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**Prof. Dr. Ljubiša Stanković
**

University of Montenegro, Montenegro

Digital signal processing

Digital signal processing

CHAPTERS:

1. Discrete-time signals and systems

2. Discrete Fourier transform

3. z-transform

4. Spectral estimation

5. Time-frequency analysis

6. Multidimensional signal processing

Discrete-Time signals and systems

Discrete signals

Discrete signal x[n] can be represented as a

sequence of real or complex numbers, where with

n is denoted n – th number.

Some important and very often used discrete

signals (sequence):

Unite impulse

Unite impulse is called also delta function or

simply impulse, and defined as:

Discrete-Time signals and systems

Unite step

A discrete time unite step function is defined to be:

Real exponential functions

A real discrete-time exponential function is given

by:

Discrete-Time signals and systems

Sinusoidal and complex exponential signals

Periodicity

A signal x[n] is said to be periodic with N if x[n] =

x[n+N]. A complex exponential sequence is

periodic only if 2π/ω

0

is integer number or if it is

rational number p/q.

0

jw n

e

Discrete-Time signals and systems

Examples

Solutions:

a)

for k = 1 N = 3.

Discrete-Time signals and systems

b)

for k =7 we have N = 6.

c)

From above equation we can conclude that

sequence is not periodic.

Discrete-Time signals and systems

It is important to note that an arbitrary sequence

can be written in the form:

Some important definition for a discrete signals

1. Magnitude of signal:

2. Energy of a signal:

Discrete-Time signals and systems

Linear shift-invariant discrete systems

Discrete systems can be described by transform

which maps output sequence x[n] into output

equence y[n]:

where T is operator which denote transform of a

system.

Discrete-Time signals and systems

Linear system

System is linear if:

Examples:

Consider following systems and check their linearity:

Discrete-Time signals and systems

Solutions:

Thus, the system is not linear.

Thus, the system is linear.

Discrete-Time signals and systems

Shift invariance

A system is shift invariant if the characteristics of

system are not function of time, i.e. if a input signal

x(n) produce output signal is y(n), than input

signal x(n- N) will produce output signal y(n − N)

Examples

Consider systems:

Discrete-Time signals and systems

Solutions:

Obviously, the system is shift-invariant.

The system is not shift-invariant.

Discrete-Time signals and systems

Description of a linear shift-invariant systems through

a response of a systems on the unite impulse:

If a systems are shift-invariant than:

This is convolution sum and it can be written as:

It is very easy to show:

Discrete-Time signals and systems

Causality and stability

System is causal if :

System is stabile if for we have

for any value n.

Proof:

From above equation follows:

Discrete-Time signals and systems

Difference equations

The general form of linear difference equations

with constant coefficients is presented by:

For this equation there is exist family of solutions,

so it is necessary to set initial conditions for

uniquely solution. If we assume that system is

causal i.e. if x(n) = 0 for n < N than y(n) = 0 for n <

N, we can write:

Discrete-Time signals and systems

Example:

Consider difference equation in the form:

We will find impulse response if we take

and assume that system is causal

Discrete-Time signals and systems

This is an example of a system with infinite impulse

response-IIR system.

If we take Aj = 0 for j > 0 than in our example we

have:

This system has finite impulse response-FIR system.

Discrete-Time signals and systems

Fourier transform of a discrete signals

Consider complex exponential signal:

For this signal we can write:

With relation:

is defined Fourier transform of a discrete signal h(n).

Discrete-Time signals and systems

Thus:

In general case:

Uniform convergence:

Having in mind:

We can conclude that Fourier transform of discrete

signals is periodic with 2π.

Since is periodic we can consider it as a

Fourier’s series.

Discrete-Time signals and systems

A inverse transform can be obtained:

We have:

Examples:

By using definition compute Fourier transform for:

Discrete-Time signals and systems

Solutions:

1.

2.

In this case the amplitude characteristic will be:

Discrete-Time signals and systems

Properties of the Fourier transform of discrete

signals

1. Linearity

2. Shifting in the time

3. Modulation

Discrete-Time signals and systems

4. Convolution

By substitution m = n − k we obtain:

Discrete-Time signals and systems

5. Product

If we interchange order of integration and

summation, we have:

Finally we have:

Discrete-Time signals and systems

Sampling of continuous-time signals

According to Shannon theorem continuous signal

can be recovered from its discrete version if

discrete signal has been sampled by periodic

sampling T which satisfy condition:

T =1/(2fm)

where fm is maximal frequency of a signal.

Consider an analog signal x

a

(t) that has the

Fourier representation:

Discrete-Time signals and systems

Assume that signal xa(t) has limited bandwidth,

i.e.:

Consider now a periodic form X

p

(jω

a

) of the X

a

(jω

a

)

that has a period 2 ω

0

. The Fourier transform

X

a

(jω

a

) can be recovered from X

p

(jω

a

) if ω

0

≥ ω

a

.

