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**DIGITAL SIGNAL PROCESSING LAB 1
**

MATLAB: An Introduction

1. MATLAB Commands

To do work in MATLAB, you type commands at the >> prompt. Often these

commands will look like standard arithmetic. Or function calls similar to many other

computer languages. By doing this, you can assign sequences to variables and then

manipulate them many ways. You can even write your own functions and programs

using MATLAB's control structures. The following sections will describe the most

commonly used commands on MATLAB and give simple examples using them.

To make life easier, MATLAB includes a simple line-editing facility. If you make an

error while typing a command, you don't have to retype the whole command. You

can just call up the last line you typed and fix the error. To recall the last line you

typed, hit the up arrow. MATLAB actually saves several of your last commands, so

you can continue moving backwards through your prior commands by hitting the up

arrow more than once. If you happen to move too many commands backwards, you

can hit the down arrow to move to the next command (i.e. the one you just went past

by hitting the up arrow). The left and right arrows move you left and right one

character, respectively. Similarly, Ctrl-L and Ctrl-R will move you left and right one

word. Ctrl-B will move you to the beginning of the line, and Ctrl-E will move you to the

end of the line. Ctrl-A will toggle you between insert and overwrite mode for line-

editing. Finally, as you might well expect, backspace will delete the character to the

left of the cursor.

1.1. Basic Terminology

MATLAB has some basic terms and predefined variables you should get familiar with

before going any further. One of these is the variable ans. When you type most

commands to MATLAB, they will return values. You can assign these values to

variables by typing in equations. For example, if you type

>>x=5

MATLAB will print

x =

5

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DIGITAL SIGNAL PROCESSING LAB 2

and assign the number five to the variable x. MATLAB uses ans for any expression

you don't assign to a variable. For instance, if you type

>> 5

to MATLAB, MATLAB will return

ans =

5

and assign the value 5 to the variable ans. Thus, ans will always be assigned to the

most recently calculated value you didn't assign to anything else.

MATLAB creates a variable called eps every time you run it. eps is the distance from

1.0 to the next largest floating point number. Basically, it is the maximum resolution

of the system for floating point arithmetic. On the 486 PC workstations, eps =

2.2204e-016. MATLAB also keeps a permanent variable named pi, which is defined

as 3.1415.... The permanent variable Inf represents the IEEE positive infinity. This

should be returned by evaluating an expression such as 1.0/0.0. Similarly, the

permanent variable NaN represents a value which is not a number, such as the result

of evaluating 0.0/0.0.

As the earlier examples showed, whenever you type a command to MATLAB,

MATLAB returns the value returned, and the variable to which that value was

assigned. Sometimes, you don't want to see this. For example, if you assign b to be

the integers from I to 1000, you probably don't want to wait for MATLAB to print them

all out. If you terminate a command with a semi-colon, MATLAB will suppress the

printing of the variable name and value resulting from the calculation. For example, if

you type

>> x = 5;

MATLAB will assign the value five to the variable x, but rather than tell you it did that,

it will just return another >> prompt.

MATLAB works with two basic types of data objects: scalars and matrices. MATLAB

also has vectors. Vectors are a special case of matrices which are only 1 row by any

number of columns. Vectors can be used to represent discrete-time sequences. We

showed earlier how to assign a scalar, such as five, to a variable. To assign x to be

a matrix by explicitly entering the elements, you type the list of elements separated

by blanks or commas surrounded by [ and ], and use semi-colons to separate the

rows. For example, typing

>>x=[2468; 1 357]

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DIGITAL SIGNAL PROCESSING LAB 3

results in

x = 2 4 6 8

1 3 5 7

The MATLAB workspace is defined as the collection of all the variables you have

defined during the current MATLAB session. A session is defined as beginning when

you type the command matlab to Athena, and ends when you type the quit command

to MATLAB. While MATLAB has room for a substantial amount of variables, it can

get very full if you are working with large vectors or matrices. Shortly, we will show

you how to see what variables are in your workspace, and how to save your

workspace so you can continue your session later.

MATLAB can do complex arithmetic. However, it does not explicitly create a variable

named i. You will need to do this yourself if you want to enter complex numbers. You

can do this by typing

>> i=sqrt(- 1)

MATLAB will return:

i=

O + 1.000I

You can name this variable j if your tastes run that way. Now, you can use complex

numbers of the form a + bi simply by typing a + b*i. For example, to set y = 2 + 3i,

you would type

>> y = 2+3*i

At times, you will want to deal with just a part of a vector or matrix. To do this, you

need to use MATLAB's indexing facility. To get the nth element of the vector x, you

type x(n). MATLAB starts counting from one when numbering vector elements, so the

first element of x is x(l) and not x(O). C hackers should be especially careful about

this. You can also use indices on matrices. The element in the ith row, jth column of

x is x(i,j).

1.2. Basic Arithmetic

MATLAB uses a straightforward notation for basic arithmetic on scalars. The symbol

+ is used to add scalars, so x=1+5 will give x the value 6. Similarly, MATLAB uses -

for subtraction, * for multiplication, / for division, and ^ for exponentiation. All of these

work for two scalars. In addition, you can add, subtract, multiply or divide all the

elements of a vector or matrix by a scalar. For example, if x is a matrix or vector,

then x+1 will add one to each element of x, and x/2 will divide each element of x by 2.

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DIGITAL SIGNAL PROCESSING LAB 4

x^2 will not square each element of x. We'll show you how to do that later. All of the

basic operations (+, -, *, /, and ^) are defined to work with complex scalars.

Another useful operator is the colon. You can use the colon to specify a range of

numbers. Typing

>>x= 1:4will return

x =

1 2 3 4

You can optionally give the colon a step size. For instance,

>>x=8:-1:5 will give

x =

8 7 6 5

and

>> x = 0:0.25: 1.25

will return

x =

0 0.25 0.5 0.75 1.0 1.25

The colon is a subtle and powerful operator, and we'll see more uses of it later.

1.3. Help

MATLAB has a fairly good help facility. The help function knows about all the

commands listed in this manual. Typing help function will tell you the syntax for the

function, i.e. what arguments it expects. It will also give you a short description of

what the command does. If you think you are doing something right, but MATLAB

claims you are in error, try looking at the help for the functions you are using. Later,

when we discuss writing your own functions, we will show you how to include help

info for your functions. This can be very useful for other people using your function,

or for your own use if you haven't used the function for a while.

1.4. Basic Matrix Constructors and Operators

MATLAB has a variety of built-in functions to make it easier for you to construct

vectors or matrices without having to enumerate all the elements.The ones function

will create a matrix whose elements are all ones. Typing ones(m,n) will create an m

row by n column matrix of ones. To create a discrete-time signal named y assigned

to a vector of 16 ones, you would type y = ones(1,16);. Also, giving ones a matrix as

its only argument will cause it to return a matrix of ones the same size as the

argument. This will not affect the original matrix you give as an argument,

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DIGITAL SIGNAL PROCESSING LAB 5

Though, If x = [ I 2 3 4; 0 9 3 8], typing ones(x) will create a matrix of ones that is two

rows by four columns. The zeros function is similar to the ones function. Typing

zeros (m, n) will create an m-by-n matrix of zeros, and zeros(x) will create a two-by-

four matrix of zeros, if x is defined the same way as above.

The maximum and minimum functions are used to find the largest and smallest

values in a vector. If z = [1 2 -9 3 -3 -5], max(z) will return 3. If you call max with a

matrix as an argument, it will return a row vector where each element is the

maximum value of each column of the input matrix. Max is also capable of returning

a second value: the index of the maximum value in the vector. To get this, you

assign the result of the call to max to be a two element vector instead of just a single

variable. If z is defined as above, [a b] = max(z) will assign a to be 3, the maximum

value of the vector, and b to be 4, the index of that value. The MATLAB function min

is exactly parallel to max except that it returns the smallest value, so min(z) will return

-9.

Sum and prod are two more useful functions for matrices. If z is a vector, sum(z) is

the sum of all the elements of z. Similarly, prod(z) is the product of all the elements

of z.

Often, it is useful to define a vector as a subset of a previously defined vector. This is

another use of the colon operator. If we have z defined as in the min/max examples,

z(2:5) will be the vector [2 -9 3 -3]. Again, remember that MATLAB indexes vectors

such that the first element has index one, not zero.

The MATLAB size function will return a two element vector giving the dimensions of

the matrix it was called with. If x is a 4 row by 3 column vector, size(x) will return the

vector [4 3]. You can also define the result to be two separate values as shown in

the max example. If you type [m n] = size(x), m will be assigned to the value 4, and n

to the value 3. The length operator will return the length of a vector. If z is defined

as 14 3 91, length(z) will return 3. Basically, length(z) is equivalent to max(size(z)).

1.5. Element-wise Operations

We will often want to perform an operation on each element of a vector while doing a

computation. For example, we may want to add two vectors by adding all of the

corresponding elements. The addition (+) and subtraction (- operators are defined to

work on matrices as well as scalars. For example, if x = [1 2 3] and y = [5 6 2], then

x+y will return [6 8 5]. Again, both addition and subtraction will work even if the

elements are complex numbers.

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DIGITAL SIGNAL PROCESSING LAB 6

Multiplying two matrices element by element is a little different. The * symbol is

defined as matrix multiplication when used on two matrices. To specify element-wise

multiplication, we use, *. So, using the x and y from above, x.*y will give [5 12 6]. We

can do exponentiation on a series similarly. Typing x.^2 will square each element of

x, giving [ I 4 9]. Finally, we can't use / to divide two matrices element-wise, since /

and \ are reserved for left and right matrix "division." So, we use the./function,

which means that y./x will give [5 3 0.6666]. Again, all of these operations work for

complex numbers.

The abs operator returns the magnitude of its argument. If applied to a vector, it

returns a vector of the magnitudes of the elements. For instance, if x = [2 -4 3-4*i -3

kid, typing

>> y = abs(x)

will return

y=

2 4 5 3

The angle operator will return the phase angle of its operand in radians. The angle

operator will also work element-wise across a vector. Typing

>> phase = angle(x)

will give

phase =

0 3. 14 1 6 -0.9273 - 1.5708

The sqrt function is another commonly used MATLAB function. As you might well

expect, it computes the square root of its argument. If its argument is a matrix or

vector, it computes the square root of each argument. If we define x = [4 -9 i 2-2*I],

then typing

>> y = sqrt(x)

gives

y= 2.00000 3.0000i 0.7071 i 1.5538 -0.6436i

MATLAB also has operators for taking the real part, imaginary part, or complex

conjugate of a complex number. These functions are real, imag and conj,

respectively. They are defined to work element-wise on any matrix or vector.

MATLAB includes several operators to round fractional numbers to integers. The

round function rounds its argument to the nearest integer. The fix function rounds its

argument to the nearest integer towards zero, e.g. rounds "down" for positive

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 7

numbers, and "up" for negative numbers. ceil rounds its argument to the nearest

integer towards positive infinity, e.g. "up", and floor rounds its argument to the

nearest integer towards negative infinity, e.g. "down." All of these commands are

defined to work element-wise on matrices and vectors. If you apply one of them to a

complex number, it will round both the real and imaginary part in the manner

indicated. Typing

>> ceil(3.1 +2.4* i)

will return

ans = 4.0000 + 3.0000i

MATLAB can also calculate the remainder of an integer division operation. If x = y *

n + r, where n is an integer, then rem(x,y) is r.

The standard trigonometric operations are all defined as element-wise operators.

The operators sin, cos and tan calculate the sine, cosine and tangent of their

arguments. The arguments to these functions are angles in radians. Note that the

functions are also defined on complex arguments, which can cause problems if you

are not careful. For instance, cos(x+iy) = cos(x)cosh(y) - i sin(x)sinh(y). The inverse

trig functions (acos, asin and atan) are also defined to operate element-wise across

matrices. Again, these are defined on complex numbers, which can lead to problems

for the incautious user. The arctangent is defined to return angles between pi/2 and -

pi/2.

In addition to the primary interval arctangent discussed above, MATLAB has a full

four quadrant arctangent operator, atan2. atan2(y,x) will return the angle between -pi

and pi whose tangent is the real part of y/x. If x and y are vectors, atan2(y,x) will

divide y by x element-wise, then return a vector where each element is the four-

quadrant arctangent of corresponding element of the y/x vector.

MATLAB also includes functions for exponentials and logarithms. The exp operator

computes e to the power of its argument. This works element-wise, and on complex

numbers. So, to generate the complex exponential with a frequency of pi/4, we could

type

>> n = 0:7;

>> s = exp(i*(pi/4)*n)

s = Columns 1 through 4

1.0000 0.7071 + 0.7071 I 0.0000 + 1.0000i -0.7071 + 0.7071 i

Columns 5 through 8

-1.0000 + 0.0000i -0.7071 i - 0.7071 i 0.0000 - 1.0000i

0.7071 -0.7071 i

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 8

MATLAB also has natural and base-10 logarithms. The log function calculates

natural logs, and log l0 calculates base-10 logs. Again, both operate element-wise

for vectors. Both also come complete with the now-familiar caveat that they are

defined for complex values, and that you should be careful about passing them

complex arguments if you don't know how complex logs are defined.

