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Digital Processing of Digital Processing of

Continuous-Time Signals Continuous-Time Signals


• Digital processing of a continuous-time
• Conversion of a continuous-time signal into
signal involves the following basic steps:
digital form is carried out by an analog-to-
(1) Conversion of the continuous-time digital (A/D) converter
signal into a discrete-time signal,
• The reverse operation of converting a
(2) Processing of the discrete-time signal, digital signal into a continuous-time signal
(3) Conversion of the processed discrete- is performed by a digital-to-analog (D/A)
time signal back into a continuous-time converter
signal
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Digital Processing of Digital Processing of


Continuous-Time Signals Continuos-Time Signals
• Since the A/D conversion takes a finite • To prevent aliasing, an analog anti-aliasing
amount of time, a sample-and-hold (S/H) filter is employed before the S/H circuit
circuit is used to ensure that the analog • To smooth the output signal of the D/A
signal at the input of the A/D converter converter, which has a staircase-like
remains constant in amplitude until the waveform, an analog reconstruction filter
conversion is complete to minimize the is used
error in its representation

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Digital Processing of
Sampling of Continuous-Time
Continuous-Time Signals
Complete block-diagram
Signals
Anti- Digital Reconstruction
• As indicated earlier, discrete-time signals in
aliasing S/H A/D processor D/A
filter filter many applications are generated by
sampling continuous-time signals
• Since both the anti-aliasing filter and the
reconstruction filter are analog lowpass • We have seen earlier that identical discrete-
filters, we review first the theory behind the time signals may result from the sampling
design of such filters of more than one distinct continuous-time
• Also, the most widely used IIR digital filter function
design method is based on the conversion of
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an analog lowpass prototype 6
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Sampling of Continuous-Time Sampling of Continuous-Time
Signals Signals
• In fact, there exists an infinite number of • If these conditions hold, then it is possible
continuous-time signals, which when to recover the original continuous-time
sampled lead to the same discrete-time signal from its sampled values
signal • We next develop this correspondence and
• However, under certain conditions, it is the associated conditions
possible to relate a unique continuous-time
signal to a given discrete-time signals

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Copyright © 2005, S. K. Mitra Copyright © 2005, S. K. Mitra

Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• Now, the frequency-domain representation of
• Let g a (t ) be a continuous-time signal that is
g a (t ) is given by its continuos-time Fourier
sampled uniformly at t = nT, generating the
transform (CTFT):
sequence g[n] where ∞
g [n] = g a (nT ), − ∞ < n < ∞ Ga ( jΩ) = ∫−∞ g a (t )e − jΩt dt
• The frequency-domain representation of g[n]
with T being the sampling period
is given by its discrete-time Fourier transform
• The reciprocal of T is called the sampling (DTFT):
frequency FT , i.e.,
G ( e jω ) = ∑ ∞
n = −∞ g[ n] e
− jω n
FT = 1
9 T 10
Copyright © 2005, S. K. Mitra Copyright © 2005, S. K. Mitra

Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• p(t) consists of a train of ideal impulses
• To establish the relation between Ga ( jΩ)
with a period T as shown below
and G (e jω ) , we treat the sampling operation
mathematically as a multiplication of g a (t )
by a periodic impulse train p(t):

p (t ) = ∑ δ(t − nT )
n = −∞ • The multiplication operation yields an
g a (t ) × g p(t ) impulse train: ∞
p (t ) g p (t ) = g a (t ) p(t ) = ∑ g a ( nT )δ(t − nT )
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Effect of Sampling in the Effect of Sampling in the
Frequency Domain
Frequency Domain
• There are two different forms of G p ( jΩ):
• g p (t ) is a continuous-time signal consisting
of a train of uniformly spaced impulses with • One form is given by the weighted sum of
the impulse at t = nT weighted by the the CTFTs of δ(t − nT ) :
sampled value g a (nT ) of g a (t ) at that instant G p ( jΩ ) = ∑ ∞
n = −∞ g a ( nT ) e
− jΩnT

