mN ] with
m = −∞
sw

period N. Then X swp [ k ] = X sw ( k / N ) , k an integer and

X swp [ k ] = fs ⎡⎣ e− jπ F ( N −1) N drcl ( F, N ) ∗ X ( fs F ) ⎤⎦ .
F→ k / N

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 Sampling in Frequency
Sampling in frequency corresponds to periodic repetition in time.

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 CTFT-DFT Relationship

The last step in the process

is to periodically repeat
the time-domain signal.
The corresponding effect
in the frequency domain is
sampling. Then there are
two periodic impulse
signals which are related
to each other through the
DFT.

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 CTFT-DFT Relationship
The original signal and the final signal are related by

X swp [ k ] = fs ⎡⎣ e− jπ F ( N −1) N drcl ( F, N ) ∗ X ( fs F ) ⎤⎦

F→ k / N

W(F)

In words, the CTFT of the original signal is transformed by

replacing f with fs F. That result is convolved with the
DTFT of the window function. Then that result is transformed
by replacing F by k / N. Then that result is multiplied by fs .

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 CTFT-DFT Relationship
In moving from the CTFT of a continuous-time signal to the DFT of samples
of the continuous-time signal taken over a finite time, we do the following.
In the time domain
1. Sample the continuous time signal,
2. Window the samples by multiplying them by a window function,
and 3. Periodically repeat the non-zero samples from step 2.
In the frequency domain
1. Find the DTFT of the sampled signal which is a scaled-and-periodically-
repeated version of the CTFT of the original signal.
2. Periodically convolve the DTFT of the sampled signal with the DTFT of
the window function,
and 3. Sample in frequency the result of step 2.

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 Approximating the CTFT with
the DFT
If x ( t ) is a causal energy signal then its CTFT can be
approximated at discrete frequencies kfs / N, k an integer, by
N −1

( )
X ( kfs / N ) ≅ Ts ∑ x ( nTs ) e− j 2 π kn/N ≅ Ts × DFT x ( nTs ) , k << N
n=0

where N is an integer and NTs covers all or most of the energy

of x ( t ) .

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 Approximating the Inverse
CTFT with the DFT
If X ( kfs / N ) is known in the range − N << −kmax ≤ k ≤ kmax << N
and if the magnitude of X ( kfs / N ) is negligible outside that range
then the inverse CTFT of X can be approximated by
x ( nTs ) ≅ (1 / Ts ) × DFT −1
( X ( kf
ext s / N) )
where
⎧⎪ X ( kfs / N ) , − kmax ≤ k ≤ kmax
X ext ( kfs / N ) = ⎨
⎪⎩0 , kmax < k ≤ N / 2
and
X ext ( kfs / N ) = X ext (( k + mN ) fs / N )

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 Approximating the DTFT with
the DFT
If x [ n ] is a causal energy signal its DTFT at discrete cyclic frequency
values k / N can be computed by
X ( F )F→k /N = X ( k / N ) ≅ DFT ( x [ n ])
or at discrete radian frequencies by
( )
X e jΩ Ω→2 π k /N
( )
= X e j 2 π k /N ≅ DFT ( x [ n ]) .
If x [ n ] is also time limited to a discrete time nmax < N, the computed
DTFT is exact at those frequency values.

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 Approximating Continuous-Time
Aperiodic Convolution with the DFT
If x ( t ) and h ( t ) are both aperiodic energy signals and
y ( t ) = x ( t ) ∗ h ( t ) their aperiodic convolution at times nTs
can be approximated by

y ( nTs ) ≅ Ts × DFT -1
( DFT ( x ( nT )) × DFT ( h ( nT )))
s s

for n << N.