Since X

p

(jω

a

) is periodic, it can be expanded in the

Taylor series:

Discrete-Time signals and systems

where is T =π/ ω

0

. The Fourier’s coefficient can be

obtained from:

If we compare last equation with equations that

give x

a

(t), we can conclude:

Finally, we have that samples of signal x

a

(t) and

X

p

(jω

a

) form a Fourier pair.

Now, we have:

Discrete-Time signals and systems

Discrete-Time signals and systems

Examples:

1. If output signal is:

and impulse response of a system is:

Compute signal on the output of the system.

Solution:

On the output of the system amplitude and phase

of the input signal will be changed.

Discrete-Time signals and systems

Namely:

Discrete-Time signals and systems

2. If:

find signal h(n).

Solution:

Discrete-Time signals and systems

3. Find the Fourier transform of the signal

h(t) = e

t

u(t) and draw its amplitude characteristic.

Write a discrete form of the signal and draw its

amplitude characteristics for the cases T = 1 and

T =0. 2.

Solution:

Thus, we have

By discretization we obtain:

and its Fourier transform:

Discrete-Time signals and systems

4. Compute the sum:

Consider the sequence with the Fourier transform:

From above equation follows:

Discrete Fourier Transform

Definition of the discrete Fourier transform

We have seen that the Fourier transform of a

discrete signal is continual and periodic function

with period 2π. If we want to use this transform in

digital signal processing, we need its discrete

version, i.e. we have to sample it in frequency

domain.

Consider the Fourier transform of a signal x(n).

Assume that X(e

jω

) is sampled with rate Δω = 2π/N,

where N is number of samples along the period.

Discrete Fourier Transform

Since samples of signal and its Fourier transform

are the transformation’s pair, we have that

sampling in the frequency domain cause periodical

signal in the time domain and vice versa.

Thus, for the discrete Fourier transform we have

the periodic signal x

p

(n) obtained from x(n).

For the sampling rate in frequency domain

Δ ω (Δ ω

a

=Δ ω /T)

we have periodic series in the time domain with

the period t

p

= N

p

T:

From the above equation we have:

Np = N

It means the following: If we want that the

periodic signal contains the original signal x(n),

discretization must be done with the same (or

greater) number of samples as a duration of the

signal x(n).

Discrete Fourier Transform

Definition of the discrete Fourier transform:

where it is assumed:

X

p

(k) is the discrete Fourier transform.

Discrete Fourier Transform

If we use notation:

the discrete Fourier transform can be written in the

form:

Or matrix form:

We see that the computation of the Discrete

Fourier Transform require approximately N

2

multiplications and additions.

Discrete Fourier Transform

Inverse discrete Fourier transform

Inverse form of the discrete Fourier transform can

be obtained by multiplication of DFT definition by

W

N

−km

. Thus we have:

Taking:

Discrete Fourier Transform

From the above equations follows the definition of

the inverse Fourier transform:

If the duration of the signal x(n) is smaller than

N we have:

Discrete Fourier Transform

i.e.

where w(n) denotes the window function defined

by:

Discrete Fourier Transform

Examples

1. Find the discrete Fourier transform of the

sequence:

Solution:

From the equation for x(n) we see that the

duration of x(n) is N

p

= 5. Thus we have to

use N ≥ N

p

=5. Taking N = 5 we have:

The function Xp(k) is periodic with N = 5 as well.

Discrete Fourier Transform

2. If we have the periodic sequence X

p

(n) with

period N

p

= N, and X

p

(k) is its DTF, find the DFT for

the same sequence, taking N

p

= 2N.

Solution

for N

p

=2N we have:

the above equation can be written in the form:

Discrete Fourier Transform

because x(n) = x(n + N) and W

2N

kN

=(−1)

k

.

Now we can write relation between X

p

(k) and

X

p

′ (k):

Xp ′ (k) = 2X

p

(k) for k even, and X

p

′ (k) = 0 for k odd.

Discrete Fourier Transform

Relationship between frequency and k − th

number in the discrete Fourier transform

If the signal x(n) is obtained by sampling of the

analog signal x

a

(t ), than the frequency of the

discrete signal ω can be represented through the

analog frequency ω

a

, by:

By discretization of the Fourier transform we have:

From the previous two equations we have:

Discrete Fourier Transform

Note that this equation holds only for k ≤ N/2 − 1.

Frequencies between N/2 − 1 and N are mapped

negative frequencies:

Discrete Fourier Transform

Zero padding

Number of samples of the discrete Fourier

transform N in frequency domain depends on the

number of samples of a signal in the time domain.