1.6. Graphics

MATLAB has several commands to allow you to display results of your computations

graphically. The plot command is the simplest way of doing this. If x is a vector,

plot(x) will plot the elements of x against their indices. The adjacent values of x will

be connected by lines. For example, to plot the discrete-time sequence that is a

sinusoid of frequency pi/6, you would type:

>>n=0:11;

>> y = sin((pi/6)*n);

>> plot(y)

Shortly after you enter the plot command, MATLAB will prompt you for a location to

open the graphics window. Just move the mouse to put the window where you want

it, then click the left button. After MATLAB plots the graph, it will wait for you to hit a

key while the mouse is in either the MATLAB text window or graphics window before

it will continue with the MATLAB >> prompt. If you run the commands above, you will

notice that the first value graphed has an abscissa value of one, and not zero. This is

because MATLAB indexes vector elements beginning with one, not zero. Plot will

use the values of y for the y-axis, and their indices for the x-axis. To obtain a graph

of y versus n, you would type

>> plot (n, y)

If plot gets two vectors for arguments, it creates a graph with the first argument as

the abscissa values and the second vector as ordinate values. You can also change

the type of line used to connect the points by including a third argument specifying

line type. The format for this is plot(x, y,'line-type') where line-type is - for a solid line,

- for a dashed line, for a dotted line, and -. For a line of alternating dots and dashes

whichever character you chose to use must be enclosed by single quotes. For

instance, plot (n, y,':') would create the same graph as above, except that the points

would be connected by a dotted line. The default line type is solid. Thinking carefully,

we see that in this case, it is misleading to connect the adjacent values by lines,

since we are graphing a discrete-time sequence. Instead, we should just put a mark

to indicate each sample value. We can do this by using a different set of characters

in place of the line-type argument.

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DIGITAL SIGNAL PROCESSING LAB 9

If we use a period, each sample is marked by a point. Using a + marks each sample

with a + sign, * uses stars, o uses circles, and x uses x's. So, typing

>> plot (n, y,'o')

will plot the values of y against their indices, marking each sample with a circle. We

can also plot several graphs on the same axis. Typing

>> Plot(x 1, y 1, x2, y2)

will graph y I vs. x 1 and y2 vs. x2 on the same axis. You can also include a specific

line or point type for each graph if you want by typing

>> Plot(x me, y 1, ‘line-type 1’, x 2, y2, ‘1ine-type2’

where the line-types can be any of the characters listed above to specify line or point

types.

You can also create plots with either or both axes changed to log-scale. All of these

functions follow the same conventions for arguments and line or point types as plot.

Using log log will create a plot with both axes as log scales. For a plot with only one

axis on log scale, semi logy will create a plot where the x-axis is linear and the y-axis

is logarithmic; while semilogy will have a linear y-axis and logarithmic x-axis.

You can use additional MATLAB commands to title your graphs, or put text labels on

your axes. Typing title ('Beware of the aardvark') will label the current graph at the

top with the text enclosed in single quotes. In this case, that is "Beware of the

aardvark." Likewise, you can label your x- and y-axes by typing label ('This is the x-

axis') and label ('This is the y-axis') respectively .

The axis command is used to control manually the limits and scaling of the current

graph. Typing

>> a = axis

will assign a four-element vector to a. The first element is the minimum x-value; the

second is the maximum x-value for the current graph. The third and fourth elements

are the minimum and maximum y-values, respectively. You can set the values of the

axes by calling the axis function with a four-element vector for an argument. These

elements should be your choices for the x- and y-axis limits, in the same order as

specified above.

So, if you type

>> axis ([-10 10 -5 5])

you will rescale the axis in the graphics window so the x-axis goes from -10 to 10,

and the y-axis from -5 to 5. The axis command can be stubborn sometimes, and

round your limits up to new limits it finds easier to draw.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 10

There's really nothing you can do about it. The hold command will keep the current

plot and axes even if you plot another graph. The new graph will just be put on the

current axes as much as it fits. Typing hold a second time will toggle the hold off

again, so the screen will clear and rescale for the next graph.

You can use the subplot command to split the screen into multiple windows, and then

select one of the sub-windows as active for the next graph. The format of the

command is subplot (xyn) where x is the number of vertical divisions, y is the number

of horizontal divisions, and n is the window to select for the first plot. Both x and y

must be less than or equal to two, and n must be less than or equal to x times y. For

example, subplot (121) will create two full-heights, half-width windows for graphs,

and select the first, e.g. left, window as active for the first graph. After that, unless

you specifically indicate which window is active, MATLAB will cycle through them

with each successive plot. Typing subplot with no arguments will return the graphics

window to its original, single-window state.

You can use the print command to get a hard copy of the current graphics window.

Typing print ('printer-name') will send the current graph to the printer whose name

appears in single-quotes as the argument.

This may take a minute or two to start printing. MATLAB will just sit idle while

generating the graphics file to dump to the printer. If no printer is specified, MATLAB

will send the printout to the default printer for your workstation. Note that MATLAB

can only send printouts to PostScript printers. Also, the implementation of print with

the printer name as an argument is Athena specific, and may not be found in another

implementation of MATLAB.

1.7. Logical Operations

MATLAB allows you to perform boolean operations on vectors element-wise. For the

purpose of boolean algebra, MATLAB regards anything with a non-zero real part as

true, and everything else as false. MATLAB uses & for the boolean and operator, 1

for or, and for not. So, typing

>> [1 0 2 4] & [O 0 1 i]

gives

[

ans =

0 0 1 0

while typing

>> [1 0 2 4] 1 [0 0 1 i]

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 11

will return

ans =

1 0 1 1

In addition, you can run a cumulative boolean or boolean and across all the elements

of a matrix or vector. If v is a vector or matrix, any (v) will return true if the real part of

any element of v is non-zero. Similarly, all (v) will return true if all the elements of v

have non-zero real parts.

You can also compare two vectors element-wise with any of six basic relational

operators, e.g. less-than (<), greater-than (>), equal-to (==), not-equal-to (~=), less-

than or-equal-to (<=), and greater-than-or-equal-to (>=). Typing

>> [1 2345] <= [543 2 11

returns

ans =

1 1 1 0 0

We will see more uses of the relational operators when we discuss MATLAB

programming control structures.

1.8 Interface Controls

When you first start running MATLAB, it is case-sensitive. This means that MATLAB

distinguishes between upper and lower case variable and function names. So, you

can safely have two separate variables named aardvark, and AARDVARK. If you

type casesen, MATLAB will toggle to being non-case-sensitive. Typing it again will

toggle back to the original mode, e.g. case-sensitive.

MATLAB allows you to clear either the command (text) window, or the graphics

window. The clc command will clear the command window, and give you a fresh >>

prompt. The clg command will clear the graphics window and leave it blank.

You can see all the variables defined in your workspace by using the who command.

The whos command will give you the names of all the variables just like who does.

In addition, it will tell you the size of each variable, if that variable is complex, and the

total bytes of space used by all your variables. If you run the who command, and see

that you have variables you are no longer using, you can use the clear command to

remove obsolete variables. For example, typing clear godzilla would delete the

variable named godzilla.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 12

MATLAB also defines some basic file manipulation utilities, so you can do these

things without quitting and restarting MATLAB. The dir command is identical to the

UNIX ls command. The chair is the same as cd for UNIX, type is UNIX’s cat, and

delete is rm. You can also pass any other UNIX command through by preceding it by

a! So, typing! date will give you the current date and time. All pathnames and

wildcards should work the same as they do in UNIX.

MATLAB's diary command allows you to make a transcript file recording your

session. Typing diary filename will record all the commands you type in filename,

and will also record "most of the resulting output," according to the MATLAB manual.

Graphs will not be recorded, but almost all printed results will be. Typing diary off will

turn the transcript off, and diary on will turn it back on. The file created is in ASCII

format, suitable for editing with Emacs and including in other reports. In fact, this

function was used to generate almost all the examples in this manual.

MATLAB includes commands that allow you to save all or some of your workspace

so you can continue your session later. Typing save filename will save all of your

workspace in a compressed format in a file named filename. mat. If you want to

reload that workspace later to continue, you just type load filename. If you only want

to save some of the variables, you can give the save command the names of the

variables to save.

If you type

>> save aardvark moose wildebeest

MATLAB will save the variables moose and wildebeest in the file aardvark.mat.

1.9. Signal Processing Functions

MATLAB comes with several useful signal processing functions already defined. For

example, if a and b are vectors (a.k.a. discrete-time sequences), then conv (a, b) will

convolve the two vectors. Typing aft (a) will return the Discrete Fourier Transform of

a, and idft (a) is the inverse DFT a. fft (a) will return the Fast Fourier Transform of a,

and ifft (a) is the Inverse FFT. If a is not a radix-two in length, fft and ifft will zero-pad

it so that the number of samples in the sequence is a power of two.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 13

If we define a and b to represent a system described by the difference equation:

y(n) =: )( i ) *x(n) + b(2) Din- 1 ) + ...+ b(nb)*x(n-nb+l)

- a(2)*y(n-1) -…-a(na)*y(n-na+1)

typing

>>filter (b, a, x)

will return the sequence that is the result of filtering x with that the system defined by

that difference equation. The b and a coefficient vectors should be ordered by

ascending value of delay. If a (I) is not I, filter will normalize a so that it is by dividing

all of a by a (1).

If we define a and b for a system as above, we can also use the freqz command to

evaluate the Z-transform of the system. There are several ways to input values

depending on how you want to evaluate the transform. Typing

>> freqz (b, a, n)

evaluate the z-transform at n evenly-spaced points around the upper half of the unit

circle.

If you type

>> freqz (b, a, n 'whole')

MATLAB will evaluate the transform at n evenly-spaced points around the whole unit

circle. If you assign the result of freqz to a single variable, you will get back the

values at the frequencies, which may be complex. If you assign the result to a two-

element vector, the first element will be assigned to the complex values of the

samples, while the second element will be assigned to a vector of the frequencies the

z-transform was evaluated at. For example, to look at the very simple system y[n] =

x [n - 1], you would type:

>>a= [1];

>>b= [0 1];

>> [h w] = freqz (b, a, 16);

After doing this, h would be defined as the complex-valued samples of the z-

transform, and w would be

the frequencies at which the z-transform was evaluated to get those values. If you

choose n to be a power of two, freqz will run much faster, since it will use the fft

operator. If you don't want evenly-spaced samples around the unit-circle, you can

explicitly specify at which frequencies the z-transform should be evaluated.

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To do this, type freqz (b, a, w) where w is a vector of the frequencies (in radians) at

which to evaluate the z-transform.

1.10. Polynomial Operations

Vectors can also be used to represent polynomials. If you want to represent an Nth-

order polynomial, you use a length N+1 vector where the elements are the

coefficients of the polynomial arranged in descending order of exponent. So, to

define y = x2 - 5x + 6, you would type:

>> y = [1 -5 6];

The MATLAB roots function will calculate the roots of a polynomial for you. If we use

the y from above,

>> roots(y)

will return

ans =

3 2

MATLAB also has the poly function, which takes a vector and returns the polynomial

whose roots are the elements of that vector.

You can use MATLAB to multiply two polynomials using the cony function described

above. The convolution of the coefficient vectors is equivalent to multiplying the

polynomials. The polyval function can tell you the value of a polynomial at a specific

point. For example, polyval(y, 1) would return 2. polyval also works element-wise

across a vector of points, returning the vector where each element is the value of the

polynomial at the corresponding element of the input vector.

1.11. Control Structures

MATLAB includes several control structures to allow you to write programs. The for

command allows you to make a command or series of commands be executed

several times. It is functionally very similar to the for function in C. For example,

typing

for i= 1:4

end

will cause MATLAB to make the variable i count from 1 to 4, and print its value for

each step. So, you would see

I = 1

I = 2

I = 3

I = 4

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Every command must have a matching end statement to indicate which commands

should be executed several times. You can have nested for loops. For example,

typing

Form = 1:3

for n= 1:3

x(m,n)=m+n*i;

end

end

will define x to be the matrix

x =

1.0000 + 1.0000i 1.0000 + 2.0000i 1.0000 + 3.0000i

2.0000 + 1.0000i 2.0000 + 9.0000i 2.0000 + 3.0000i

3.0000 + 1.0000i 3.0000 + 2.0000i 3.0000 + 3.0000i

The indentations in the for structure are optional, but they make it easier to figure out

what the commands are doing.

The if command lets you have programs that make decisions about what commands

to execute. The basic command looks like

if a > 0

x=a^2;

end

This command will assign x to be the value of a squared, if a is positive. Again, note

that it has to have an end to indicate which commands are actually part of the if. In

addition, you can define an else clause which is executed if the condition you gave

the if is not true. We could expand our example above to be

if a>0

x = a^2;

else

x = -a^2

end

For this version, if we had already set a to be 2, then x would get the value 4, but if a

was -3, x would be -9. Note that we only need one end, which comes after all the

clauses of the if. Finally, we can expand the if to include several possible conditions.

If the first condition isn't satisfied, it looks for the next, and so on, until it either finds

an else, or finds the end. We could change our example to get

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if a>0

x = a^2;

else if a == 0

x = i;

else

x = -a^2

end

For this command, it will see if a is positive, then if a is not positive, it will check if a is

zero, finally it will do the else clause. So, if a positive, x will be a squared, if a is 0, x

will be i, and if a is negative,

then x will be the negative of a squared. Again, note we only have a single end after

all the clauses.

The third major control structure is the while command. The while command allows

you to execute a group of commands until some condition is no longer true. These

commands appear between the while and its matching end statement. For instance,

if we want to keep squaring x until it is greater than a million,

we would type

while x < 1000000

x = x^2;

end

If we start with x = 2, this will run until x is 4.295 x 109. Everything between the while

line and the end will be executed until the boolean condition on the while line is no

longer true. You have to make sure this condition will eventually stop being true, or

the command will never finish. If it is not initially true, the commands will never be

executed.The pause command will cause MATLAB to wait for a key to be pressed

before continuing. This is useful when you are writing your own functions.

