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Chapter 4

Digital Processing of Continuous-Time Signals

Part A
Sampling of Continuous-Time Signals

Part A: Sampling of Continuous-Time Signals Introduction Sampling of Continuous-Time Signals ContinuousEffect of Sampling in the Frequency Domain Recovery of the Analog Signal Implication of the Sampling Process

1. Introduction
Digital processing of a continuous-time signal involves the following basic steps: (1) Conversion of the continuous-time signal into a discrete-time signal, (2) Processing of the discrete-time signal, (3) Conversion of the processed discrete-time signal back into a continuous-time signal

Sampling of Bandpass Signals

1. Introduction
Conversion of a continuous-time signal into digital form is carried out by an analog-toanalog- todigital (A/D) converter The reverse operation of converting a digital signal into a continuous-time signal is performed by a digital-to-analog (D/A) digital- toconverter

1. Introduction
Since the A/D conversion takes a finite amount of time, a sample-and-hold (S/H) sample- andcircuit is used to ensure that the analog signal at the input of the A/D converter remains constant in amplitude until the conversion is complete to minimize the error in its representation To prevent aliasing, an analog anti-aliasing aliasing antifilter is employed before the S/H circuit.

1. Introduction
To smooth the output signal of the D/A converter, which has a staircase-like waveform, an analog reconstruction filter is used. The complete block-diagram is shown blew
Anti-aliasing filter S/H A/D Digital Processing D/A Reconstruction filter

1. Introduction
Since both the anti-aliasing filter and the reconstruction filter are analog lowpass filters, we review first the theory behind the design of such filters Also, the most widely used IIR digital filter design method is based on the conversion of an analog lowpass prototype

2. Sampling of Continuous-Time Signals


As indicated earlier, discrete-time signals in many applications are generated by sampling continuous-time signals It is obvious that identical discrete time signals may result from the sampling of more than one distinct continuous-time function In fact, there exists an infinite number of continuous-time signals, which when sampled lead to the same discrete-time signal

2. Sampling of Continuous-Time Signals


1 g1(t) g1(n) g2(t) g2(n) g3(t) g3(n)

0.5

Amplitude

-0.5

-1

0.2

0.4

0.6

0.8

Time

2. Sampling of Continuous-Time Signals


However, under certain conditions, it is possible to relate a unique continuous-time signal to a given discrete-time signals If these conditions hold, then it is possible to recover the original continuous-time signal from its sampled values We next develop this correspondence and the associated conditions

2.1 Effect of Sampling in the Frequency Domain

Let ga(t) be a continuous-time signal that is sampled uniformly at t = nT, generating the sequence g(n) where g(n)= ga(nT) with T being the sampling period The reciprocal of T is called the sampling frequency FT , i.e., FT = 1/ T Now, the frequency-domain representation of ga(t) is given by its continuous-time Fourier transform (CTFT CTFT):

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

Ga ( j) = g a (t )e jt dt

The frequency-domain representation of g(n) is given by its discrete-time Fourier transform (DTFT DTFT): G (e j ) = g (n)e j n
n =

p(t) consists of a train of ideal impulses with a period T as shown below


p(t) ga(t) p(t) gp(t)

2T T

0 T 2T t

To establish the relation between Ga ( j) and G (e j ) , we treat the sampling operation mathematically as a multiplication of ga(t) by a periodic impulse train p(t):

The multiplication operation yields an impulse train: g p (t ) = g a (t ) p (t ) = g a ( nT ) (t nT )


n =

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

ga(t) is a continuous-time signal consisting of a train of uniformly spaced impulses with the impulse at t = nT weighted by the sampled value ga(nT) of ga(t) at that instant
ga(t) gp(t) ga(t) 0

There are two different forms of G p ( j) : One form is given by the weighted sum of the CTFTs of (t nT ) : G p ( j) = g a (nT )e jnT
n =

To derive the second form, we note that p(t) can be expressed as a Fourier series: 1 + j 2 kt 1 + p ( t ) = e T = e jT kt T k = T k = where 2 T = T

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

The impulse train gp(t) therefore can be expressed as 1 + g p (t ) = e jT kt g a (t ) T k = From the frequency-shifting property of the CTFT, the CTFT of e jT kt g a (t ) is given by Ga ( j ( k T )) Hence, an alternative form of the CTFT of is given by 1 G p ( j) = Ga ( j ( k T )) T k =

Therefore, G p ( j) is a periodic function of consisting of a sum of shifted and scaled replicas of Ga ( j) , shifted by integer multiples of T and scaled by 1/T The term on the RHS of the previous equation for k = 0 is the baseband portion of G p ( j) , and each of the remaining terms are the frequency translated portions of G p ( j)