If we want to get more samples within the basic

period of the Fourier transform (interpolation), by

using the discrete Fourier transform, then it is

necessary to take more samples as a signal period.

Discrete Fourier Transform

It can be easy obtained by adding zero values at

end of the signal sequence.

Number of zero values depends from our will, i.e.,

on how many samples we want to have in the

discrete Fourier transform.

This procedure can be understood as an

interpolation of the discrete Fourier transform.

Discrete Fourier Transform

Some properties of the discrete Fourier transform -

Convolution of periodic signals

Consider following properties of the discrete

Fourier transform:

1. If :

Then:

We have the sequence:

Discrete Fourier Transform

By substitution n − m = l follows:

It can be shown that:

Discrete Fourier Transform

Convolution of periodic versions of signals

Consider signals x

p1

(n) and x

p2

(n) which are

periodic versions of signals x

1

(n) and x

2

(n).

In order to derive period for this convolution

consider the following example.

Illustrative example

In this example we will consider signals:

x

1

(n) = u(n) − u(n − 5), and x

2

(n) = u(n) − u(n − 5).

Discrete Fourier Transform

Since signals have period N = 5, then for the

DFT calculation, we can form periodicals versions

with N ≥ 5. In this example assume N = 7.

Discrete Fourier Transform

From the Figure it is easy to obtain

x

p3

(n) = x

p1

(n) ∗ x

p2

(n):

From the Figures we see that results are

different for the original signals and their periodic

versions.

The reason is in overlapping of fictive periods.

Discrete Fourier Transform

If we want to avoid this effect we have to

introduce enough number of zero values.

Namely, if either sequence has duration N then

the period has to be 2N-1, or in general if duration

of the first one is N and duration of the second one

is M, then the period has to be M N−1.

Discrete Fourier Transform

When the duration of input signal is

significantly different from the duration of

sequence of impulse response (duration of the

impulse response is significantly shorter), we can

decompose the input sequence into few

subsequence, i.e.

Discrete Fourier Transform

Discrete Fourier Transform

Thus output is obtained as:

Here we must be careful since before every

convolution we must add L − 1 zero values (it is

assumed that L duration of a subsequence).

Discrete Fourier Transform

Fast Fourier transform – FFT

Algorithm called the Fast Fourier transform or FFT

algorithm plays very important role in digital signal

processing.

This algorithm is an interesting research topic.

That is the reason why exist various forms of this

algorithm.

By using the FFT time needed for computation

of the discrete Fourier transform can significantly

be reduced comparing time for computation of the

discrete Fourier transform by definition.

Discrete Fourier Transform

In our consideration we will present an

approach that belongs to the group of algorithms

called Decimations-in-Frequency.

The aim of this decimation is to decompose

X

p

(k) Into subsequences, then further,

subsequences into subsequences etc.

For this algorithm it is necessary that the

number of samples is of the form:

Discrete Fourier Transform

Now decompose sequence X

p

(k) into two

sequences:

Discrete Fourier Transform

From the previous equation we have:

Having in mind:

and that summations in both terms are from 0 to

N/2 − 1, we can write:

Discrete Fourier Transform

If we separate the previous equation for k = 2r and

k = 2r + 1, we get:

Where:

and

Discrete Fourier Transform

Since:

we have:

We see that the resulting transform is in the

form of two transforms with N/2 terms.

Discrete Fourier Transform

Thus, one discrete Fourier transform with N

terms is decomposed into two discrete Fourier

transform with N/2 terms.

We have concluded that for the calculation of

the DFT with N elements, by definition, we need

approximately N2 operations. For two DFTs of N/2

elements we need 2(N/2)

2

= N

2

/2 calculations.

Discrete Fourier Transform

This procedure can be continued in n steps, we

have the elements as a simple multiplication and

summation.

Consider an example with N = 2

3

.

Illustration

Finally, we can conclude that:

N log

2

N

is number of necessary multiplications and

summations, as well.

Discrete Fourier Transform

It is interesting to find ratio N

2

/(N log

2

N) which

illustrates efficiency of the FFT algorithm in

comparison with calculation of the discrete Fourier

transform by definition.

For example, for N = 512, if we need 1 minute

to compute the discrete Fourier transform on a

computer, by using the FFT it will be calculated for

only 1 second.

Discrete Fourier Transform

Examples

1. Find the discrete Fourier transform of the

following sequences:

a) x(0) = −1, x(1) = 1, x(2) = −1

b) x(n) = a

n

(u(n) − u(n − N))

Solutions:

a) Taking N = 3 we can write:

Discrete Fourier Transform

b) Taking period N we have:

Taking for example a =1, follows:

Discrete Fourier Transform

2. Find relationship between X

p

(k) and X

p

(N − k) in

the case of real sequences x

p

(n).