• To derive the second form, we note that p(t)


can be expressed as a Fourier series:
p(t ) = 1 ∑∞ e j ( 2π / T )kT = 1 ∑∞ e jΩT kt
T k = −∞ T k = −∞
where ΩT = 2π / T
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Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• The impulse train g p (t ) therefore can be • Hence, an alternative form of the CTFT of
expressed as g p (t ) is given by

⎛ ∞ ⎞
g p (t ) = ⎜⎜ 1 ∑ e jΩT kt ⎟⎟ ⋅ g a (t )
T
G p ( jΩ ) = 1
T ∑ Ga ( j (Ω − kΩT ) )
⎝ k = −∞ ⎠ k = −∞

• From the frequency-shifting property of the • Therefore, G p ( jΩ) is a periodic function of


CTFT, the CTFT of e jΩT kt g a (t ) is given by Ω consisting of a sum of shifted and scaled
Ga ( j (Ω − kΩT ) ) replicas of Ga ( jΩ) , shifted by integer
multiples of ΩT and scaled by 1
15 16 T
Copyright © 2005, S. K. Mitra Copyright © 2005, S. K. Mitra

Effect of Sampling in the


Effect of Sampling in the
Frequency Domain
Frequency Domain
• Assume g a (t ) is a band-limited signal with a
• The term on the RHS of the previous CTFT Ga ( jΩ) as shown below
equation for k = 0 is the baseband portion
of G p ( jΩ) , and each of the remaining terms
are the frequency translated portions of
G p ( jΩ ) • The spectrum P ( jΩ) of p(t) having a
sampling period T = Ω2 π is indicated below
• The frequency range T
Ω Ω
− T ≤Ω≤ T
2 2
• is called the baseband or Nyquist band
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Effect of Sampling in the
Frequency Domain Effect of Sampling in the
• Two possible spectra of G p ( jΩ) are shown
Frequency Domain
below • It is evident from the top figure on the
previous slide that if ΩT > 2Ω m , there is no
overlap between the shifted replicas of Ga ( jΩ)
generating G p ( jΩ)
• On the other hand, as indicated by the figure
on the bottom, if ΩT < 2Ω m , there is an
overlap of the spectra of the shifted replicas
of Ga ( jΩ) generating G p ( jΩ)
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Effect of Sampling in the


Effect of Sampling in the
Frequency Domain
Frequency Domain • The spectra of the filter and pertinent
If ΩT > 2Ω m , g a (t ) can be signals are shown below
recovered exactly from g a (t ) by passing it
through an ideal lowpass filter H r ( jΩ) with
a gain T and a cutoff frequency Ωc greater
than Ω m and less than ΩT − Ω m as shown
below

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Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• On the other hand, if ΩT < 2Ω m , due to the Sampling theorem - Let g a (t ) be a band-
overlap of the shifted replicas of Ga ( jΩ) , limited signal with CTFT Ga ( jΩ) = 0 for
the spectrum Ga ( jΩ) cannot be separated by Ω > Ωm
filtering to recover Ga ( jΩ) because of the • Then g a (t ) is uniquely determined by its
distortion caused by a part of the replicas samples g a (nT ) , − ∞ ≤ n ≤ ∞ if
immediately outside the baseband folded
back or aliased into the baseband ΩT ≥ 2Ω m
where ΩT = 2π / T
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Effect of Sampling in the Effect of Sampling in the
Frequency Domain Frequency Domain
• The condition ΩT ≥ 2Ω m is often referred to • Given {g a (nT )}, we can recover exactly g a (t )
as the Nyquist condition by generating an impulse train
• The frequency ΩT is usually referred to as g p (t ) = ∑∞
n = −∞ g a ( nT )δ(t − nT )
2
the folding frequency and then passing it through an ideal lowpass
filter H r ( jΩ) with a gain T and a cutoff
frequency Ωc satisfying
Ω m < Ωc < (ΩT − Ω m )