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

The frequency range T / 2 T / 2 is called the baseband or Nyquist band The above result is more commonly known as the sampling theorem: Let ga(t) be a bandlimited signal with Ga ( j) = 0 for > m ,then ga(t) is uniquely determined by its samples ga(nT), n ,if 2 T 2 m where T = T

Illustration of the frequency-domain effects of time-domain sampling


Ga ( j)

1
m 0

G p ( j)

P ( j)

2T

3T

1/T
m

T m

T
2T

0 m
G p ( j )

2T

1/T
T m

T m m T T
2T

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

If T 2 m , ga(t) can be recovered exactly from gp(t) by passing it through an ideal lowpass filter H r ( j) with a gain T and a cutoff frequency c greater than m and less than T m as shown below
ga(t) gp(t)

The spectra of the filter and pertinent signals are shown below G ( j)
p

1/T
m

T m

T
2T

0 m T
H r ( j )

m < c < ( T m )

H r ( j)

g a (t )

c Ga ( j)

p(t)

1
m 0 m

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain


T 2 m T / 2 m 2m

On the other hand, if T < 2 m , due to the overlap of the shifted replicas of Ga ( j) , the spectrum Ga ( j) cannot be separated by filtering to recover because of the distortion caused by a part of the replicas Ga ( j) immediately outside the baseband folded back or aliased into the baseband. Several terms: T 2 m

Nyquist condition Folding frequency Nyquist frequency Nyquist rate Oversampling Undersampling Critical sampling

When T > 2 m When T < 2 m When T = 2 m

Note: A pure sinusoid may not be recoverable from its critically sampled version

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

Example 1 In high-quality analog music signal processing, a bandwidth of 20 kHz has been determined to preserve the fidelity Hence, in compact disc (CD music systems, CD) a sampling rate of 44.1 kHz, which is slightly higher than twice the signal bandwidth, is used

Example 2 Consider the three continuous time sinusoidal signals: g1 (t ) = cos(6 t ), g 2 (t ) = cos(14 t ), g3 (t ) = cos(26 t ) Their corresponding CTFTs are:
G1 ( j) = [ ( 6 ) + ( + 6 ) ] G2 ( j) = [ ( 14 ) + ( + 14 ) ] G3 ( j) = [ ( 26 ) + ( + 26 )]

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

These three transforms are plotted below


G1 ( j)

6 0 G2 ( j)

14

14

G3 ( j)

These continuous-time signals sampled at a rate of T = 0.1 sec, i.e., with a sampling frequency T = 20 rad/sec The sampling process generates the continuous-time impulse trains, g1p(t), g2p (t), and g3p (t) Their corresponding CTFTs are given by
Glp ( j) = 10 Gl ( j ( k T )), 1 l 3

26

26

k =

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

Plots of the 3 CTFTs are shown below


G1 p ( j)

Spectrum lines painted by Blue and Green Colors designate aliases

10

H r ( j)

0.1 c 20 6 0 G2 p ( j) 0.1 c 20 14 6 0
G3 p ( j)

20

10 H r ( j)

c 14 20

The cutoff frequency of the lowpass filter is chosen as

We now derive the relation between the DTFT of g(n) and the CTFT of gp(t) To this end we compare
G (e j ) =

c = 10

with

10 H r ( j)

c
40

0.1
6 0
6

c
20

G p ( j) = g a (nT )e jnT n = and make use of g ( n) = g a (nT ), < n <

n =

g ( n )e

j n

20

40

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

Observation: We have G (e j ) = G p ( j) = / T or, equivalently, G p ( j) = G (e j ) =T From the above observation and 1 G p ( j) = Ga ( j ( k T )) T k =

we arrive at the desired result given by 1 G (e j ) = Ga ( j jk T ) T k = = / T


= = 1 Ga j jk T T k = T 1 2 k Ga j j T T k = T

2.1 Effect of Sampling in the Frequency Domain

2.1 Effect of Sampling in the Frequency Domain

The relation derived on the previous slide can be alternately expressed as 1 G (e jT ) = Ga ( j jk T ) T k = form G (e j ) = G p ( j) = / T or from G p ( j) = G (e j ) =T

It follows that G (e j ) is obtained from G p ( j) by applying the mapping = / T Now, the CTFT G p ( j) is a periodic function of with a period T = 2 / T Because of the mapping, the DTFT G (e j ) is a periodic function of with a period 2

2.2 Recovery of the Analog Signal

2.2 Recovery of the Analog Signal

We now derive the expression for the output ga ( fsdft ) of the ideal lowpass reconstruction filter H r ( j) as a function of the samples g(n) The impulse response hr(t) of the lowpass reconstruction filter is obtained by taking the inverse DTFT of : T , c H r ( j ) = 0, > c

Thus, the impulse response is given by 1 T c jt j t hr (t ) = H r ( j)e d = 2 c e d 2 sin( c t ) , t = ct / 2 The input to the lowpass filter is the impulse train gp(t) g p (t ) = g ( n) (t nT )
n =