Sometimes you will want to terminate a for or while loop early. You can use the

break command to jump out of a for or while command. For example, you could

rewrite our while example from above to be:

while 1

if x > 1000000

break;

end

x = x^2

end

which will have exactly the same effect.

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1.12. Functions

Finally, we come to the apex of our discussion: writing your own functions for

MATLAB. You do this using M-files. An M-file is an ASCII text file that has a

filename ending with .m, such as aardvark.m. You can create and edit them from

Emacs. There are two classes of M-files, functions and scripts. Functions are more

interesting, so we'll talk about them first. An M-file is a function if the first word in it is

function. A function can just take several arguments or none at all, and return any

number of values.

We will look at several examples grouped by the number of values they return. The

first line of the file specifies the name of the function, along with the number and

names of input arguments and output values.

3.12.1 Functions Returning No Values

A common example of a function that doesn't return any values is one that draws

graphs. It just draws the graphs, then finishes. Here is an example of such a function

function stars (t)

%STARS(T) draws stars with parameter t

n=t*50;

plot (rand (1, n), rand (1, n),',')

% that line plots n random points

title ('My God, Its Full of Stars!’);

% label the graph

This will draw a random sprinkling of dots across the screen. The first line defines

this as a function names stars that takes one argument named t, and doesn't return

anything. The next line is the help comment. Any comments coming immediately

after the first line will re returned by help stars, which would be

%STARS (T) draws stars with parameter t

for this function. A line of comments is indicated by the % character. You can have as

many of these as you want, and the help command will print all of them. Next, the

function defines an internal variable named n to be Fifty times t. This n is totally

unrelated to any variable already defined in the main MATLAB workspace. Assigning

this value will not alter any n you had already defined fore calling the stars function.

You can also see how we put a comment in the middle of the function indication

which command actually drew the stars.

Sometimes, you will need to write several versions of a function before it works

properly. When you use an M-file function for the first time during a session,

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MATLAB reads it in, compiles it, and then remembers that version, If you change the

M-file after using the function, MATLAB won't see it. It will just use the first version it

compiled. To remove the first version, and force MATLAB to read the new one in, you

should use the clear command. If you type clear orangutan, MATLAB will erase the

function orangutan from MATLAB's previously know functions, so the next time you

call orangutan it will read in your new version of orangutan. Make sure you do this

when you fix a bug in a function, or you will be very confused about why MATLAB is

ignoring your changes.

1.12.2 Functions Returning One Value

Next, we look at an example of a function that returns one value.

function y = fact (n)

%Y = FACT (N) computes the factorial of n;

y= 1;

for i = 2:n\

y=y*i;

end

This function will calculate the factorial of its argument. The first line indicates that

this is a function of one argument n, and should return the value of the variable y for

its answer. The for loop will repeatedly multiply y by every value between 2 and n.

Whatever value is finally in y will be returned as the result of the function. For this

function, it is a scalar, but functions can also return vectors or matrices. Again, note

that the variables y, i, and n have absolutely no effect on any variables of the same

name in the main MATLAB workspace.

1.12.3 Functions Returning More Than One Value

If you want to return more than one argument, you can do it by having the function

return a vector of values. The following function is an example of a function that

does this. For instance, the following function returns two vectors. The first vector is

the prime factors less than 10 of the argument. The second indicates how many

times each of these factors is used.

function [factors, times] = primefact (n)

% [FACTORS TIMES] = PRIMEFACT (N)

find prime factors of n

primes = [2 3 5 7]

for i= 1:4

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temp = n;

if (rem (temp, primes (i)) ==0)

factors = [factors primes (i)];

times = [times 0];

while (rem (temp, primes (i)) ==0)

temp = tamp/primes (i);

times (length (times)) = times (length

(times)) +1;

end

end

end

If we call this function with just one argument, i.e. a = prime fact( 10) we will get back

the vector of prime factors, but not the vector indicating how many times each factor

appears in the number. To get both vectors, we should call the function in this

manner:

>> [a b] = prime fact (180)

a=

2 3 5

b =

2 2 1

This way, we get both vectors returned: the primes in a and the number of times

each prime was used in b. From these results, we see that 180=2x2x3x3x5.

1.12.4 Script M-files

As we mentioned earlier, there is a second kind of M-file. The script type M-file is

just a list of commands to execute in sequence. This differs from a function M-file in

that no arguments are needed. Also, the variables inside a script file are the same

ones as in the main MATLAB workspace. If I have a variable named n = 1000, then

execute a script that includes the line n = 2, my variable will now be 2, and not 1000.

To create a script file, just create a file with Emacs that contains the commands you

want executed. A script file should not have the word function in the first line, and

doesn't need the comments for the help command. The filename should still end in

.m though.

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1.13. Advanced Features

The features listed in the following sections are not essential for using MATLAB on

Athena, but are included for the more curious users who want to push MATLAB as

much as possible.

1.13.1 Selective Indexing

Sometimes, you only want to perform an operation on certain elements of a vector,

such as all the elements of the vector that are less than 0.

One way to do this is a for loop that checks to see if each element is less than zero,

and if so, does the appropriate function. However, MATLAB includes another way to

do this. If you say

>>x(x<0) = -x(x<0)

MATLAB will change all the negative elements of the vector x to be positive. The

following sequence of commands illustrates this:

>>x = [-3 -2 0 2 4]

x =

-3 -2 0 2 4

>>x(x<0) = -x(x<0)

x =

3 2 0 2 4

Though this notation can be more confusing than a for loop, MATLAB is written such

that this operation executes much, much faster than the equivalent for loop.

You can also perform operations on a vector conditionally based upon the value of

the corresponding element of another vector. For instance, if you want to divide two

vectors element-wise, you have to worry about what happens if the denominator

vector includes zeros. One way to deal with this is shown below.

>>x= [32024]

x =

3 2 0 2 4

>>y= [1 1 1 1 1]

y=

1 1 1 1 1

>>q = zeros (1, length(y))

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q=

0 0 0 0 0

>>q(x~=0) = y(x~=0). / X(x~=0)

q=

0.3333 0.5000 0 0.5000 0.2500

You can perform this type of conditional indexing with any boolean operator

discussed earlier, or even with boolean operators on the results of functions on

elements of vectors.

For example

>>q ((x<=3) & (q<=0.4)) = q ((x<=3) & (q<=0.4)) + 14

q=

14.3333 0.5000 0 0.5000 0.2500

1.13.2 Functions with Variable Number of Arguments

Anytime you call a function, MATLAB defines a variable inside that function called

margin. margin is the number of arguments with which the function was called. This

allows you to write functions that behave differently when called with different

numbers of arguments. If you specify the function rhino such that the first line of its

M-file file reads:

function hippo = rhino (a, b, c, d)

MATLAB will not get upset if you call rhino with only two arguments. It will just

assume the last two arguments were optional, and not used. MATLAB will set nargin

= 2, and execute the function. If the function tries to do something using the variables

c or d, MATLAB will generate an error message. MATLAB assumes that you will use

nargin to avoid referring to any optional arguments that weren't supplied.

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MATLAB BASICS

A= [16 3 2 13

5 10 11 8

9 6 7 12

4 15 14 1];

SUM (A)

MATLAB replies with

Ans=

34 34 34 34

A’ produces

Ans=

16 5 9 4

3 10 6 15

2 11 7 14

13 8 12 1

And

Sum (A’)’ produces a column vector containing the row sums

Ans=

34

34

34

34

The sum of the elements on the main diagonal is easily obtained with the help of the

diag function, which picks off that diagonal.

diag (A) produces

ans=

16

10

7

1

and

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sum (diag (A)) produces ans = 34

A (1, 4) + A (2, 4) + A (3, 4) + A (4, 4)

This produces ans=34

X = A;

X (4, 5) = 17

X=

[16 3 2 13 0

5 10 11 8 0

9 6 7 12 0

4 15 14 1 17]

The colon operator

The colon is one of MATLab’s most important operators. It occurs in several different

forms. The expression

1:10

is a row vector containing the integers from 1 to 10

1 2 3 4 5 6 7 8 9 10

To obtain nonunit spacing, specify an increment. For example

100:-7:50

is

100 93 86 79 72 65 58 51

And 0: pi/4: pi is 0 0.7854 1.5708 2.3562 3.1416

Subscript expressions involving colons refer to portions of a matrix.

A (1: k, j) is the first k elements of the jth column of A. So sum (A (1:4, 4))

The Magic Function

B = magic (4)

B =

[16 2 3 13

5 11 10 8

9 7 6 12

4 14 15 1]

Operators

Expressions use familiar arithmetic operators and precedence rules.

+ Addition

- Subtraction

* Multiplication

/ Division

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DIGITAL SIGNAL PROCESSING LAB 24

\ Left division

(Described in the section on matrices and linear algebra in using MATLAB)

^ Power

‘Complex conjugate transpose

( ) Specify evaluation order

Generating Matrices

MATLAB provides four functions that generate basic matrices:

Some examples:

Z = zeros (2, 4)

Z =

0 0 0 0

0 0 0 0

F = 5*ones (3, 3)

F =

5 5 5

5 5 5

5 5 5

N = fix (10*rand (1, 10))

N =

4 9 4 4 8 5 2 6 8 0

Zeros all zeros

Ones All ones

rand Uniformly distributed random elements

randn Normally distributed random elements

15

R =

1.0668 0.2944-0.6918-1.4410

0.0593-1.3362 0.8580 0.5711

-0.0956 0.7143 1.2540-0.3999

-0.8323 1.6236-1.5937 0.6900

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1. TO STUDY THE ARCHITECTURE OF DSP CHIPS –

TMS 320C SX/6X INSTRUCTIONS.

The C6711™ DSK builds on TI's industry-leading line of low cost, easy-to-use DSP

Starter Kit (DSK) development boards. The high-performance board features the

TMS320C6711 floating-point DSP. Capable of performing 900 million floating-point

operations per second (MFLOPS), the C6711 DSP makes the C6711 DSK the most

powerful DSK development board on the market.

The DSK is a parallel port interfaced platform that allows TI, its customers, and third-

parties, to efficiently develop and test applications for the C6711. The DSK consists

of a C6711-based printed circuit board that will serve as a hardware reference design

for TI’s customers’ products. With extensive host PC and target DSP software

support, including bundled TI tools, the DSK provides ease-of-use and capabilities

that are attractive to DSP engineers.

TMS320C6711 DSK Block Diagram

The basic function block diagram and interfaces of the C6711 DSK are displayed

below.

Many of the parts of the block diagram are linked to specific topics.

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DIGITAL SIGNAL PROCESSING LAB 26

The C6711 DSK board is displayed below.

Many of the parts of the board illustration are linked to specific topics. Move the

mouse over the illustration to determine which parts are linked. Click on the portion of

the illustration for which you want to see information.

The C6711DSK has the following features:

• 150-MHz C6711DSP capable of executing 900 million floating-point

operations per second (MFLOPS)

• Dual clock support; CPU at 150MHz and external memory interface (EMIF) at

100MHz

• Parallel port controller (PPC) interface to standard parallel port on a host PC

(EEP or bi-directional SPP support)

• 16M Bytes of 100 MHz synchronous dynamic random access memory

(SDRAM)

• 128K Bytes of flash programmable and erasable read only memory (ROM)

• 8-bit memory-mapped I/O port

• Embedded JTAG emulation via the parallel port and external XDS510 support

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DIGITAL SIGNAL PROCESSING LAB 27

• Host port interface (HPI) access to all DSP memory via the parallel port

• 16-bit audio codec

• Onboard switching voltage regulators for 1.8 volts direct current (VDC) and

3.3 VDC

The C6711 DSK has approximate dimensions of 5.02 inches wide, 8.08 inches long

and 0.7 inches high. The C6711 DSK is intended for desktop operation while

connected to the parallel port of your PC or using an XDS510 emulator. The DSK

requires that the external power supply be connected in either mode of operation (an

external power supply and a parallel cable are provided in the kit).

The C6711 DSK has a TMS320C6711 DSP onboard that allows full-speed

verification of code with Code Composer Studio. The C6711 DSK provides:

• a parallel peripheral interface

• SDRAM and ROM

• a 16-bit analog interface circuit (AIC)

• an I/O port

• embedded JTAG emulation support

Connectors on the C6711 DSK provide DSP external memory interface (EMIF) and

peripheral signals that enable its functionality to be expanded with custom or third

party daughter boards.

The DSK provides a C6711 hardware reference design that can assist you in the

development of your own C6711-based products. In addition to providing a reference

for interfacing the DSP to various types of memories and peripherals, the design also

addresses power, clock, JTAG, and parallel peripheral interfaces.