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 Approximating Continuous-Time
Periodic Convolution with the DFT
If x ( t ) and h ( t ) are both periodic signals with common period
T sampled N times at a rate which is an integer multiple of their
fundamental periods and above the Nyquist rate and y ( t ) = x ( t ) ! h ( t )
their periodic convolution at times nTs can be approximated by

y ( nTs ) ≅ Ts × DFT -1
( DFT ( x ( nT )) × DFT ( h ( nT ))).
s s

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 Approximating Discrete-Time
Aperiodic Convolution with the DFT
If x [ n ] and h [ n ] are both energy signals and most or all of their
energy occurs in the time range 0 ≤ n < N and y [ n ] = x [ n ] ∗ h [ n ]
then

y [ n ] ≅ DFT −1
( DFT ( x [ n ]) × DFT ( h [ n ]))
for n << N.

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Discrete-Time Periodic Convolution
with the DFT
If x [ n ] and h [ n ] are both periodic signals with common period
N and y [ n ] = x [ n ] ! h [ n ] their periodic convolution at times n
can be computed by

y [ n ] = DFT −1
( DFT ( x [ n ]) × DFT ( h [ n ]))
using N points in the DFT, and the computation is exact.

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 Discrete-Time Sampling

A discrete-time signal x [ n ] is sampled

by multiplying it by a discrete-time
periodic impulse to form x s [ n ]. The time
between samples is the period of the
periodic impulse N s .
x s [ n ] = x [ n ]δ N s [ n ]

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 Discrete-Time Sampling

Aliases appear in the DTFT of the

sampled signal and, if they do
not overlap, the original signal can
be recovered from the samples. The
minimum sampling rate for recovering
the signal is 2Fm , twice the highest
discrete-time cyclic frequency in the
signal.

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 Discrete-Time Sampling
The original signal can be recovered from the samples by
interpolation using a lowpass digital filter.
X ( F ) = X s ( F ) × (1 / Fs ) rect ( F / 2Fc ) ∗ δ1 ( F )
!"# !$$$$"$$$$#
DTFT of Lowpass Digital Filter
Sampled
Signal

A discrete-time sinc function is the ideal interpolating

function.
x [ n ] = x s [ n ] ∗ ( 2Fc / Fs ) sinc ( 2Fc n )

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 Discrete-Time Sampling
When a discrete-time signal is sampled, all the values of the signal
not at the sample times are set to zero. For efficient transmission
of the sampled signal these zero values are omitted and only the
sample values are transmitted. This is decimation or downsampling.
The decimated signal is x d [ n ] = x s [ N s n ] = x [ N s n ]. The DTFT of the
∞ ∞
decimated signal is X d ( F ) = ∑ x [n]e
d
− j 2 π Fn
= ∑ x [ N n]e
s s
− j 2 π Fn
.
n=−∞ n=−∞

Let m = N s n. Then
∞
Xd ( F ) = ∑ x s [ m ] e− j 2 π Fm/N s = X s ( F / N s )
m=−∞
m=integer
multiple of N s

Decimation in time corresponds to expansion in frequency by a factor of N s .

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 Discrete-Time Sampling

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 Discrete-Time Sampling
The opposite of decimation is interpolation or upsampling
which is used to restore the original signal from the sampled-
and-decimated signal. Let the decimated signal be x [ n ]. Then
the upsampled signal is
⎧x [ n / N s ] , n / N s an integer
xs [n] = ⎨
⎩0 , otherwise
The zeros that were removed in decimation are restored. The
corresponding effect in the frequency domain of this expansion in
the time domain is compression by a factor of N s , X s ( F ) = X ( N s F ) .

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 Discrete-Time Sampling
The next step is to lowpass filter the time-expanded signal x s [ n ]
to form x i [ n ].
Xi ( F ) = X s ( F ) × rect ( N s F ) ∗ δ1 ( F )
!"# !$$ $"$$$ #
DTFT of Lowpass Filter
Time-
Expanded
Signal

In the time domain

x i [ n ] = x s [ n ] ∗ (1 / N s ) sinc ( n / N s ) .

Except for a gain factor, this is the same as the original signal that was
first sampled.

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 Discrete-Time Sampling

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