Solution

Since:

We can conclude that in the case of real

sequence x

p

(n) we have:

Discrete Fourier Transform

3. If g(n) and f(n) are real sequences, show that

their discrete Fourier transforms (G(k) and

F(k)) can be obtained from the discrete Fourier

transform Y(k) of the sequence y(n) = g(n)+ jf(n).

Solution

From the signal y(n) = g(n) + jf(n) we can write:

The discrete Fourier transform of y(n) is:

Discrete Fourier Transform

Conjugate complex value of the previous

equation gives:

From the above equation the discrete Fourier

transform of the signal y

∗

(n) follows:

Discrete Fourier Transform

4. The relationship between k − th number in the

discrete Fourier transform and analog value of the

frequency is given by:

If we want to avoid shifting of the discrete Fourier

transform for k = N/2 − 1 we can multiply input

signal x(n) by (−1).

Proof:

The discrete Fourier transform of the (−1)x(n)

is:

Discrete Fourier Transform

for k ≤ N/2 − 1 we have:

Thus we have:

For the case k > N/2 − 1 follows:

Thus we have

Illustration additionally can show results of this

transformation.

Z

Z

–

–

Transform

Transform

The z-transform can be understood as a

generalization of the Fourier transform.

Applications of this transform are mainly for

description and realization of systems.

Definition of the z-transform

Z-transform of the signal x(n) is defined as:

where z is complex.

Z

Z

–

–

Transform

Transform

X(z) is defined for z where previous sum converges.

The region of convergence of the z-transform is

defined by two annular ring with r

1

and r

2

which

contain the poles of the function X(z).

The values r

1

and r

2

depend from the behavior

of the signal x(n) in the cases when n tends plus

infinity and minus infinity.

Z

Z

–

–

Transform

Transform

Example 1

Find the z-transform of the signal x(n)=u(n)

Solution:

According to definition we have:

We know that previous sum converge for

|z

−1

| <1 i.e. |z| > 1.

Z

Z

–

–

Transform

Transform

Thus the region of convergence is exterior (to the

pole location z = 1) of the unit circle |z| = 1. The

poles are denoted by ”x ”, while the zeros by ”o ”.

Z

Z

–

–

Transform

Transform

Example 2

Find the z-transform of the signal x(n) = −u(−n − 1).

Solution:

According to definition we have:

where the region of convergence is defined by

|z| < 1.

Z

Z

–

–

Transform

Transform

From the previous two examples we can conclude

that either have the same X(z).

Thus we can conclude that by using z-transform a

signal is not uniquely determined. However if we

have also the region of convergence uniquely will

be satisfied.

Consider now, four important sequence and find

their z-transform.

1. Causal series x(n) = 0 for n < 0

The z-transform of this signal is:

Z

Z

–

–

Transform

Transform

We see that z → ∞ belong to the region of

convergence.

Thus we can conclude that region of convergence

will be annular ring exterior to the pole location

with the longest distance R from origin, so we

have: R < |z| < ∞.

Z

Z

–

–

Transform

Transform

2. Non causal series x(n) = 0 for n > 0.

We se that sum converge for z = 0. The region

of convergence is the disk centered at the origin

and interior to the pole location R. Where R is the

pole the nearest to the origin, 0 ≤ |z| < R.

Z

Z

–

–

Transform

Transform

3. Sum of the causal and anticausal series

For this case we have:

From the previous considerations we have

concluded that first series converge for:

0 ≤ |z| < R1

and second one for:

R2 < |z| < ∞

Z

Z

–

–

Transform

Transform

The resultant region of convergence is:

R2 < |z| < R1

This is the annular ring. If R2 > R1 than the region

of convergence is ⊘.

Z

Z

–

–

Transform

Transform

Example

Find the z-transform and the region of

convergence for the series:

X(n) = a

n

u(n) − b

n

u(−n − 1).

Solution

The first sum converge for |z| >a, while the

second one for |z| < b. thus the region of

convergence is:

a < |z| < b

Z

Z

–

–

Transform

Transform

4. Finite length sequences x(n) = 0 for n ≤ n1 and

n ≥ n2

We conclude that sum converge for any z except 0

and/or ∞ what depends from conditions are

n

1

and n

2

positive or negative numbers.

Z

Z

–

–

Transform

Transform

Inverse z-transform

The inverse z-transform is defined by:

If we multiply right and left side of the previous

equations by: z

k−1

and if we perform integration

along restricted closed path C which resides

within the region of convergence, we obtain:

Z

Z

–

–

Transform

Transform

Since integral on the right side of the equation is

different from zero only for k = n, we

have:

This is the general form for determination of the

inverse z-transform. The previous integral can

be calculated by using the theorem of residuum:

The residuum of the function F(z) in the pole

z = z

0

, that is pole of order k, can be calculated

with:

Z

Z

–

–

Transform

Transform

The inverse z-transform will be calculated on the

base of expansion of X(z) in the series with

respect z−1. In that case we write X(z) in the form:

Than by comparing the previous equation and

definition of the inverse z-transform we see that

X(n) = X

n

.