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Top Side of the Board:

Bottom Side of the Board:

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TMS320C6711 DSP Features:

• VelociTI advanced very long instruction word (VLIW) architecture

• Load-store architecture

• Instruction packaging for reduced code size

• 100% conditional instructions for faster execution

• Intuitive, reduced instruction set computing with RISC-like instruction set

• CPU

• Eight highly independent functional units (including six ALUs and two

multipliers)

• 32 32-bit general-purpose registers

• 900 million floating-point operations per second (MIPS)

• 6.7-ns cycle time

• Up to eight 32-bit instructions per cycle

• Byte-addressable (8-, 16-, 32-bit data)

• 32-bit address range

• 8-bit overflow protection

• Little- and big-endian support

• Saturation

• Normalization

• Bit-field instructions (extract, set, clear)

• Bit-counting

• Memory/peripherals

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L1/L2 memory architecture:

32K-bit (4-K byte) L1P program cache (direct mapped)

32K-bit (4K-byte) L1D data cache (2-way set-associative)

512K-bit (64K byte) L2 unified map RAM/cache (flexible data/program allocation)

32-bit external memory interface (EMIF):

Glue less interface to synchronous memories: SDRAM and SBSRAM

Glue less interface to asynchronous memories: SDRAM and EPROM

Enhanced direct-memory-access (EDMA) controller

16-bit host-port interface (HPI) (access to entire memory map)

Two multi-channel buffered serial ports (McBSPs)

Two 32-bit general-purpose timers

Flexible phase-locked-loop (PLL) clock generator

Miscellaneous

IEEE-1149.1 (JTAG) boundary-scan-compatible for emulation and test support

256-lead ball grid array (BGA) package (GFN suffix)

0.18-mm/5-level metal process with CMOS technology

3.3-V I/Os, 1/8-V internal

C67x (Specific) Floating-Point Instructions:

Alphabetical Listing of 'C67x (Specific) Floating-Point Instructions

A - F G - L M - R S – Z

ABSDP INTDP MPYDP SPDP

ABSSP INTDPU MPYI SPINT

ADDAD INTSP MPYID SPTRUNC

ADDDP INTSPU MPYSP SUBDP

ADDSP LDDW RCPDP SUBSP

CMPEQDP RCPSP

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CMPEQSP RSQRDP

CMPGTDP RSQRSP

CMPGTSP

CMPLTDP

CMPLTSP

DPINT

DPSP

DPTRUNC

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2. TO VERIFY LINEAR CONVOLUTION

Aim: To compute the linear convolution of two discrete sequences.

Theory: Consider two finite duration sequences x (n) and h (n), the duration of x

(n) is n1 samples in the interval ) 1 ( 0

1

− ≤ ≤ n n . The duration of h (n) is n2 samples;

that is h (n) is non – zero only in the interval ) 1 ( 0

2

− ≤ ≤ n n . The linear or a periodic

convolution of x (n) and h (n) yields the sequence y (n) defined as,

Y (n) =

∑

=

−

n

m

m n x m h

0

) ( ) (

Clearly, y (n) is a finite duration sequence of duration (n1+n2 -1) samples.

The convolution sum of two sequences can be found by using following steps

Step1: Choose an initial value of n, the starting time for evaluating the output

sequence y (n). If x (n) starts at n=n

1

and h (n) starts at n= n

2

then n = n

1

+ n

2

-1 is a

good choice.

Step2: Express both sequences in terms of the index m.

Step3: Fold h (m) about m=0 to obtain h (-m) and shift by n to the right if n is positive

and left if n is negative to obtain h (n-m).

Step4: Multiply two sequences x (n-m) and h (m) element by element and sum the

products to get y (n).

Step5: Increment the index n, shift the sequence x (n-m) to right by one sample and

repeat step4.

Step6: Repeat step5 until the sum of products is zero for all remaining values of n.

Program:

%Linear convolution of two sequences

a=input('enter the input sequence1=');

b=input('enter the input sequence2=');

n

1

=length (a)

n

2

=length (b)

x=0:1:n1-1;

Subplot (2, 2, 1), stem(x, a);

title ('INPUT SEQUENCE1');

xlabel ('---->n');

ylabel ('---->a (n)');

y=0:1:n2-1;

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subplot (2, 2, 2), stem(y, b);

title ('INPUT SEQUENCE2');

xlabel ('---->n');

ylabel('---->b(n)');

c=conv (a, b)

n

3

=0:1:n1+n2-2;

subplot (2, 1, 2), stem (n3, c);

title ('CONVOLUTION OF TWO SEQUENCES');

xlabel ('---->n');

ylabel ('---->c (n)');

Output:

Enter the input sequence1= [2 3 4]

Enter the input sequence2= [1 2 3]

c = 2 7 16 17 12

Enter the input sequence1= [2 3 4]

Enter the input sequence2= [1 2 3]

n

1

= 3

n

2

=3

c = 2 7 16 17 12

Result: Linear convolution of two discrete sequences is computed.

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Inference: The length of the Linear convolved sequence is n

1

+n

2

-1 where n

1

and

n

2

are lengths of sequences.

Questions & Answers

1. What do you understand by Linear Convolution.

a. The convolution of discrete – time signals is known as discrete convolution. Let x

(n) be the input to an LTI system and y (n) be the output of the system. Let h (n) be

the response of the system to an impulse. The output y (n) can be obtained by

convolving the impulse response h (n) and the input signal x (n)

Y (n) =

∑

∞

−∞ =

−

m

m n x m h ) ( ) ( the above equation that gives the response y (n)

of an LTI system as a function of the input signal x (n) and the impulse response h

(n) is called a convolution sum.

2. What are the properties of convolution.

a. 1. Commutative property x (n) * h (n) = h (n) * x (n)

2. Associative property [x (n) * h

1

(n)] * h

2

(n) = x (n) * [h

1

(n) * h

2

(n)]

3. Distributive property X (n) * [h

1

(n) + h

2

(n)] = x (n) * h

1

(n) + x (n) * h

2

(n)].

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3. TO VERIFY CIRCULAR CONVOLUTION

Aim: To compute the circular convolution of two discrete sequences.

Theory: Two real N-periodic sequences x(n) and h(n) and their circular or periodic

convolution sequence y(n) is also an N- periodic and given by,

∑

−

=

− =

1

0

)) (( ) ( ) (

N

m

N

m n x m h n y , for n=0 to N-1.

Circular convolution is some thing different from ordinary linear convolution

operation, as this involves the index ((n-m))N, which stands for a ‘modula-N’

operation. Basically both type of convolution involve same six steps. But the

difference between the two types of convolution is that in circular convolution the

folding and shifting (rotating) operations are performed in a circular fassion by

computing the index of one of the sequences with ‘modulo-N’ operation. Either one of

the two sequences may be folded and rotated without changing the result of circular

convolution. That is,

∑

−

=

− =

1

0

)) (( ) ( ) (

N

m

N

m n h m x n y , for n=0 to (N-1).

If x (n) contain L no of samples and h (n) has M no of samples and that L > M, then

perform circular convolution between the two using N=max (L,M), by adding (L-M) no

of zero samples to the sequence h (n), so that both sequences are periodic with

number.

Two sequences x (n) and h (n), the circular convolution of these two sequences

can be found by using the following steps.

1. Graph N samples of h (n) as equally spaced points around an outer circle in

counterclockwise direction.

2. Start at the same point as h (n) graph N samples of x (n) as equally spaced

points around an inner circle in clock wise direction.

3. Multiply corresponding samples on the two circles and sum the products to

produce output.

4. Rotate the inner circle one sample at a time in counter clock wise direction

and go to step 3 to obtain the next value of output.

5. Repeat step No.4 until the inner circle first sample lines up with first sample of

the exterior circle once again.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 36

Program:

%circular convolution

a=input('enter the input sequence1=');

b=input('enter the input sequence2=');

n

1

=length (a)

n

2

=length (b)

n

3

=n

1

-n

2

N=max (n

1

, n2);

if (n

3

>0)

b= [b, zeros (1, n1-n

2

)]

n

2

=length (b);

else

a= [a, zeros (1, N-n

1

)]

n

1

=length (a);

end;

k=max (n

1

, n2);

a=a';

b=b';

c=b;

for i=1: k-1

b=circshift (b, 1);

c=[c, b];

end;

disp(c);

z=c*a;

disp ('z=')

disp (z)

subplot (2, 2, 1);

stem (a,'filled');

title ('INPUT SEQUENCE1');

xlabel ('---->n');

ylabel ('---->Amplitude')

subplot (2, 2, 2);

stem (b,'filled');

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DIGITAL SIGNAL PROCESSING LAB 37

title ('INPUT SEQUENCE2');

xlabel ('---->n');

ylabel ('---->Amplitude');

subplot (2, 1, 2);

stem (z,'filled');

title ('CONVOLUTION OF TWO SEQUENCES');

xlabel ('---->n');

ylabel ('---->Amplitude');

Output:

Enter the input sequence1= [2 3 4 5]

Enter the input sequence2= [1 1 0]

n

1

=4

n

2

=3

n

3

=1

b =

1 1 0 0

1 0 0 1

1 1 0 0

0 1 1 0

0 0 1 1

z=

7

5

7

9

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 38

Result: Circular convolution of two discrete sequences is computed.

Inference: The length of the circularly convolved sequence is max (n

1

, n2) where

n

1

and n

2

are the lengths of given sequences.

Questions & Answers

1. Define circular convolution?

a. Let x

1

(n) and x

2

(n) are finite duration sequences both of length N with DFTs X

1

(k)

and x

2

(k).if X

3

(k)=X

1

(k)X

2

(k), then the sequence x

3

(n) can obtained by circular

convolution, defined as x

3

(n)=

N

N

m

m n x m x )) (( ) (

2

1

0

1

−

∑

−

=

.

2. What do you understand by periodic convolution?

a. Let x1p (n) and x

2

p (n) with a period of N samples so that x

p

(n) =x

p

(n+lN); Then

discrete Fourier series of the sequence x

p

(n) is defined as X

p

(k)

=

∑

−

=

Π −

1

0

/ 2

) (

N

n

N kn j

p

e n x

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 39

4. DESIGN FIR FILTER (LP/HP) USING WINDOWING

TECHNIQUE

a) USING RECTANGULAR WINDOW

Aim: To design FIR high pass filter using rectangular window.

Theory: The design frequency response H

d

(e

jw

) of a filter is periodic in frequency

and can be expanded in a Fourier series. The resultant series is given by

H

d

(e

jw

) = ∑

∞

−∞ =

−

n

jwn

d

e n h ) (

Where

H

d

(n) =

ω

ω ω

d e e H

n j j

∫

Π

Π −

Π ) ( 2 / 1

and known as fourier coefficients having infinite length. One possible way of

obtaining FIR filter is to truncate the infinite fourier series at n = +/- [N-1/2], Where N

is the length of the desired sequence. But abrupt truncation of the Fourier series

results in oscillation in the passband and stopband. These oscillations are due to

slow convergence of the fourier series and this effect is known as the Gibbs

phenomenon. To reduce these oscillations, the Fourier coefficients of the filter are

modified by multiplying the infinite impulse response with a finite weighing sequence

ω (n) called a window.

where

ω (n) = ω (-n) ≠ 0 for |n| [ ≤ (N-1/2]/2]

=0 for |n|> (N-1/2]/2]

After multiplying window sequence ω (n) with h

d

(n), we get a finite duration

sequence h (n) that satisfies the desired magnitude response.

h (n) = h

d

(n) ω (n) for all |n| ≤[N-1/2]

=0 for |n|> [N-1/2]

The frequency response H (e

jw

) so the filter can be obtained by convolution of H

d

(e

JW

)

and W (e

jw

) given by

H(e

jw

) = 1/ θ

θ ω θ

d e W e H

j j

d

) (

( ) ( 2

−

Π

Π −

∫

Π

H

d

(e

jw

) * W (e

jw

)

Because both H

d

(e

jw

) and W (e

jw

) are periodic function, the operation often called as

periodic convolution.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 40

The rectangular window sequence is given by

WR (n) = 1 for – (N-1/2 ≤n ≤(n-1)/2

= 0 otherwise

Program:

%Design of high pass filter using rectangular window

WC=0.5*pi;

N=25;

b=fir1 (N-1, wc/pi,'high', rectwin (N));

disp ('b=');

disp (b)

w=0:0.01: pi;

h=freqz (b, 1, w);

plot (w/pi, abs (h));

xlabel ('Normalised Frequency');

ylabel ('Gain in dB')

title ('FREQUENCY RESPONSE OF HIGHPASS FILTER USING

RECTANGULAR WINDOW');

Output:

b= Columns 1 through 17

-0.0000 0.0297 -0.0000 -0.0363 -0.0000 0.0467 -0.0000 -0.0654 -

0.0000 0.1090 -0.0000 -0.3269 0.5135 -0.3269 -0.0000 0.1090 -

0.0000

Columns 18 through 25

-0.0654 -0.0000 0.0467 -0.0000 -0.0363 -0.0000 0.0297 -0.0000

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 41

Result: FIR high pass filter is designed by using rectangular window.

Inference: Magnitude response of FIR high pass filter designed using

Rectangular window as significant sidelobes.

Questions & Answers

1. What are "FIR filters"?

a. FIR filters are one of two primary types of digital filters used in Digital Signal

Processing (DSP) applications (the other type being IIR).

2. What does "FIR" mean?

a.” FIR" means "Finite Impulse Response".

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 42

3. Why is the impulse response "finite"?

a. The impulse response is "finite" because there is no feedback in the filter; if you

put in an impulse (that is, a single "1" sample followed by many "0" samples),

zeroes will eventually come out after the "1" sample has made its way in the delay

line past all the coefficients.

4. How do you pronounce "FIR"?

a. Some people say the letters F-I-R; other people pronounce as if it were a type of

tree. We prefer the tree. (The difference is whether you talk about an F-I-R filter or

a FIR filter.)

5. What is the alternative to FIR filters?

a. DSP filters can also be "Infinite Impulse Response" (IIR). IIR filters use feedback,

so when you input an impulse the output theoretically rings indefinitely.

6. How do FIR filters compare to IIR filters?

a. Each has advantages and disadvantages. Overall, though, the advantages of FIR

filters outweigh the disadvantages, so they are used much more than IIRs.