2.2 Recovery of the Analog Signal

2.2 Recovery of the Analog Signal

Therefore, the output g a (t ) of the ideal lowpass filter is given by:


ga (t ) = g p (t ) hr (t ) =
n=

g(n)h (t nT )
r

Substituting hr (t ) = sin( ct ) / c t / 2 in the above and assuming c = T / 2 = / T for simplicity, we get sin[ (t nT ) / T ] g a (t ) = g ( n) (t nT ) / T n =

It can be shown that when c = T / 2 in sin c t hr (t ) = T t / 2 hr(0)=1 and hr(nT)=0 for n0 As a result, from sin[ (t nT ) / T ] g a (t ) = g ( n) (t nT ) / T n = we observe g a (rT ) = g (r ) = g a (rT ) < r < for all integer values of r in the range

2.2 Recovery of the Analog Signal

2.3 Implication of the Sampling Process

The relation g a (rT ) = g (r ) = g a (rT ) holds whether or not the condition of the sampling theorem is satisfied However, g a (t ) = g a (t ) for all values of t only if the sampling frequency T satisfies the condition of the sampling theorem

Consider again the three continuous-time signals: g1 (t ) = cos(6 t ) , g 2 (t ) = cos(14 t ) , and g3 (t ) = cos(26 t ) The plot of the CTFT G1 p ( j) of the sampled version of g1(t) is shown below
G1 p ( j)
0.1

10 H r ( j)

20

c 6

20

2.3 Implication of the Sampling Process

2.3 Implication of the Sampling Process

From the plot, it is apparent that we can recover any of its frequency-translated versions cos ( 20k 6 ) t outside the baseband by passing through an ideal analog bandpass filter g1p(t) with a passband centered at = ( 20k 6 ) For example, to recover the signal cos(34 t ) cos(34pt), it will be necessary to employ a bandpass filter with a frequency response 0.1, (34 ) (34 + ) H r ( j ) = A small number otherwise 0,

Likewise, we can recover the aliased baseband component cos(6 t ) from the cos(6pt) sampled version of either g2p(t) or g3p(t) by passing it through an ideal lowpass filter with a frequency response:
0.1, 0 (6 + ) H r ( j ) = otherwise 0,

2.3 Implication of the Sampling Process

3. Sampling of Bandpass Signals


The conditions developed earlier for the unique representation of a continuous-time signal by the discrete-time signal obtained by uniform sampling assumed that the continuous-time signal is bandlimited in the frequency range from dc to some frequency Such a continuous-time signal is commonly referred to as a lowpass signal

There is no aliasing distortion unless the original continuous-time signal also contains the component cos(6 t ) Similarly, from either g2p(t) or g3p(t) we can recover any one of the frequency-translated versions, including the parent continuous-time signal cos(14t) or cos(26t) as the case may be, by employing suitable filters

3. Sampling of Bandpass Signals


There are applications where the continuoustime signal is bandlimited to a higher frequency range L H with L > 0 a Such a signal is usually referred to as the bandpass signal To prevent aliasing a bandpass signal can of course be sampled at a rate greater than twice the highest frequency, i.e. by ensuring T 2 H

3. Sampling of Bandpass Signals


However, due to the bandpass spectrum of the continuous-time signal, the spectrum of the discrete-time signal obtained by sampling will have spectral gaps with no signal components present in these gaps Moreover, if H is very large, the sampling rate also has to be very large which may not be practical in some situations

3. Sampling of Bandpass Signals


A more practical approach is to use underundersampling Let = H L define the bandwidth of the bandpass signal Assume first that the highest frequency H contained in the signal is an integer multiple of the bandwidth, i.e., H = M ()

3. Sampling of Bandpass Signals


We choose the sampling frequency T to satisfy the condition 2 H T = 2() = M which is smaller than 2 H , the Nyquist rate Substitute the above expression for T in 1 G p ( j) = Ga ( j jk T ) T k =

3. Sampling of Bandpass Signals


1 Ga ( j j 2k () ) T k = As before, G p ( j) consists of a sum of Ga ( j) and replicas of Ga ( j) shifted by integer multiples of twice the bandwidth DWand scaled by 1/T The amount of shift for each value of k ensures that there will be no overlap between all shifted replicas no aliasing

3. Sampling of Bandpass Signals


Figure below illustrate the idea behind
Ga ( j)

This leads to G p ( j) =

0
G p ( j)

3. Sampling of Bandpass Signals


As can be seen, ga(t) can be recovered from gp(t) by passing it through an ideal bandpass filter with a passband given by L H and a gain of T Note: Any of the replicas in the lower frequency bands can be retained by passing gp(t) through bandpass filters with passbands L k () H k () , providing a translation to lower frequency ranges 1 k M 1

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