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Even though this course is primarily about the discrete time signal processing, most signals we encounter in

daily life are continuous in time such as speech, music and images. Increasingly discrete-time signals

processing algorithms are being used to process such signals. For processing by digital systems, the discrete

time signals are represented in digital form with each discrete time sample as binary word. Therefore we need

the analog to digital and digital to analog interface circuits to convert the continuous time signals into discrete

time digital form and vice versa. As a result it is necessary to develop the relations between continuous time

and discrete time representations.

Let be a continuous time signal that is sampled uniformly at t = nT generating the sequence

where

T is called sampling period, the reciprocal of T is called the sampling frequency. The frequency domain

representation of is given by its Fourier transform

where the frequency-domain representation of is given by its discrete time fourier transform

To establish relationship between the two representation, we use impulse train sampling. This should be

understood as mathematically convenient method for understanding sampling. Actual circuits can not produce

continuous time impulses. A periodic impulse train is given by

(8.1)

Fig 8.1

(8.2)

(8.3)

Fig 8.2

where

(8.4)

Thus is a periodic function of with period , consisting of superposition of shifted replicas

of scaled by. Figure 8.3 illustrates this for two cases.

Fig 8.3

can be recovered from by means of lowpass filtering with gain T and cut off frequency between

and. This result is known as Nyquist sampling theorem.

Sampling Theorem

samples , if

The frequency is called Nyquist rate, while the frequency is called the Nyquist frequency. The

Fig 8.4

(8.5)

Assuming we get

(8.6)

The above expression (8.5) shows that reconstructed continuous time signal is obtained by shifting in

time the impulse response of low pass filter by an amount nT and scaling it in amplitude by a

factor for all integer values n. The interpolation using the impulse response of an ideal low pass

filter in (8.6) is referred to as bandlimited interpolation, since it implements reconstruction if is

bandlimited and sampling frequency satisfies the condition of the sampling theorem. The reconstruction is in the

mean square sense i.e.

1 2 3 4 5 6 7

Chapter 8 : Discrete time processing of continous time signals

We have seen earlier that spectrum is not faithfully copied when. The terms in (8.4) overlap. The

signal is no longer recoverable from. This effect, in which individual terms in equation (8.4) overlap is

called aliasing.

Hence

Thus at the sampling instants the signal values of the original and reconstructed signal are same for any

sampling frequency.

1 2 3 4 5 6 7

Chapter 8 : Discrete time processing of continous time signals

DTFT of the discrete time signal

(8.7)

(8.8)

or

(8.9)

Comparing equation (8.4) and(8.9) we see that is simply a frequency scaled version of with

frequency scaling specified by. This can be thought of as a normalization of frequency axis so that

frequency in is normalized to in. For the example in figure 8.3 the is shown

in figure (8.5)

From equation (8.5) we see that

Fig 8.5

(8.10)

We refer to the system that implements as ideal continuous-to-discrete time (C/D) convertor

and is depicted in figure (8.6)

1 2 3 4 5 6 7

Chapter 8 : Discrete time processing of continous time signals

Discrete time processing of continuous time signal

Figure (8.8) shows a system for discrete time processing of continuous time system

Fig 8.8

Signals in Natural Domain

Chapter 8 : Discrete time processing of continous time signals

Figure 8.10

We have

Therefore the overall system behaves as a discrete time system where frequency response is

(8.13)

Example

when is an integer, this system is delay by

but when is not an integer, we can not write the above equation. Suppose that we implement this using

system in figure (8.10). Then we have

(8.14)

So that overall system has frequency response. The equation (8.13) represents a time delay secs in

continuous time whether is integer or not, thus

The signal is bandlimited interpolation of and is obtained by sampling. Thus y[n] are samples

of band limited signal delayed by.

1 2 3 4 5 6 7

Chapter 8 : Discrete time processing of continous time signals

In analogy with continuous time sampling, the sampling of a discrete time signal can be represented as shown

in figure 8.12

FIGURE 8.12

(8.15)

where

Thus we get

(8.16)

If or equivalently or there will be no aliasing (i.e non zero portions

of do not overlap) and the signal can be recovered from by passing through an ideal

low-pass filter with gain equal to N and cut off equal to

FIGURE 8.14

If , there will be aliasing, and so will be different from. However as in continuous time case

with we get

1 2 3 4 5 6 7

Chapter 8 : Discrete time processing of continous time signals

In analogy with continuous time sampling, the sampling of a discrete time signal can be represented as shown

in figure 8.12

FIGURE 8.12

(8.15)

where

Thus we get

(8.16)

If or equivalently or there will be no aliasing (i.e non zero portions

of do not overlap) and the signal can be recovered from by passing through an ideal

low-pass filter with gain equal to N and cut off equal to

FIGURE 8.14

If , there will be aliasing, and so will be different from. However as in continuous time case

with we get

1 2 3 4 5 6 7

of do not overlap) and the signal can be recovered from by passing through an ideal

low-pass filter with gain equal to N and cut off equal to

FIGURE 8.14

If , there will be aliasing, and so will be different from. However as in continuous time case

independently of whether there is aliasing or not.

1 2 3 4 5 6 7

Sign

Cha

Discrete time decimation and interpolation

sequence which retains only the non zero values

by

Fig 8.15

be viewed as reduction in the sampling rate by a factor of N.

often referred as down sampling. The block diagram for this i

Fig 8.16

process is referred to as upsampling or interpolation. This

sequence we obtain an expanded sequence

The interpolated sequence is obtained by low pass

1 2 3 4 5

1 2 3 4 5 6 7

Fig 8.17

We can get a non integer change in rate if it is ratio of two integers by using upsampling and downsampling

operations.

1 2 3 4 5 6 7

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