7. What are the advantages of FIR Filters (compared to IIR filters)?

a. Compared to IIR filters, FIR filters offer the following advantages:

• They can easily be designed to be "linear phase" (and usually are).

Put simply, linear-phase filters delay the input signal, but don’t distort

its phase.

• They are simple to implement. On most DSP microprocessors, the

FIR calculation can be done by looping a single instruction.

• They are suited to multi-rate applications. By multi-rate, we mean

either "decimation" (reducing the sampling rate), "interpolation"

(increasing the sampling rate), or both. Whether decimating or

interpolating, the use of FIR filters allows some of the calculations to

be omitted, thus providing an important computational efficiency.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 43

In contrast, if IIR filters are used, each output must be individually

calculated, even if it that output will discarded (so the feedback will be

incorporated into the filter).

• They have desirable numeric properties. In practice, all DSP filters

must be implemented using "finite-precision" arithmetic, that is, a

limited number of bits. The use of finite-precision arithmetic in IIR

filters can cause significant problems due to the use of feedback, but

FIR filters have no feedback, so they can usually be implemented

using fewer bits, and the designer has fewer practical problems to

solve related to non-ideal arithmetic.

• They can be implemented using fractional arithmetic. Unlike IIR filters,

it is always possible to implement a FIR filter using coefficients with

magnitude of less than 1.0. (The overall gain of the FIR filter can be

adjusted at its output, if desired.) This is an important consideration

when using fixed-point DSP's, because it makes the implementation

much simpler.

8. What are the disadvantages of FIR Filters (compared to IIR filters)?

a. Compared to IIR filters, FIR filters sometimes have the disadvantage that they

require more memory and/or calculation to achieve a given filter response

characteristic. Also, certain responses are not practical to implement with FIR

filters.

9. What terms are used in describing FIR filters?

• Impulse Response - The "impulse response" of a FIR filter is actually

just the set of FIR coefficients. (If you put an "impulse" into a FIR filter

which consists of a "1" sample followed by many "0" samples, the

output of the filter will be the set of coefficients, as the 1 sample

moves past each coefficient in turn to form the output.)

• Tap - A FIR "tap" is simply a coefficient/delay pair. The number of FIR

taps, (often designated as "N") is an indication of 1) the amount of

memory required to implement the filter, 2) the number of calculations

required, and 3) the amount of "filtering" the filter can do; in effect,

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 44

more taps means more stopband attenuation, less ripple, narrower

filters, etc.)

• Multiply-Accumulate (MAC) - In a FIR context, a "MAC" is the

operation of multiplying a coefficient by the corresponding delayed

data sample and accumulating the result. FIRs usually require one

MAC per tap. Most DSP microprocessors implement the MAC

operation in a single instruction cycle.

• Transition Band - The band of frequencies between passband and

stopband edges. The narrower the transition band, the more taps are

required to implement the filter. (A "small" transition band results in a

"sharp" filter.)

• Delay Line - The set of memory elements that implement the "Z^-1"

delay elements of the FIR calculation.

• Circular Buffer - A special buffer which is "circular" because

incrementing at the end causes it to wrap around to the beginning, or

because decrementing from the beginning causes it to wrap around to

the end. Circular buffers are often provided by DSP microprocessors

to implement the "movement" of the samples through the FIR delay-

line without having to literally move the data in memory. When a new

sample is added to the buffer, it automatically replaces the oldest one.

b) USING KAISER WINDOW

Aim: To design FIR low pass filter using Kaiser Window for different values of

Beeta and verify its characteristics.

Theory: The Kaiser window is given by

W

k

(n) = I

o

[ ) ( / ) ) 1 / 2 ( 1 (

0

2

α α I N n − − ] for |n| ≤ N-1/2

=0 otherwise

Where α is an independent parameter.

I

o

(x) is the zeroth order Bessel function of the first kind

I

0

(x) = 1+

2 k

1

] (x/2) ! / 1 [

∑

∞

= k

k

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 45

Advantages of Kaiser Window

1. It provides flexibility for the designer to select the side lobe level and N.

2. It has the attractive property that the side lobe level can be varied

continuously from the low value in the Blackman window to the high value

in the rectangular window.

Program:

%Design of FIR lowpass filter using Kaiser Window

WC=0.5*pi;

N=25;

b=fir1 (N, wc/pi, kaiser (N+1, 0.5))

w=0:0.01: pi;

h=freqz (b, 1, w);

plot (w/pi, 20*log10 (abs (h)));

hold on

b=fir1 (N, wc/pi, kaiser (N+1, 3.5))

w=0:0.01: pi;

h=freqz (b, 1, w);

plot (w/pi, 20*log10 (abs (h)));

hold on

b=fir1 (N, wc/pi, kaiser (N+1, 8.5))

w=0:0.01: pi;

h=freqz (b, 1, w);

plot (w/pi, 20*log10 (abs (h)));

xlabel ('Normalised Frequency');

ylabel ('Magnitude in dB');

title ('FREQUENCY RESPONSE OF LOWPASS FILTER USING

KAISER WINDOW');

hold off

Output:

b = Columns 1 through 17

0.0170 -0.0186 -0.0206 0.0229 0.0258 -0.0294 -0.0341 0.0405

0.0497 -0.0641 -0.0899 0.1501 0.4507 0.4507 0.1501 -0.0899 -

0.0641

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 46

Columns 18 through 26

0.0497 0.0405 -0.0341 -0.0294 0.0258 0.0229 -0.0206 -0.0186

0.0170

b = Columns 1 through 17

0.0024 -0.0041 -0.0062 0.0089 0.0124 -0.0169 -0.0227 0.0305

0.0412 -0.0573 -0.0849 0.1471 0.4496 0.4496 0.1471 -0.0849 -

0.0573

Columns 18 through 26

0.0412 0.0305 -0.0227 -0.0169 0.0124 0.0089 -0.0062 -0.0041

0.0024

b = Columns 1 through 17

0.0000 -0.0002 -0.0006 0.0015 0.0032 -0.0062 -0.0109 0.0182

0.0293 - 0.0467 -0.0766 0.1416 0.4473 0.4473 0.1416 -0.0766 -

0.0467

Columns 18 through 26

0.0293 0.0182 -0.0109 -0.0062 0.0032 0.0015 -0.0006 -0.0002

0.0000

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 47

Result: FIR low pass filter is designed by using Kaiser Window for different values

of Beeta.

Inference: The response of stop band improves as beeta increases.

Questions & Answers

1. What are the design techniques of designing FIR filters?

a. There are three well-known methods for designing FIR filters with linear

phase.These are (1) window method (2) frequency sampling method (3) optimal or

minimax design.

2. What is the reason that FIR filter is always stable?

a. FIR filter is always stable because all its poles are at the origin.

3. What are the properties of FIR filter?

• FIR filter is always stable.

• A realizable filter can always be obtained.

• FIR filter has a linear phase response.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 48

5. DESIGN IIR FILTER (LP/HP)

a)

Aim: To design IIR Butterworth low pass filter and verify its characteristics.

Theory: The most common technique used for designing IIR digital filters known

as indirect method, involves first designing an analog prototype filter and then

transforming the prototype to a digital filter. For the given specifications of a digital

filter, the derivation of the digital filter transfer function requires three steps.

1. Map the desired digital filter specifications into those for an equivalent

analog filter.

2. Derive the analog transfer function for the analog prototype.

3. Transform the transfer function of the analog prototype into an equivalent

digital filter transfer function.

There are several methods that can be used to design digital filters having an infinite

during unit sample response. The techniques described are all based on converting

an analog filter into digital filter. If the conversion technique is to be effective, it

should posses the following desirable properties.

1. The jΩ-axis in the s-plane should map into the unit circle in the z-plane.

Thus there will be a direct relationship between the two frequency

variables in the two domains.

2. The left-half plane of the s-plane should map into the inside of the unit

circle in the z-plane. Thus a stable analog filter will be converted to a

stable digital filter.

Design a digital filter using impulse invariance method

1. For the given specifications, find H

a

(s), the transfer function of an analog

filter.

2. Select the sampling rate of the digital filter, T seconds per sample.

3. Express the analog filter transfer function as the sum of single-pole filters.

H

a

(s) =

∑

=

−

N

K

k

p s ck

1

/

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 49

4. Compute the z-transform of the digital filter by using the formula

H (z) =

∑

=

−

−

N

K

pkT

k

z e c

1

1

1 /

For high sampling rates use

H (z) =

∑

=

−

−

N

K

pkT

k

z e Tc

1

1

1 /

Design digital filter using bilinear transform technique.

1. From the given specifications, find prewarping analog frequencies using formula

Ω= 2/T tan w/2.

2. Using the analog frequencies find H (s) of the analog filter.

3. Select the sampling rate of the digital filter, call it T seconds per sample.

4. Substitute s= 2/T (1-z

-1

/1+ z

-1

) into the transfer function found in step2.

Program:

%Design of IIR Butterworth lowpass filter

alphap=input ('enter the pass band attenuation=');

alphas=input ('enter the stop band attenuation=');

fp=input ('enter the passband frequency=');

fs=input ('enter the stop band frequency=');

F=input ('enter the sampling frequency=');

omp=2*fp/F;

oms=2*fs/F;

[n, wn]=buttord (omp, oms, alphap, alphas)

[b, a]=butter (n, wn)

w=0:0.1: pi;

[h, ph]=freqz (b, a, w);

m=20*log (abs (h));

an=angle (h);

subplot (2, 1, 1);

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DIGITAL SIGNAL PROCESSING LAB 50

plot (ph/pi, m);

grid on;

ylabel ('Gain in dB');

xlabel ('Normalised Frequency');

title (‘FREQUENCY RESPONSE OF BUTTERWORTH LOWPASS

subplot (2, 1, 2);

title (‘PHASE RESPONSE OF BUTTERWORTH LOWPASS FILTER’);

plot (ph/pi, an);

grid on;

ylabel ('Phase in Radians');

xlabel ('Normalised Frequency');

Output:

Enter the pass band attenuation=0.4

Enter the stop band attenuation=60

Enter the pass band frequency=400

Enter the stop band frequency=800

Enter the sampling frequency=2000

n = 6

Wn =0.4914

b =0.0273 0.1635 0.4089 0.5452 0.4089 0.1635 0.0273

a = 1.0000 -0.1024 0.7816 -0.0484 0.1152 -0.0032 0.0018

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 51

Result: IIR Butterworth low pass filter is designed.

Inference: There are no ripples in the passband and stopband.

Questions & Answers

1. What are the advantages of IIR filters (compared to FIR filters)? What are IIR

filters? What does "IIR" mean?

a. IIR filters are one of two primary types of digital filters used in Digital Signal

Processing (DSP) applications (the other type being FIR). "IIR" means "Infinite

Impulse Response".

2. Why is the impulse response "infinite"?

a. The impulse response is "infinite" because there is feedback in the filter; if you put

in an impulse (a single "1" sample followed by many "0" samples), an infinite

number of non-zero values will come out (theoretically).

3. What is the alternative to IIR filters?

a. DSP filters can also be "Finite Impulse Response" (FIR). FIR filters do not use

feedback, so for a FIR filter with N coefficients, the output always becomes zero

after putting in N samples of an impulse response.IIR filters can achieve a given

filtering characteristic using less memory and calculations than a similar FIR filter.

4. What are the disadvantages of IIR filters (compared to FIR filters)?

• They are more susceptible to problems of finite-length arithmetic, such

as noise generated by calculations, and limit cycles. (This is a direct

consequence of feedback: when the output isn't computed perfectly

and is fed back, the imperfection can compound.)

• They are harder (slower) to implement using fixed-point arithmetic.

• They don't offer the computational advantages of FIR filters for

multirate (decimation and interpolation) applications.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 52

b)

Aim: To design IIR Chebyshew type I high pass filter.

Program:

%Design of IIR Chebyshew type I high pass filter

alphap=input ('enter the pass band attenuation=');

alphas=input ('enter the stop band attenuation=');

fp=input ('enter the pass band frequency=');

fs=input ('enter the stop band frequency=');

F=input ('enter the sampling frequency=');

omp=2*fp/F;

oms=2*fs/F;

[n, wn]=cheb1ord (omp, oms, alphap, alphas)

[b, a]=cheby1 (n, wn, high)

w=0:0.1: pi;

[h, ph]=freqz (b, a, w);

m=20*log (abs (h));

an=angle (h);

subplot (2, 1, 1);

plot (ph/pi, m);

grid on;

ylabel ('Gain in dB');

xlabel ('Normalised Frequency');

title (‘FREQUENCY RESPONSE OF CHEBYSHEW TYPE I HIGH PASS FILTER’);

subplot (2, 1, 2);

plot (ph/pi, an);

grid on;

ylabel ('Phase in Radians');

xlabel ('Normalised Frequency');

title (‘PHASE RESPONSE OF CHEBYSHEW TYPE I HIGH PASS FILTER’);

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DIGITAL SIGNAL PROCESSING LAB 53

Output:

enter the pass band attenuation=1

enter the stop band attenuation=15

enter the passband frequency=0.2*pi

enter the stop band frequency=0.1*pi

n = 3

Wn =0.2000

b = 0.4759 -1.4278 1.4278 -0.4759

a = 1.0000 -1.6168 1.0366 -0.1540

Result: IIR Chebyshew type I high pass filter is designed on DSP Processors.

Inference: The transition band is less in Chebyshev filter compared to Butterworth

filter.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 54

Questions & Answers

1. What are the parameters that can be obtained from the Chebyshev filter

specifications?

a. From the given Chebyshev filter specifications we can obtain the parameters like

the order of the filter N, є, transition ratio k, and the poles of the filter.