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Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• Oversampling - The sampling frequency is
• The highest frequency Ω m contained in g a (t ) higher than the Nyquist rate
is usually called the Nyquist frequency
• Undersampling - The sampling frequency is
since it determines the minimum sampling
lower than the Nyquist rate
frequency ΩT = 2Ω m that must be used to
fully recover g a (t ) from its sampled version • Critical sampling - The sampling frequency
is equal to the Nyquist rate
• The frequency 2Ω m is called the Nyquist
rate • Note: A pure sinusoid may not be
recoverable from its critically sampled
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version
Copyright © 2005, S. K. Mitra Copyright © 2005, S. K. Mitra

Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• In digital telephony, a 3.4 kHz signal • In high-quality analog music signal
bandwidth is acceptable for telephone processing, a bandwidth of 20 kHz has been
conversation determined to preserve the fidelity
• Here, a sampling rate of 8 kHz, which is • Hence, in compact disc (CD) music
greater than twice the signal bandwidth, is systems, a sampling rate of 44.1 kHz, which
used is slightly higher than twice the signal
bandwidth, is used

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Effect of Sampling in the
Effect of Sampling in the
Frequency Domain
Frequency Domain
• These three transforms are plotted below
• Example - Consider the three continuous-
time sinusoidal signals:
g1(t ) = cos(6πt )
g 2 (t ) = cos(14πt )
g3 (t ) = cos( 26πt )
• Their corresponding CTFTs are:
G1( jΩ) = π[δ(Ω − 6π) + δ(Ω + 6π)]
G2 ( jΩ) = π[δ(Ω − 14π) + δ(Ω + 14π)]
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G3 ( jΩ) = π[δ(Ω − 26π) + δ(Ω + 26π)] 32
Copyright © 2005, S. K. Mitra Copyright © 2005, S. K. Mitra

Effect of Sampling in the Effect of Sampling in the


Frequency Domain
Frequency Domain
• Plots of the 3 CTFTs are shown below
• These continuous-time signals sampled at a
rate of T = 0.1 sec, i.e., with a sampling
frequency ΩT = 20π rad/sec
• The sampling process generates the
continuous-time impulse trains, g1 p (t ) ,
g 2 p (t ) , and g3 p (t )
• Their corresponding CTFTs are given by
Glp ( jΩ) = 10∑∞
k = −∞ Gl ( j (Ω − kΩT ) ), 1 ≤ l ≤ 3
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Effect of Sampling in the Effect of Sampling in the


Frequency Domain
Frequency Domain
• These figures also indicate by dotted lines
the frequency response of an ideal lowpass • Moreover, the reconstructed output is
filter with a cutoff at Ωc = ΩT / 2 = 10π and precisely the original continuous-time
a gain T = 0.1 signal
• The CTFTs of the lowpass filter output are • In the other two cases, the sampling rate
also shown in these three figures does not satisfy the Nyquist condition,
resulting in aliasing and the filter outputs
• In the case of g1(t ), the sampling rate
are all equal to cos(6πt)
satisfies the Nyquist condition, hence no
aliasing
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Effect of Sampling in the
Effect of Sampling in the
Frequency Domain
Frequency Domain
• Note: In the figure below, the impulse
appearing at Ω = 6π in the positive • We now derive the relation between the
frequency passband of the filter results from DTFT of g[n] and the CTFT of g p (t )
the aliasing of the impulse in G2 ( jΩ) at • To this end we compare
Ω = −14π G ( e jω ) = ∑ ∞ − jω n
n = −∞ g[ n] e
• Likewise, the impulse appearing at Ω = 6π with
in the positive frequency passband of the G p ( jΩ ) = ∑ ∞n = −∞ g a ( nT ) e
− jΩnT
filter results from the aliasing of the impulse and make use of g[n] = g a (nT ), − ∞ < n < ∞
in G3 ( jΩ) at Ω = 26π
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Effect of Sampling in the Effect of Sampling in the


Frequency Domain Frequency Domain
• Observation: We have we arrive at the desired result given by
G ( e jω ) = G p ( jΩ ) ∞
G (e jω ) = T1 ∑ Ga ( jΩ − jkΩT )
Ω =ω / T
or, equivalently, k = −∞ Ω =ω / T