Z

Z

–

–

Transform

Transform

Example 1.

Find the inverse z-transform for:

Solution:

Expanding the previous equations into series for

|z| > 1/4 we have:

Thus we can conclude:

X(n) = (1/4)

n

u(n)

Z

Z

–

–

Transform

Transform

In the case when the region of convergence is

|z| < 1/4 coefficients of series must be less

than 1, so X(z) has to be transformed in the form:

Thus,

Z

Z

–

–

Transform

Transform

Example 2.

Find x(n) if

Solution

Consider first:

thus we have:

x(n) = a

n

u(n)

Z

Z

–

–

Transform

Transform

If we find differential of X

1

(z), we obtain:

Now we have:

x(n) = a

n-1

u(n)

Z

Z

–

–

Transform

Transform

Table of the z-transform

Z

Z

–

–

Transform

Transform

Properties of the z-transform

Derivations of the properties of the z-transform are

analogy with the properties of the Fourier

transform.

1. Linearity

If we have y(n) = ax(n) + bh(n) than

Y(z) = aX(z) + bY(z).

2. Shifting in the time domain

For the signal x(n − n

0

), we have:

Z

Z

–

–

Transform

Transform

Example

Consider the difference equation

x(n − 1) − 2x(n − 2) = y(n)+ y(n + 1)

and represent its in z domain.

Solution:

Z

Z

–

–

Transform

Transform

3. Multiplication by complex exponential sequence

4. Convolution

If we have y(n) = x(n) ∗ h(n) than follows:

Z

Z

–

–

Transform

Transform

Example

By using z-transform find convolution of the

signals:

x(n) = u(n) and h(n) = (1/3)

n

u(n)

Solution

By using property 3, we obtain:

Z

Z

–

–

Transform

Transform

Now, we have:

The region of convergence is |z| >1. y(n) will

be obtain by using inverse z-transform. First we

will write previous equation in the following form:

where B = 3/2 and C = −1/2.

Thus we can write:

Z

Z

–

–

Transform

Transform

Relationship between z-transform, Fourier

transform and discrete Fourier transform

If we compare definition of the Fourier transform

of discrete signals and z-transform definition we

see that Fourier transform is equal to the z-

transform for |z| = 1. Thus the values of the z-

transform on the |z| = 1 in z domain are the values

of the Fourier transform of the sequence.

By expressing the complex variable z in polar form

as z = re

j ω

, we obtain:

Z

Z

–

–

Transform

Transform

taking r = 1 follows:

In general case the z-transform on the circuits

defined by r is equal to the Fourier transform of

the sequence x(n) multiplied by r−n.This is

reason why the z-transform exist in the some cases

when the Fourier transform does not exist.

Z

Z

–

–

Transform

Transform

One example that confirm the previous

statement is x(n) = u(n).

The Fourier transform of this sequence does not

converge, but the z-transform converge for r > 1.

From the previous considerations we know that

the values of discrete Fourier transform are the

samples of the Fourier transform of discrete

signals.

This means that the values of

discrete Fourier transform are equal to the samples

of the z-transform for |z| = 1.

Z

Z

–

–

Transform

Transform

Example

Find the z-transform of the sequence:

x(n) = u(n) − u(n − 4)

and in the case N = 8 find the Fourier transform

and the discrete Fourier transform by using result

obtained for z-transform.

Solution:

Z

Z

–

–

Transform

Transform

System function

Consider a system where is:

y(n) = x(n) ∗ h(n)

Having in mind properties of the z-transform

follows:

Y(z) = X(z)H(z)

Z

Z

–

–

Transform

Transform

The z-transform of the impulse response is referred

to as the system function.

The system function evaluated on the unit circle

(|z| = 1) is the frequency impulse response of the

system.

From the previous considerations we know that

stable system must satisfied condition:

Z

Z

–

–

Transform

Transform

Consider now the z-transform of the h(n):

From the previous equation follows that in the

case of stabile systems unit circle |z| = 1

must belong to the region of convergence of the

function H(z).

For causal system the region convergence must be

exterior of a circle passing through the pole of

H(z) that is farthest from the origin.

Z

Z

–

–

Transform

Transform

Example

Check the causality of the system:

Solution

We see that the region of convergence is |z| > 1/2.

Thus the system is causal.

Z

Z

–

–

Transform

Transform

Consider now the system described by a linear

Constant – coefficient difference equation, i.e.

the system that satisfy the general N − th order

difference equation:

Applying the z-transform to each side of

previous equation, we have:

Z

Z

–

–

Transform

Transform

where the property of the z-transform:

is used.