2. Distinguish between Butterworth and Chebyshev (type-I) filter?

• The magnitude response of Butterworth filter decreases monotonically as the

frequency Ω increases from 0 to ∞, where as the magnitude response of the

Chebyshev filter exhibits ripple in the passband and monotonically decreasing

in the stopband.

• The transition band is more in Butterworth filter compared to Chebyshev filter.

• The poles of the Butterworth filter lies on a circle whereas the poles of the

Chebyshev filter lies on an ellipse.

• For the same specifications the number of poles in Butterworth filter is more

when compared to Chebyshev filter, i.e., the order of the Chebyshev filter is

less than that of Butterworth.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 55

6. N-POINT FFT ALGORITHM

Aim: To determine the N-point FFT of a given sequence.

Theory: The Fast Fourier transform is a highly efficient procedure for computing

the DFT of a finite series and requires less no of computations than that of direct

evaluation of DFT. It reduces the computations by taking the advantage of the fact

that the calculation of the coefficients of the DFT can be carried out iteratively. Due to

this, FFT computation technique is used in digital spectral analysis, filter simulation,

autocorrelation and pattern recognition. The FFT is based on decomposition and

breaking the transform into smaller transforms and combining them to get the total

transform. FFT reduces the computation time required to compute a discrete Fourier

transform and improves the performance by a factor 100 or more over direct

evaluation of the DFT. The fast fourier transform algorithms exploit the two basic

properties of the twiddle factor (symmetry property:

k

N

N k

N

w w − =

+ 2 /

,

periodicity property:

k

N

N k

N

w w =

+

) and reduces the number of complex

multiplications required to perform DFT from N

2

to N/2 log

2

N. In other words, for

N=1024, this implies about 5000 instead of 10

6

multiplications – a reduction factor of

200. FFT algorithms are based on the fundamental principal of decomposing the

computation of discrete fourier transform of a sequence of length N into successively

smaller discrete fourier transform of a sequence of length N into successively smaller

discrete fourier transforms. There are basically two classes of FFT algorithms. They

are decimation-in-time and decimation-in-frequency. In decimation-in-time, the

sequence for which we need the DFT is successively divided into smaller sequences

and the DFTs of these subsequences are combined in a certain pattern to obtain the

required DFT of the entire sequence. In the decimation-in-frequency approach, the

frequency samples of the DFT are decomposed into smaller and smaller

subsequences in a similar manner.

Radix – 2 DIT – FFT algorithm

1. The number of input samples N=2

M

, where, M is an integer.

2. The input sequence is shuffled through bit-reversal.

3. The number of stages in the flow graph is given by M=log

2

N.

4. Each stage consists of N/2 butterflies.

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DIGITAL SIGNAL PROCESSING LAB 56

5. Inputs/outputs for each butterfly are separated by 2

m-1

samples, where m

represents the stage index, i.e., for first stage m=1 and for second stage m=2

so on.

6. The no of complex multiplications is given by N/2 log

2

N.

7. The no of complex additions is given by N log

2

N.

8. The twiddle factor exponents are a function of the stage index m and is given

by . 1 2 ,... 2 , 1 , 0

2

1

− = =

− m

m

t

t

N

k

9. The no of sets or sections of butterflies in each stage is given by the formula

2

M-m.

10. The exponent repeat factor (ERF), which is the number of times the exponent

sequence associated with m is repeated is given by 2

M-m.

Radix – 2 DIF – FFT Algorithm

1. The number of input samples N=2

M

, where M is number of stages.

2. The input sequence is in natural order.

3. The number of stages in the flow graph is given by M=log

2

N.

4. Each stage consists of N/2 butterflies.

5. Inputs/outputs for each butterfly are separated by 2

M-m

samples, Where m

represents the stage index i.e., for first stage m=1 and for second stage m=2

so on.

6. The number of complex multiplications is given by N/2 log

2

N.

7. The number of complex additions is given by N log

2

N.

8. The twiddle factor exponents are a function of the stage index m and is given

by . 1 2 ,... 2 , 1 , 0 ,

2

1

− = =

−

+ −

m m

m M

t

t

N

k

9. The number of sets or sections of butterflies in each stage is given by the

formula 2

m-1.

10. The exponent repeat factor (ERF), which is the number of times the

exponent sequence associated with m repeated is given by 2

m-1

.

For decimation-in-time (DIT), the input is bit-reversed while the output is in natural

order. Whereas, for decimation-in-frequency the input is in natural order while the

output is bit reversed order.

The DFT butterfly is slightly different from the DIT wherein DIF the complex

multiplication takes place after the add-subtract operation.

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DIGITAL SIGNAL PROCESSING LAB 57

Both algorithms require N log

2

N operations to compute the DFT. Both algorithms

can be done in-place and both need to perform bit reversal at some place during the

computation.

Program:

% N-point FFT algorithm

N=input ('enter N value=');

Xn=input ('type input sequence=');

k=0:1: N-1;

L=length (xn)

if (N<L)

error ('N MUST BE>=L');

end;

x1= [xn zeros (1, N-L)]

for c=0:1:N-1;

for n=0:1:N-1;

p=exp (-i*2*pi*n*C/N);

x2(c+1, n+1) =p;

end;

Xk=x1*x2';

end;

MagXk=abs (Xk);

angXk=angle (Xk);

subplot (2, 1, 1);

stem (k, magXk);

title ('MAGNITUDE OF DFT SAMPLES');

xlabel ('---->k');

ylabel ('-->Amplitude');

subplot (2, 1, 2);

stem (k, angXk);

title ('PHASE OF DFT SAMPLES');

xlabel ('---->k');

ylabel ('-->Phase');

disp (‘abs (Xk) =');

disp (magXk)

disp ('angle=');

disp (angXk)

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DIGITAL SIGNAL PROCESSING LAB 58

Output:

enter N value=5

type input sequence= [1 1 0]

L= 3

x1 = 1 1 0 0 0

abs (Xk) = 2.0000 1.6180 0.6180 0.6180 1.6180

angle= 0 0.6283 1.2566 -1.2566 -0.6283

Result: N-point FFT of a given sequence is determined.

Inference: FFT reduces the computation time required to compute discrete

Fourier transform

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DIGITAL SIGNAL PROCESSING LAB 59

Questions & Answers

1. What are the applications of FFT algorithms?

a. The applications of FFT algorithm includes

• Linear filtering

• Correlation

• Spectrum analysis.

2. What is meant by radix-2 FFT?

a. The FFT algorithm is most efficient in calculating N- point DFT. If the number of

output points N can be expressed as a power of 2, that is, N=2

M,

where M is an

integer, then this algorithm is known as radix -2 FFT algorithm.

3. What is an FFT "radix"?

a. The "radix" is the size of FFT decomposition. In the example above, the radix was

2. For single-radix FFT's, the transform size must be a power of the radix. In the

example above, the size was 32, which is 2 to the 5th power.

4. What are "twiddle factors"?

a. "Twiddle factors" are the coefficients used to combine results from a previous

stage to form inputs to the next stage.

5. What is an "in place" FFT?

a. An "in place" FFT is simply an FFT that is calculated entirely inside its original

sample memory. In other words, calculating an "in place" FFT does not require

additional buffer memory (as some FFT's do.)

6. What is "bit reversal"?

a. "Bit reversal" is just what it sounds like: reversing the bits in a binary word from left

to write. Therefore the MSB's become LSB's and the LSB's become MSB's. But what

does that have to do with FFT's? Well, the data ordering required by radix-2 FFT's

turns out to be in "bit reversed" order, so bit-reversed indexes are used to combine

FFT stages. It is possible (but slow) to calculate these bit-reversed indices in

software; however, bit reversals are trivial when implemented in hardware. Therefore,

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DIGITAL SIGNAL PROCESSING LAB 60

almost all DSP processors include a hardware bit-reversal indexing capability (which

is one of the things that distinguishes them from other microprocessors.)

7. What is "decimation in time" versus "decimation in frequency"?

a. FFT's can be decomposed using DFT's of even and odd points, which is called a

Decimation-In-Time (DIT) FFT, or they can be decomposed using a first-half/second-

half approach, which is called a "Decimation-In-Frequency" (DIF) FFT. Generally, the

user does not need to worry which type is being used.

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DIGITAL SIGNAL PROCESSING LAB 61

7. MATLAB PROGRAM TO GENERATE SUM OF SINUSOIDAL

SIGNALS

Aim: To generate sum of sinusoidal signals.

Theory:

Sinusoidal sequence:

X (n) = cos (w

0

n +θ ), n ∀

Where θ is the phase in radians. A MATLAB function cos (or sin) is used to

generate sinusoidal sequences.

Signal addition: This is a sample-by-sample addition given by

{X1 (n)} + {x

2

(n)} = {x

1

(n) + x

2

(n)}

It is implemented in Matlab by the arithmetic operator ‘’+’’. However, the lengths of

x

1

(n) and x

2

(n) must be the same. If sequences are of unequal lengths, or if the

sample positions are different for equal length sequences, then we cannot directly

use the operator + . We have to first augment x

1

(n) and x

2

(n) so that they have the

same position vector n (and hence the same length). This requires careful attention

to MATLab’s indexing operations. In particular, logical operation of intersection ‘’&’’

relational operations like ‘’<=’’ and ‘’==’’ and the find function are required to make

x

1

(n) amd x

2

(n) of equal length.

Program:

%Generation of sum of sinusoidal sequences

n=-4:0.5:4;

y1=sin (n)

subplot (2, 2, 1)

stem (n, y1,'filled')

grid on;

xlabel ('-->Samples');

ylabel ('--->Magnitude');

title ('SINUSOIDAL SIGNAL1');

y2=sin (2*n)

subplot (2, 2, 2)

stem (n, y2,'filled')

grid on;

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DIGITAL SIGNAL PROCESSING LAB 62

xlabel ('-->Samples');

ylabel ('--->Magnitude');

title ('SINUSOIDAL SIGNAL2');

y3=y1+y2

subplot (2, 1, 2);

stem (n, y3,'filled');

xlabel ('-->Samples');

ylabel ('--->Magnitude');

title ('SUM OF SINUSOIDAL SIGNALS');

Output:

y1 = 0.7568 0.3508 -0.1411 -0.5985 -0.9093 -0.9975 -0.8415 -

0.4794

0 0.4794 0.8415 0.9975 0.9093 0.5985 0.1411 -0.3508 -

0.7568

y2 = -0.9894 -0.6570 0.2794 0.9589 0.7568 -0.1411 -0.9093 -

0.8415

0 0.8415 0.9093 0.1411 -0.7568 -0.9589 -0.2794 0.6570

0.9894

y3 =-0.2326 -0.3062 0.1383 0.3605 -0.1525 -1.1386 -1.7508 -

1.3209

0 1.3209 1.7508 1.1386 0.1525 -0.3605 -0.1383 0.3062

0.2326

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DIGITAL SIGNAL PROCESSING LAB 63

Result: Sum of sinusoidal signals is generated.

Inference: The lengths of x

1

(n) and x

2

(n) must be the same for sample-by-

sample addition.

Questions & Answers

1. What is deterministic signal? Give example.

a. A deterministic signal is a signal exhibiting no uncertainty of value at any given

instant of time. Its instantaneous value can be accurately predicted by specifying a

formula, algorithm or simply its describing statement in words.

Example; v (t) =A

o

sinwt

2. Define a periodic signal?

a. A signal x (n) is periodic with period N if and only if x (n+N) =x (n) for all n.

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DIGITAL SIGNAL PROCESSING LAB 64

8. MATLAB PROGRAM TO FIND FREQUENCY RESPONSE OF

ANALOG LP/HP FILTERS.

a)

Aim: To find frequency response of Butterworth analog low pass filter.

Theory: The magnitude function of the butterworth lowpass filter is given by

2 / 1 2

] ) / ( 1 /[ 1 ) (

N

c

j H Ω Ω + = Ω N = 1, 2, 3,

Where N is the order of the filter and Ω

c

is the cutoff frequency.

The following expression is used for the order of the filter.

P

S

P

S

N

Ω

Ω

−

−

=

log

1 10

1 10

log

1 . 0

1 . 0

α

α

Two properties of Butterworth lowpass filter

1. The magnitude response of the butterworth filter decreases

monotonically as the frequency Ω increases from 0 to ∞.

2. The magnitude response of the butterworth filter closely approximates

the ideal response as the order N increases.

3. The poles of the Butterworth filter lies on a circle.

Steps to design an analog butterworth lowpass filter

1. From the given specifications find the order of the filter N.

2. Round off it to the next higher integer.

3. Find the transfer function H (s) for Ω

c

= 1 rad/sec for the value of N.

4. Calculate the value of cutoff frequency Ω

c.

5. Find the transfer function H

a

(s) for the above value of Ω

c

by substituting

s s/Ω

c

in H(s).