G p ( jΩ ) = G ( e jω ) = 1 ∑ Ga ( j ω − jkΩT )
ω=ΩT T T
• From the above observation and k = −∞
∞ ∞
G p ( jΩ ) = 1
T ∑ Ga ( j (Ω − kΩT ) ) = 1 ∑ Ga ( j ω − j 2 π k )
T T T
k = −∞ k = −∞

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Effect of Sampling in the Effect of Sampling in the


Frequency Domain
Frequency Domain
• The relation derived on the previous slide
can be alternately expressed as • Now, the CTFT Gp ( jΩ) is a periodic
function of Ω with a period ΩT = 2π / T
G (e jΩT ) = T1 ∑∞ k = −∞ Ga ( jΩ − jkΩT )
• From • Because of the mapping, the DTFT G (e jω )
G ( e jω ) = G p ( jΩ ) is a periodic function of ω with a period 2π
Ω =ω / T
or from
G p ( jΩ ) = G ( e jω )
ω=ΩT
it follows that G (e jω ) is obtained from Gp ( jΩ)
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by applying the mapping Ω = ω 42
T
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Recovery of the Analog Signal Recovery of the Analog Signal
• We now derive the expression for the output
g^ a (t ) of the ideal lowpass reconstruction • Thus, the impulse response is given by
filter H r ( jΩ) as a function of the samples hr (t ) = 1 ∞ H ( jΩ) e jΩt dΩ = T Ω c e jΩt dΩ

2 π −∞ r 2 π −Ωc ∫
g[n]
sin(Ωct )
• The impulse response hr (t ) of the lowpass = , −∞ ≤t ≤ ∞
ΩT t / 2
reconstruction filter is obtained by taking • The input to the lowpass filter is the
the inverse DTFT of H r ( jΩ): impulse train gp(t ) :
T , Ω ≤ Ωc
H r ( jΩ) = ⎧⎨ g p (t ) = ∑∞
⎩ 0, Ω > Ωc n = −∞ g[ n] δ(t − nT )
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Recovery of the Analog Signal Recovery of the Analog Signal


• Therefore, the output g^ a (t ) of the ideal • The ideal bandlimited interpolation process
lowpass filter is given by: is illustrated below

g^ a (t ) = hr (t ) * g p (t ) = ∑ g[n]hr (t − nT )
n = −∞

• Substituting hr (t ) = sin(Ωct ) /(ΩT t / 2) in the


above and assuming for simplicity
Ωc = ΩT / 2 = π / T , we get
∞ sin[ π(t − nT ) / T ]
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g^ (t ) =
a ∑ g[ n]
π(t − nT ) / T Copyright © 2005, S. K. Mitra 46
n = −∞ Copyright © 2005, S. K. Mitra

Recovery of the Analog Signal Recovery of the Analog Signal


• It can be shown that when Ωc = ΩT / 2 in
sin( Ω ct )
hr (t ) = • The relation
ΩT t / 2
h r(0) = 1 and h r(nT ) = 0 for n ≠ 0 g^ a (rT ) = g[r ] = g a (rT )
• As a result, from holds whether or not the condition of the
sin[ π(t − nT ) / T ]
g^ a (t ) = ∑∞n = −∞ g[ n] π(t − nT ) / T
sampling theorem is satisfied
we observe • However, g^a (rT ) = g a (rT ) for all values of
g a (rT ) = g[r ] = g a (rT ) t only if the sampling frequency ΩT satisfies
the condition of the sampling theorem
for all integer values of r in the range
47 −∞ < r <∞ 48
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Implication of the Sampling Implication of the Sampling
Process Process
• Consider again the three continuous-time • From the plot, it is apparent that we can
signals: g1(t ) = cos(6πt ) , g 2 (t ) = cos(14πt ) , recover any of its frequency-translated
and g3 (t ) = cos(26πt ) versions cos[(20k ± 6)π t] outside the
• The plot of the CTFT G1p ( jΩ) of the baseband by passing g1p (t ) through an ideal
sampled version g1p (t ) of g1(t ) is shown analog bandpass filter with a passband
below centered at Ω = ( 20k ± 6) π