Now, we can write:

In the case where A

j

= 0 for j > 0, the system

with finite impulse response is obtained (FIR) and

in that case we have:

Z

Z

–

–

Transform

Transform

Example

Find the impulse response of the causal system

described by:

and check its stability.

Solution:

Z

Z

–

–

Transform

Transform

Poles of this function are: z

−1

= 1 and z

−1

= 1/4.

Since the system is causal the region of

convergence is |z| > 4. This means that the

system is not stabile (unit circle does not belong to

the region of convergence).

In order to determinate h(n), write H(z) in the form:

Z

Z

–

–

Transform

Transform

Examples:

1. Find the z-transform of the sequence

x(n) =δ(n − 5).

Solution:

2. If X(z) is the z-transform of x(n), find the z-

transform of:

Z

Z

–

–

Transform

Transform

By substitution n + k = m, we have:

Y(z) = X(z)X(1/z)

3. Find the impulse response of the system with z-

transform:

Solution:

Having in mind expansion in the series:

we can write:

Z

Z

–

–

Transform

Transform

Thus,

4. Find the causal sequence x(n), if its z-transform

has the form:

Solution:

Write X(z) in the form:

Z

Z

–

–

Transform

Transform

Thus we have:

x(n) =[ 1.25 − 0.25(0.2)

n

]u(n)

Z

Z

–

–

Transform

Transform

5. For the system shown in Figure, find system

function, check stability, and determine response

on the signal

x(n) = δ(n) − 2 δ(n − 1).

Solution:

From the Figure we have:

Z

Z

–

–

Transform

Transform

The system function is:

The pole of this system is z = 2, this fact means if

the system is causal it is not stabile.

If x(n) = δ(n) − 2 δ(n − 1) than:

ESTIMATION THEORY

Introduction to random signals

At the beginning, we will give some important

definitions:

Mean of the process is defined as:

The operator of mean value E is linear, i.e.:

If the random variables are independent or

uncorrelated then:

ESTIMATION THEORY

A sufficient condition for independence is:

In this case the random variables are statistically

independent.

Mean square value of x(n) is:

ESTIMATION THEORY

Correlations and covariances

The autocorrelation is defined as:

or,

where ∗ denotes complex conjugation.

The cross-correlation of two random processes x(n)

and y(n) is defined as:

ESTIMATION THEORY

The autocovariance is defined as:

If n = m the variance is obtained:

In the case of stationary process, the variance is

independent of time and denoted as

In the case of random processes that are

stationary in the wide sense we have:

ESTIMATION THEORY

In this case autocorrelation depends only on the

time difference m − n, thus:

Also we have:

ESTIMATION THEORY

White noise

The signal that has the autocorrelation in the form:

is called white noise.

The name comes from the fact that the Fourier

transform of this is constant

It means that power density spectrum of this

function is constant, what is the property of the

white light.

In the case of real noise w we have:

ESTIMATION THEORY

Power density spectrum

Consider the z-transform of the autocorrelation

r

zz

(n) (in the case of stationary signals):

Define now S

xx

(ω) as values of the z-transform

on the unit circle:

Note that S

xx

(ω) is a real-valued function, since

ESTIMATION THEORY

From the above equation we

can write:

Having in mind the definition of r

xx

(n), we have:

Thus, the expected signal power is equal to the

integral of S

xx

(ω) . This is the reason why S

xx

(ω) is

called power spectral density. Later, it will be

shown that the signal energy within the frequency

region [ω

1

, ω

2

] is equal to the integral of S

xx

(ω)

from ω

1

to ω

2.

ESTIMATION THEORY

Linear systems and random signals

For a linear system we know that:

If the signal x(n) is stationary, i.e. E{x(n − k)}= MI

x

,

we can write:

ESTIMATION THEORY

The auto-correlation of the output signal is

defined as:

In the case of stationary signal when

r

xx

(n − i,m − k)= r

xx

(n − m k − i), we have:

We can conclude,if the signal x(n) is stationary

in the wide sense, then the signal at the output of

the linear system is stationary in the wide sense,as

well.

ESTIMATION THEORY

Find, now, the z-transform of the r

yy

(n,m).

We have:

By substitution l=p − k + i, we obtain:

If h(n) is real:

ESTIMATION THEORY

Power spectral density is:

Therefore if |H(e

j

)|2 is an ideal band-pass

filter for the interval [ω

1

, ω

2

] then the expected

power of the output signal is:

since S

xx

(ω

1

) could be considered as a constant

within [ω

1

, ω

2

] for small ω

2

− ω

1

. This proves

that S

xx

(ω

1

) is the spectral power density.