Program:

%Design of Butterworth lowpass filter

alphap=input ('enter the pass band attenuation=');

alphas=input ('enter the stop band attenuation=');

fp=input ('enter the passband frequency=');

fs=input ('enter the stop band frequency=');

F=input ('enter the sampling frequency=');

omp=2*fp/F;

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DIGITAL SIGNAL PROCESSING LAB 65

oms=2*fs/F;

[N, wn]=buttord (omp, oms, alphap, alphas,'s')

[b, a]=butter (n, wn)

w=0:0.01: pi;

[h, ph]=freqz (b, a, w);

m=20*log (abs (h));

an=angle (h);

subplot (2, 1, 1);

plot (ph/pi, m);

grid on;

ylabel ('Gain in dB');

xlabel ('Normalised Frquency');

title ('FREQUENCY RESPONSE OF BUTTERWORTH LOWPASS FILTER');

subplot (2, 1, 2);

plot (ph/pi, an);

grid on;

ylabel ('Phase in Radians');

xlabel ('Normalisd Frequency');

title ('PHASE RESPONSE OF BUTTERWORTH LOWPASS FILTER');

Output:

Enter the pass band attenuation=0.4

Enter the stop band attenuation=60

Enter the pass band frequency=400

Enter the stop band frequency=800

Enter the sampling frequency=2000

n = 12

Wn = 0.4499

b = 0.0003 0.0040 0.0220 0.0735 0.1653 0.2644 0.3085

0.2644

0.1653 0.0735 0.0220 0.0040 0.0003

a = 1.0000 -1.1997 2.2442 -1.7386 1.5475 -0.7945 0.4106

-0.1362

0.0412 -0.0080 0.0013 -0.0001 0.0000

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DIGITAL SIGNAL PROCESSING LAB 66

Result: Butterworth analog low pass filter is designed and its frequency response

is found.

Inference: The magnitude response of the butterworth filter decreases

monotonically as the frequency increases.

Questions & Answers

1. Give any two properties of Butterworth lowpass filters?

• The magnitude response of the Butterworth filter decreases monotonically as

the frequency Ω increases from 0 to∞.

• The magnitude response of the Butterworth filter closely approximates the

ideal response as the order N increases.

• The poles of the Butterworth filter lies on a circle

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DIGITAL SIGNAL PROCESSING LAB 67

b)

Aim: To find frequency response of Chebyshew type I analog high pass Filter.

Theory: There are two types of Chebyshev filters. Type I Chebyshev filters are all-

pole filters that exhibits equiripple behaviour in the passband and a monotonic

characteristics in the stopband. On the other hand, the family of type II Chebyshev

filter contains both poles and zeros and exhibits a monotonic behaviour in the

passband and equiripple behaviour in the stopband.

The magnitude square response of N

th

order type I filter can be expressed as

|H(jΩ) |

2

= 1/1+є

2

C

2

N

[Ω/Ω

p

] N = 1, 2,----

Where є is a parameter of the filter related to the ripple in the passband and C

N

(x) is

the N

th

order Chebyshev polynomial defined as

C

N

(x) = cos (N cos

-1

x), |x| ≤ 1 (passband)

And

C

N

(x) = cosh (N cosh

-1

x), |x| > 1 (stopband)

Steps to design an analog Chebyshev lowpass filter

1. From the given specifications find the order of the filter N.

2. Round off it to the next higher integer.

3. Using the following formulas find the value of a and b, which are minor and

major axis of the ellipse respectively.

a = Ω

p

[µ

1/N

- µ

-1/N

]

2 ;

b=Ω

p

[µ

1/N

- µ

-1/N]

2

Where µ = є

-1

+ є

-2

+ 1

Є = 10

0.1

p

α

- 1

Ω

p =

passband frequency

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DIGITAL SIGNAL PROCESSING LAB 68

p

α =

Maximum allowable attenuation in the passband

(For normalized Chebyshev filter Ω

p

= 1 rad/sec)

4. Calculate the poles of Chebyshev filter which lies on an ellipse by using the

formula.

S

k

= a cosФ

k

+ jb sin Ф

k

k= 1, 2, .N

Where Ф

k

= Π/2 + [2k-1/2N] Π k= 1, 2, .N

5. Find the denominator polynominal of the transfer function using above poles.

6. The numerator of the transfer function depends on the value of N.

a) For N odd substitute s = 0 in the denominator polynominal and find

the value. This value is equal to the numerator of the transfer

function.

(For N odd the magnitude response |H(jΩ)| starts at 1.)

b) For N even substitute s=0 in the denominator polynominal and divide

the result by 1+ Є

2

. This value is equal to the numerator.

The order N of Chebyshev filter is given by

N= cos h

-1

[(10

0.1

s

α

-1)/ (10

0.1

p

α

-1)]/cos h

-1

(

Ωs/ Ωp

)

Where

s

α

is stopband attenuation at stopband frequency

Ωs

and

p

α

is passband

attenuation at passband frequency

Ωp

. The major minor axis of the ellipse is given

by

b=

Ωp

[(µ

1/N

+ µ

-1/N

)/2] and a =

Ωp

[ (µ

1/N

- µ

-1/N

)/2]

Where µ = Є

-1

+ 1+ Є

-2

And Є = (10

0.1

p

α

-1)

Program:

%Design of IIR Chebyshew type I high pass filter

alphap=input ('enter the pass band attenuation=');

alphas=input ('enter the stop band attenuation=');

wp=input ('enter the pass band frequency=');

ws=input ('enter the stop band frequency=');

[n, wn]=cheb1ord (wp/pi, ws/pi, alphap, alphas,'s')

[b, a]=cheby1 (n, alphap, wn,'high')

w=0:0.01: pi;

[h, ph]=freqz (b, a, w);

m=20*log (abs (h));

an=angle (h);

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DIGITAL SIGNAL PROCESSING LAB 69

subplot (2, 1, 1);

plot (ph/pi, m);

grid on;

ylabel ('Gain in dB');

xlabel ('Normalised Frquency');

title ('FREQUENCY RESPONSE OF CHEBYSHEW TYPE I HIGHPASS

FILTER');

subplot (2, 1, 2);

plot (ph/pi, an);

grid on;

ylabel ('Phase in Radians');

xlabel ('Normalisd Frequency');

title ('PHASE RESPONSE OF CHEBYSHEW TYPE I HIGHPASS FILTER');

Output:

Enter the pass band attenuation=1

Enter the stop band attenuation=15

Enter the pass band frequency=0.2*pi

Enter the stop band frequency=0.1*pi

n =3

wn = 0.2000

b =0.4759 -1.4278 1.4278 -0.4759

a = 1.0000 -1.6168 1.0366 -0.1540

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DIGITAL SIGNAL PROCESSING LAB 70

Result: Chebyshew type I analog high pass filter is designed and its frequency

response is observed.

Inference: Chebyshev approximation provides better characteristics near the

cutoff frequency and near the stop band edge.

Questions & Answers

1. How one can design digital filters from analog filters?

• Map the desired digital filter specifications into those for an equivalent analog

filter.

• Derive the analog transfer function for the analog prototype.

• Transform the transfer function of the analog prototype into an equivalent

digital filter transfer function.

2. What are the properties of Chebyshev filter?

• The magnitude response of the Chebyshev filter exhibits ripple either in

passband or in stopband according to type.

• The poles of the Chebyshev filter lies on ellipse.

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DIGITAL SIGNAL PROCESSING LAB 71

9. POWER DENSITY SPECTRUM OF A SEQUENCE

Aim: To compute power density spectrum of a sequence.

Theory: PSD Power Spectral Density estimate.

Pxx = PSD(X, NFFT, Fs, WINDOW) estimates the Power Spectral Density of

a discrete-time signal vector X using Welch's averaged, modifiedperiodogram

method.X is divided into overlapping sections, each of which is detrended (according

to the detrending flag, if specified), then windowed by the WINDOW parameter, then

zero-padded to length NFFT. The magnitude squared of the length NFFT DFTs of

the sections are averaged to form Pxx. Pxx is length NFFT/2+1 for NFFT even,

(NFFT+1)/2 for NFFT odd, or NFFT if the signal X is complex. If you specify a scalar

for WINDOW, a Hanning window of that length is used. Fs is the sampling frequency

which doesn't affect the spectrum estimate but is used for scaling the X-axis of the

plots. [Pxx, F] = PSD(X, NFFT, Fs, WINDOW, NOVERLAP) returns a vector of

frequencies the same size as Pxx at which the PSD is estimated, and overlaps

the sections of X by NOVERLAP samples. [Pxx, Pxxc, F] = PSD(X, NFFT, Fs,

WINDOW, NOVERLAP, P) where P is a scalar between 0 and 1, returns the P*100%

confidence interval for Pxx. PSD(X, DFLAG), where DFLAG can be 'linear', 'mean' or

'none', specifies a detrending mode for the prewindowed sections of X. DFLAG can

take the place of any parameter in the parameter list (besides X) as long as it is last,

e.g. PSD(X,'mean'); PSD with no output arguments plots the PSD in the current

figure window, with confidence intervals if you provide the P parameter. The default

values for the parameters are NFFT = 256 (or LENGTH(X), whichever is smaller),

NOVERLAP = 0, WINDOW = HANNING (NFFT), Fs = 2, P = .95, and DFLAG =

'none'. You can obtain a default parameter by leaving it off or inserting an empty

matrix [], e.g. PSD(X, [], 10000).

Program:

%Power density spectrum of a sequence

x=input ('enter the length of the sequence=');

y=input ('enter the sequence=');

fs=input ('enter sampling frequency=');

N=input ('enter the value of N=');

q=input ('enter the window name1=');

m=input ('enter the window name2=');

s=input ('enter the window name3=');

pxx1=psd(y, N, fs, q)

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plot (pxx1,'c');

hold on;

grid on;

pxx2=psd(y, N, fs, m)

plot (pxx2,'k');

hold on;

pxx3=psd(y, N, fs, s)

plot (pxx3,'b');

hold on;

xlabel ('-->Frequency');

ylabel ('-->Magnitude');

title ('POWER SPECTRAL DENSITY OF A SEQUENCE');

hold off;

Output:

Enter the length of the sequence=5

Enter the sequence= [1 2 3 4 5]

Enter sampling frequency=20000

Enter the value of N=4

Enter the window name1=rectwin (N)

Enter the window name2=Kaiser (N, 4.5)

Enter the window name3=triang (N)

pxx1 =25.0000

2.0000

1.0000

pxx2 =14.2828

5.7258

0.3065

pxx3 = 20.0000

3.4000

0.0000

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DIGITAL SIGNAL PROCESSING LAB 73

Result: Power density spectrum of a sequence is computed.

Inference: Power spectral density does not have any phase information.

Questions & Answers

1. Define power spectral density?

a. The power spectral density is a measure of how the power in a signal changes

over frequency.

2. What is window function? What are the applications of window functions?

a. In signal processing, a window function is a function that is zero-valued outside of

some chosen interval. For instance, a function that is constant inside the interval

and zero elsewhere is called a rectangular window, which describes the shape of

its graphical representation. When another function or a signal (data) is multiplied

by a window function, the product is also zero-valued outside the interval.

Applications of window functions include spectral analysis, filter design and beam

forming.

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DIGITAL SIGNAL PROCESSING LAB 74

10. FFT OF A GIVEN 1-D SIGNAL

Aim: To find the FFT of a given 1-D signal and plot the characteristics.

Theory: The DFT of a sequence can be evaluated using the formula

1 - N k 0 ) ( ) (

1

0

/ 2

≤ ≤ =

∑

−

=

−

N

n

N nk j

e n x k X

π

Substituting W

N

=

N j

e

/ 2π −

, we have

1 - N k 0 ) ( ) (

1

0

≤ ≤ =

∑

−

=

N

n

nk

N

W n x k X

The Fast Fourier transform is a highly efficient procedure for computing the DFT of a

finite series and requires less no of computations than that of direct evaluation of

DFT. It reduces the computations by taking the advantage of the fact that the

calculation of the coefficients of the DFT can be carried out iteratively. Due to this,

FFT computation technique is used in digital spectral analysis, filter simulation,

autocorrelation and pattern recognition. The FFT is based on decomposition and

breaking the transform into smaller transforms and combining them to get the total

transform. FFT reduces the computation time required to compute a discrete Fourier

transform and improves the performance by a factor 100 or more over direct

evaluation of the DFT. The fast fourier transform algorithms exploit the two basic

properties of the twiddle factor (symmetry property:

k

N

N k

N

w w − =

+ 2 /

,

periodicity property:

k

N

N k

N

w w =

+

) and reduces the number of complex

multiplications required to perform DFT from N

2

to N/2 log

2

N. In other words, for

N=1024, this implies about 5000 instead of 10

6

multiplications – a reduction factor of

200. FFT algorithms are based on the fundamental principal of decomposing the

computation of discrete fourier transform of a sequence of length N into successively

smaller discrete fourier transform of a sequence of length N into successively smaller

discrete fourier transforms. There are basically two classes of FFT algorithms. They

are decimation-in-time and decimation-in-frequency. In decimation-in-time, the

sequence for which we need the DFT is successively divided into smaller sequences

and the DFTs of these subsequences are combined in a certain pattern to obtain the

required DFT of the entire sequence. In the decimation-in-frequency approach, the

frequency samples of the DFT are decomposed into smaller and smaller

subsequences in a similar manner.

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DIGITAL SIGNAL PROCESSING LAB 75

Radix – 2 DIT – FFT algorithm

11. The number of input samples N=2

M

, where, M is an integer.

12. The input sequence is shuffled through bit-reversal.

13. The number of stages in the flow graph is given by M=log

2

N.

14. Each stage consists of N/2 butterflies.

15. Inputs/outputs for each butterfly are separated by 2

m-1

samples, where m

represents the stage index, i.e., for first stage m=1 and for second stage m=2

so on.

16. The no of complex multiplications is given by N/2 log

2

N.

17. The no of complex additions is given by N log

2

N.

18. The twiddle factor exponents are a function of the stage index m and is given

by . 1 2 ,... 2 , 1 , 0

2

1

− = =

− m

m

t

t

N

k

19. The no of sets or sections of butterflies in each stage is given by the formula

2

M-m.