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Implication of the Sampling Implication of the Sampling


Process Process
• For example, to recover the signal cos(34πt), • Likewise, we can recover the aliased
it will be necessary to employ a bandpass baseband component cos(6πt) from the
filter with a frequency response sampled version of either g 2 p (t ) or g3 p (t )
0.1, (34 − ∆ )π ≤ Ω ≤ (34 + ∆ )π by passing it through an ideal lowpass filter
H r ( jΩ) = ⎧⎨ with a frequency response:
⎩ 0, otherwise
0.1, (6 − ∆ )π ≤ Ω ≤ (6 + ∆ )π
where ∆ is a small number H r ( jΩ) = ⎨⎧
⎩ 0, otherwise

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Implication of the Sampling Sampling of Bandpass Signals


Process
• The conditions developed earlier for the
• There is no aliasing distortion unless the unique representation of a continuous-time
original continuous-time signal also signal by the discrete-time signal obtained
contains the component cos(6πt) by uniform sampling assumed that the
• Similarly, from either g 2 p (t ) or g3 p (t ) we continuous-time signal is bandlimited in the
can recover any one of the frequency- frequency range from dc to some frequency
translated versions, including the parent Ωm
continuous-time signal g 2(t ) or g3(t ) as the • Such a continuous-time signal is commonly
case may be, by employing suitable filters referred to as a lowpass signal
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Sampling of Bandpass Signals Sampling of Bandpass Signals
• There are applications where the continuous-
time signal is bandlimited to a higher • However, due to the bandpass spectrum of
frequency range Ω L ≤ Ω ≤ Ω H with Ω L > 0 the continuous-time signal, the spectrum of
the discrete-time signal obtained by sampling
• Such a signal is usually referred to as the
will have spectral gaps with no signal
bandpass signal
components present in these gaps
• To prevent aliasing a bandpass signal can of
• Moreover, if Ω H is very large, the sampling
course be sampled at a rate greater than
rate also has to be very large which may not
twice the highest frequency, i.e. by ensuring
be practical in some situations
ΩT ≥ 2Ω H
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Sampling of Bandpass Signals Sampling of Bandpass Signals


• A more practical approach is to use under- • We choose the sampling frequency ΩT to
sampling satisfy the condition
2Ω
• Let ∆Ω = Ω H − Ω L define the bandwidth of ΩT = 2(∆Ω) = H
M
the bandpass signal which is smaller than 2Ω H , the Nyquist
• Assume first that the highest frequency Ω H rate
contained in the signal is an integer multiple • Substitute the above expression for ΩT in
of the bandwidth, i.e., ∞
Ω H = M (∆Ω) G p ( jΩ ) = 1
T ∑ Ga ( j (Ω − k ΩT ))
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k = −∞
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Sampling of Bandpass Signals Sampling of Bandpass Signals


• This leads to • Figure below illustrate the idea behind
G p ( jΩ ) = 1 ∑ ∞ (
k = −∞ Ga jΩ − j 2k (∆Ω)
) Ga ( jΩ)
T
• As before, G p( jΩ) consists of a sum of Ga ( jΩ)
and replicas of Ga ( jΩ) shifted by integer

multiples of twice the bandwidth ∆Ω and − ΩH − ΩL 0 ΩL ΩH

scaled by 1/T G p ( jΩ )

• The amount of shift for each value of k


ensures that there will be no overlap

− ΩH − ΩL ΩL ΩH
between all shifted replicas no aliasing 0
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Sampling of Bandpass Signals
• As can be seen, g a (t ) can be recovered from
g p (t ) by passing it through an ideal
bandpass filter with a passband given by
Ω L ≤ Ω ≤ Ω H and a gain of T
• Note: Any of the replicas in the lower
frequency bands can be retained by passing
g p (t ) through bandpass filters with
passbands Ω L − k ( ∆Ω) ≤ Ω ≤ Ω H − k (∆Ω) ,
1 ≤ k ≤ M − 1 providing a translation to
61 lower frequency ranges
Copyright © 2005, S. K. Mitra

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