ESTIMATION THEORY

Optimal filtering

Consider the signal in the form:

x(n) = s(n) + w(n)

where s(n) is desired signal and w(n) is the noise.

If we assume that the signal and noise are

uncorrelated we can write:

ESTIMATION THEORY

For the case when the power spectral density of

the signal and noise are not overlapped, we

can easily obtain denoised signal..

Namely, passing the signal through the

bandpass Filter which passes only the frequency

components of S

ss

(ω), denoised signal is obtained.

However, if the noise is white (existing in the

whole frequency range) by using previous method

it is possible to obtain only partial denoised signal.

In the general case the problem is in determination

of d(n) = s(n + m) in the most accurate

way.

ESTIMATION THEORY

If m = 0 we have the case of optimal filtering.

In some cases, it is necessary to predict the values

of the signal in the future, then m > 0.

However, in some cases we need to determine

some previous value of the signal. In this case we

have optimal smoothing, and m < 0.

Processing by using IIR system

Consider an IIR system defined by:

ESTIMATION THEORY

The mean square error is:

From this we have:

Define, now, correlation functions:

From the above equations we have:

If the signal and noise are uncorrelated, we have:

ESTIMATION THEORY

Fourier domain form of the optimal filter, when

d(n) ≡ s(n), is:

ESTIMATION THEORY

Power spectrum estimation

The mean value of the n − th sample of the

sequence x(n) can be estimated by:

where x

i

(n) is the n − th sample in the i − th

measurement.

The special class of the random processes are

ergodic processes. In this case probability

averages are equal to time average, i.e.

ESTIMATION THEORY

The process is ergodic if we can estimate its

statistically properties on the base of only one

random signal.

In previous equation μ

x

is random value

because we have finite number of samples 2N + 1.

In the case N → ∞ the value of μ

x

will be

sufficiently accurate.

Estimate, now, autocorrelation function (which

is the mean value of the product

x(n + m)x

∗

(n)):

ESTIMATION THEORY

In the case of stationary process we have:

If we know x(n) only within the interval

−N ≤ n ≤ N, then x(n + m) will be known only for

n ≤ N − m for positive number m, and

n ≥ −N |m| for negative number m.

If we want to avoid calculations for positive and

negative value of m, we will introduce symmetric

product:

ESTIMATION THEORY

In this sum we have 2N + 1 − |m| terms, but we

average it with 2N + 1. This is the reason why

we have systematic error that can be avoided by:

Thus is the biased estimate of the

autocorrelation

ESTIMATION THEORY

Definition and variance of the Periodogram

Define the Fourier transform of the biased

autocorrelation function:

Since

it can be shown that:

ESTIMATION THEORY

The spectrum estimate I

N

(ω) is called the

periodogram. The expected value of the

periodogramis:

Taking

ESTIMATION THEORY

Thus, the periodogramis a biased estimate of

the power spectrum S

xx

(ω). The previous

equation can be written, using convolution, in the

form:

where W

B

is the Fourier transform of the so called

Bartlett window ( for

|m| < 2N − 1) given by:

ESTIMATION THEORY

Variance of the Periodogram

Express the periodogramin the form:

The covariance at frequencies ω

1

and ω

2

of I

N

(ω) is:

ESTIMATION THEORY

In the case of white Gaussian process we have:

Thus,

ESTIMATION THEORY

Therefore, we have:

The variance is:

ESTIMATION THEORY

Smoothed spectrum estimators

If the sequence x(n), 0 ≤ n ≤ N − 1, is divided

into K segments of M samples, the periodogram

will be:

If we assume that periodograms are independent

of one another, we have

ESTIMATION THEORY

By assumption that K periodograms are

statistically independent, then B

xx

(ω) is the

mean of the set of K independent observations of

the periodogramI

M

(ω) :

From previous equation it is clear that as K

becomes large, the variance approaches zero, so

this smoothed estimate is a consistent estimate.

ESTIMATION THEORY

EFFECTS OF FINITE REGISTER LENGTH

Signal values in digital signal processing are

stored in a binary format, using registers with

a finite length. This can cause the error.

Namely, if we have number with b bits

multiplied by another one with b bits, the result

will be data with 2b bits. If the length of register is

less than2b we will have truncation error. This

error is:

where Q[x] and x are numbers after and before the

truncation.

ESTIMATION THEORY

If we consider, now, effects of quantizations of

analog signal, we know:

Every samples must be represent by finite

length number, so we will have truncation or

rounding to the nearest quantization level and it

will cause quantization error. This error can be

expressed by noise e(n), than we have:

where x(n) is exact value and e(n) quantization

error.

ESTIMATION THEORY

In the case of rounding the errors is in the range:

−Δ/2 ≤ e(n) ≤ Δ/2

while in the case of truncation it is:

−Δ ≤ e(n) ≤ 0,

where Δ is quantization width Δ = 2

−b

.

If we want to give a model to describe the effects

of quantization we will assume:

1. The sequences of error samples {e(n)} is a

sample sequence of stationary random process.

2. The error sequence is uncorrelated with the

sequence of exact samples {x(n)} .

ESTIMATION THEORY

3. The error is a white-noise process.

4. The probability distribution of the error process

is uniform over the range of quantization

error.

Find signal to noise ratio in the case of rounding.

According assumption 4, we have that probability

distribution p

en

(e) = 1/Δ.

Thus:

ESTIMATION THEORY

Now we have:

Multidimensional discrete signals and systems

Discrete N-dimensional signal can be defined as:

where n

1

, n

2

,.....,n

N

are integers.

Multidimensional discrete signals and systems

By analogy with the one-dimensional case we can

define:

1. Unite impulse:

2. Unite step:

Multidimensional discrete signals and systems

3. Complex exponential series

Discrete multidimensional system can be defined

by:

with x(n) and y(n) are defined input and output

signal, respectively.

Multidimensional discrete signals and systems

System is linear if:

If we denote multidimensional unite impulse

response with:

Where . The previous equation is

N-dimensional convolution denoted by:

Multidimensional discrete signals and systems

Causality and stability are defined in full analogy

with the one-dimensional case.

Fourier transform of N-dimensional discrete signals

The Fourier transform of an N-dimensional

discrete signal is defined by:

The inverse Fourier transform is given by:

Multidimensional discrete signals and systems

In the case of two-dimensional signal we have:

Example:

Find the Fourier transform of the signal:

Multidimensional discrete signals and systems

Solution:

Also, by analogy with the one dimensional

sampling theorem, it is easy to show that:

Multidimensional discrete signals and systems

where it has been assumed:

Multidimensional discrete signals and systems

Multidimensional discrete Fourier transform and

FFT algorithms

Consider two-dimensional discrete Fourier

transform: the simplest 2D FFT algorithm are

based on the FFT algorithm for one-dimensional

case. Namely:

We see that for a fixed value n1, the second

sum presents one-dimensional discrete Fourier

transform which can be calculated by using some

of the FFT algorithms.

Multidimensional discrete signals and systems

Thus:

This procedure should be repeated for all n1.

Two-dimensional discrete Fourier transform

will be obtained as:

Calculations should be performed for all k

2

.

Multidimensional discrete signals and systems

Ratio of number of additions and summations

for discrete Fourier transform by using

definition and FFT algorithm is given by:

In the case of M = 128 this ratio is 1170 ( if need

one second with the FFT than 19,5 minutes would

be needed by using calculation based on the

definition).

Multidimensional discrete signals and systems

Radon transform and computer’s Tomography

Integral along line AB is:

where AB is defined by:

The previous integral can be written in the form:

Multidimensional discrete signals and systems

The previous integral can be written in the form:

Previous integral defines projection of function

f(x, y) with respect to variable t for an arbitrary

angle θ.

Is it possible to reconstruct function f(x, y) on

the base projections?

Answer is yes.

Multidimensional discrete signals and systems

Proof:

Consider the Fourier transform F(u, v) of the

function f(x, y):

Multidimensional discrete signals and systems

The Fourier transform of a projection is:

Consider as a special case the value of the F(u, v),

along the line v = 0, then we have:

Multidimensional discrete signals and systems

Thus, we have obtained that the Fourier

transform of the function f(x, y) along axis v = 0

is equal to the Fourier transform of projection for

the angle θ = 0.

This result can be generalized. It can be shown

that the Fourier transform of f(x, y) along an

arbitrary line defined by angle θ with respect to u

axis is equal to the Fourier transform of the

projection defined by angle θ with respect to x

axis.

Multidimensional discrete signals and systems

The previous claim can be proved. Denote with

f(s, t) the function f(x, y) rotated in the coordinate

system. Relationship between variables

(x, y) and (s, t) is:

Since:

The Fourier transform of the projection is:

Multidimensional discrete signals and systems

In x, y coordinate system we obtain:

Thus, the previous claim is proved.

Multidimensional discrete signals and systems

Finally, we can conclude:

Function f(x, y) can be obtained on the following

way:

1. Find the projection for 0 ≤ θ ≤ π.

2. Determine the Fourier transforms of the

projections which give the Fourier transform of

the function f(x, y).

3. Compute the inverse Fourier transform and it if

function f(x, y).

Multidimensional discrete signals and systems

Note that the Fourier transform of function

f(x,y) will be obtained in polar raster. If we

want to use FFT algorithms it is necessary to have

the Fourier transform in rectangular raster.

One possible solution is interpolation of values

from the polar to the values on the rectangular

raster.

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