20. The exponent repeat factor (ERF), which is the number of times the exponent

sequence associated with m is repeated is given by 2

M-m.

Radix – 2 DIF – FFT Algorithm

11. The number of input samples N=2

M

, where M is number of stages.

12. The input sequence is in natural order.

13. The number of stages in the flow graph is given by M=log

2

N.

14. Each stage consists of N/2 butterflies.

15. Inputs/outputs for each butterfly are separated by 2

M-m

samples, Where m

represents the stage index i.e., for first stage m=1 and for second stage m=2

so on.

16. The number of complex multiplications is given by N/2 log

2

N.

17. The number of complex additions is given by N log

2

N.

18. The twiddle factor exponents are a function of the stage index m and is given

by . 1 2 ,... 2 , 1 , 0 ,

2

1

− = =

−

+ −

m m

m M

t

t

N

k

19. The number of sets or sections of butterflies in each stage is given by the

formula 2

m-1.

20. The exponent repeat factor (ERF),Which is the number of times the exponent

sequence associated with m repeated is given by 2

m-1

.

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DIGITAL SIGNAL PROCESSING LAB 76

Program:

%To find the FFT of a given 1-D signal and plot

N=input ('enter the length of the sequence=');

M=input ('enter the length of the DFT=');

u=input ('enter the sequence u (n) =');

U=fft (u, M)

A=length (U)

t=0:1: N-1;

subplot (2, 2, 1);

stem (t, u);

title ('ORIGINAL TIME DOMAIN SEQUENCE');

xlabel ('---->n');

ylabel ('-->Amplitude');

subplot (2, 2, 2);

k=0:1:A-1;

stem (k, abs (U));

disp (‘abs (U) =');

disp (abs (U))

title ('MAGNITUDE OF DFT SAMPLES');

xlabel ('---->k');

ylabel ('-->Amplitude');

subplot (2, 1, 2);

stem (k, angle (U));

disp (‘angle (U) =')

disp (angle (U))

title ('PHASE OF DFT SAMPLES');

xlabel ('---->k');

ylabel ('-->phase');

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DIGITAL SIGNAL PROCESSING LAB 77

Output:

Enter the length of the sequence=3

Enter the length of the DFT=3

Enter the sequence u (n) = [1 2 3]

U = 6.0000 -1.5000 + 0.8660i -1.5000 - 0.8660i

A = 3

abs (U) = 6.0000 1.7321 1.7321

angle (U) = 0 2.6180 -2.6180

Result: FFT of a given 1-D signal is determined and its magnitude and phase

plots are plotted.

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Inference: FFT reduces the computation time required to compute discrete

Fourier transform.

Questions & Answers

1. What is the FFT?

a. The Fast Fourier Transform (FFT) is simply a fast (computationally efficient) way to

calculate the Discrete Fourier Transform (DFT).

2. How does the FFT work?

a. By making use of periodicities in the sines that are multiplied to do the transforms,

the FFT greatly reduces the amount of calculation required. Functionally, the FFT

decomposes the set of data to be transformed into a series of smaller data sets to be

transformed. Then, it decomposes those smaller sets into even smaller sets. At each

stage of processing, the results of the previous stage are combined in special way.

Finally, it calculates the DFT of each small data set. For example, an FFT of size 32

is broken into 2 FFT's of size 16, which are broken into 4 FFT's of size 8, which are

broken into 8 FFT's of size 4, which are broken into 16 FFT's of size 2. Calculating a

DFT of size 2 is trivial. Here’s a slightly more rigorous explanation: It turns out that it

is possible to take the DFT of the first N/2 points and combine them in a special way

with the DFT of the second N/2 points to produce a single N-point DFT. Each of

these N/2-point DFTs can be calculated using smaller DFTs in the same way. One

(radix-2) FFT begins, therefore, by calculating N/2 2-point DFTs. These are

combined to form N/4 4-point DFTs. The next stage produces N/8 8-point DFTs, and

so on, until a single N-point DFT is produced.

3. How efficient is the FFT?

a. The DFT takes N^2 operations for N points. Since at any stage the computation

required to combine smaller DFTs into larger DFTs is proportional to N, and there are

log2 (N) stages (for radix 2), the total computation is proportional to N * log2 (N).

Therefore, the ratio between a DFT computation and an FFT computation for the

same N is proportional to N / log2 (n). In cases where N is small this ratio is not very

significant, but when N becomes large, this ratio gets very large. (Every time you

double N, the numerator doubles, but the denominator only increases by 1.)

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4. Are FFT's limited to sizes that are powers of 2?

a. No, The most common and familiar FFT's are "radix 2". However, other radices are

sometimes used, which are usually small numbers less than 10. For example, radix-4

is especially attractive because the "twiddle factors" are all 1, -1, j, or -j, which can be

applied without any multiplications at all. Also, "mixed radix" FFT's also can be done

on "composite" sizes. In this case, you break a non-prime size down into its prime

factors, and do an FFT whose stages use those factors. For example, an FFT of size

1000 might be done in six stages using radices of 2 and 5, since 1000 = 2 * 2 * 2 * 5

* 5 * 5. It might also be done in three stages using radix 10, since 1000 = 10 *10 *10.

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DIGITAL SIGNAL PROCESSING LAB 80

OTHER EXPERIMENTS

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DIGITAL SIGNAL PROCESSING LAB 81

1. GENERATION OF BASIC SEQUENCES

Aim: To generate impulse, step and sinusoidal sequences.

Program:

%Generation of basic sequences

%generation of impulse sequence

t=-2:1:2;

v= [zeros (1, 2), ones (1, 1), zeros (1, 2)]

subplot (2, 2, 1);

stem (t, v,'m','filled');

grid on;

ylabel ('---->Amplitude');

xlabel ('--->n');

title (' IMPULSE SEQUENCE');

%generation of step sequence

t=0:1:5;

m= [ones (1, 6)]

subplot (2, 2, 2);

stem (t, m,'m','filled');

grid on;

ylabel ('---->Amplitude');

xlabel ('--->n');

title (' STEP SEQUENCE');

%generation of sinusoidal sequence

t=-4:1:4;

y=sin (t)

subplot (2, 1, 2);

stem (t, y,'m','filled');

grid on;

ylabel ('---->Amplitude');

xlabel ('--->n');

title ('SINUSOIDAL SEQUENCE');

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DIGITAL SIGNAL PROCESSING LAB 82

Output:

v = 0 0 1 0 0

m = 1 1 1 1 1 1

y= 0.7568 -0.1411 -0.9093 -0.8415 0 0.8415 0.9093

0.1411

-0.7568

Result: Impulse, step and sinusoidal sequences are generated.

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2. DFT OF A SEQUENCE

Aim: To compute the DFT of a sequence.

Program:

%DFT of a sequence

N=input ('enter the length of the sequence=');

x=input ('enter the sequence=');

n= [0:1: N-1];

k= [0:1: N-1];

wN=exp (-j*2*pi/N);

nk=n'*k;

wNnk=wN. ^nk;

Xk=x*wNnk;

disp ('Xk=');

disp (Xk);

mag=abs (Xk)

subplot (2, 1, 1);

stem(k,mag);

grid on;

xlabel('--->k');

title ('MAGNITUDE OF FOURIER TRANSFORM');

ylabel ('Magnitude');

phase=angle (Xk)

subplot (2, 1, 2);

stem (k, phase);

grid on;

xlabel ('--->k');

title ('PHASE OF FOURIER TRANSFORM');

ylabel ('Phase');

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DIGITAL SIGNAL PROCESSING LAB 84

Output:

Enter the length of the sequence=5

Enter the sequence= [1 2 3 4 5]

Xk= 15.0000 -2.5000 + 3.4410i -2.5000 + 0.8123i -2.5000 –

0.8123i

-2.5000 - 3.4410i

mag = 15.0000 4.2533 2.6287 2.6287 4.2533

phase = 0 2.1991 2.8274 -2.8274 -2.1991

Result: DFT of a sequence is computed.

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3. IDFT OF A SEQUENCE

Aim: To compute the IDFT of a sequence.

Program:

%IDFT of a sequence

Xk=input (‘enter X (K) =');

[N, M]=size (Xk);

if M~=1;

Xk=Xk.';

N=M;

end;

xn=zeros (N, 1);

k=0: N-1;

for n=0: N-1;

xn (n+1) =exp (j*2*pi*k*n/N)*Xk;

end;

xn=xn/N;

disp (‘x (n) = ');

disp (xn);

plot (xn);

grid on;

plot(xn);

stem(k,xn);

xlabel ('--->n');

ylabel ('-->magnitude');

title (‘IDFT OF A SEQUENCE’);

Output:

Enter the dft of X (K) = [6 -1.5+0.86i -1.5-0.86i]

x (n) =1.0000

2.0035 + 0.0000i

2.9965 - 0.0000i

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DIGITAL SIGNAL PROCESSING LAB 86

Result: IDFT of a given sequence is computed.

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DIGITAL SIGNAL PROCESSING LAB 87

CODE COMPOSER STUDIO

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DIGITAL SIGNAL PROCESSING LAB 88

TEXAS INSTRUMENT DSP STARTER KIT 6711DSK

A BRIEF INTRODUCTION

1 Introduction

The purpose of this documentation is to give a very brief overview of the structure of

the 6711DSK. Included is also a mini tutorial on how to program the DSP, download

programs and debug it using the Code Composer Studio software.

2 The DSP starter kit

The Texas Instrument 6711DSK is a DSP starter kit for the TMS320C6711 DSP chip.

The kit contains:

• An emulator board which contains:

– DSP chip, memory and control circuitry.

– Input/output capabilities in the form of

_ An audio codec (ADC and DAC) which provides 1 input and 1 output

channel sampled at 8 kHz.

_ Digital inputs and outputs

_ A connector for adding external evaluation modules, like the PCM3003

Audio daughter card which has 2 analog in and 2 analog out sampled at

48 kHz.

– A parallel port for interface to a host computer used for program

development, program download and debugging.

• The Code Composer Studio (CCS) software which is an integrated development

environment (IDE) for editing programs, compiling, linking, download to target

(i.e., to the DSK board) and debugging. The CCS also includes the DSP/BIOS

real-time operating system. The DSP/BIOS code considerably simplifies the code

development for real-time applications which include interrupt driven and time

scheduled tasks.

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3 Usage overview

Working with the DSK involves the following steps:

• The first step is the algorithm development and programming in C/C++ or

assembler. We will only consider coding in C in this document. The program is typed

using the editor capabilities of the CCS. Also the DSP/BIOS configuration is

performed in a special configuration window.

• When the code is finished it is time to compile and link the different code parts

together. This is all done automatically in CCS after pressing the Build button. If

the compilation and linking succeeded the finished program is downloaded to the

DSK board.

• The operation of DSK board can be controlled using CCS, i.e., CCS also works as

a debugger. The program can be started and stopped and single-step. Variables in

the program can be inspected with the “Watch” functionality. Breakpoints can be

inserted in the code.

4 Getting started with the 6711DSK and CCS

In this section you will be introduced to the programming environment and will

download and execute a simple program on the DSK.

4.1 Connecting the PC and DSK

The DSK is communicating with the host PC using the parallel-port interface.

1. Start Code Composer Studio (CCS) by clicking on the “CCStudio 3.01” icon on the

workspace.

2. Check that the DSK is powered up. If not, power up the DSK by inserting the

AC power cable (mains) to the black AC/DC converter. Note: Never unplug the

power cable from the DSK card.

3. Make a connection between CCS and the DSK by selecting menu command

Debug! Connect. If connection fails perform following steps:

(a) Select menu command Debug! Reset Emulator

(b) Power cycle the DSK by removing the AC power cord from the AC/DC converter,

wait about a second and then reinsert.

(c) Select menu command Debug! Connect.

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4.2 Getting familiar with CCS

CCS arranges its operation around a project and the “Project view” window is

displayed to the left. Here you can find all files which are related to a certain project.

At the bottom of the CCS window is the status output view in which status information

will be displayed such as compile or link errors as well as other types of information.

Note that the window will have several tabs for different types of output.

4.3 Loading programs from CCS

One setting might need to be changed in CCS in order to automatically download the

program after compilation and linking (a build). You find the setting under menu

command Options! Customize and tab Program Load Options. Tick the box Load

Program after Build.

4.3.1 Project view and building an application

You open an existing project by right-clicking on the project folder and select the

desired “.pjt” file or select the command from the menu Project! Open....

• Open the project “intro.pjt” which you find in the “intro”folder.

• Press the “+” sign to expand the view. Several folders appear which contain all

the project specific files.

• Open up the Source folder. Here you will find all the files with source code which

belongs to the project.

• Double click on the “intro.c” file to open up an editor window with the file “intro.c” file

loaded.

• Look around in the file. The code calculates the inner product between two integer

vectors of length “COUNT” by calling the function “dotp” and then prints the result

using the “LOG printf” command.

• Compile, link and download the code selecting the Project! Build command or

use the associated button. If the download fails. Power cycle the DSK and retry

Project! Build.

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DIGITAL SIGNAL PROCESSING LAB 91

4.3.2 Running the code after a successful build command (compile, link and

download) the program is now resident in the DSK and ready for execution. A

disassembly window will appear showing the Assembly language instructions

produced by the compiler. You can minimize the window for now. At the bottom left

you will see the status “CPU HALTED” indicating that the CPU is not executing any

code. A few options are available depending on if you want to debug or just run the

code.

To simply run the code does:

• Select Debug! Run to start the execution. At the bottom left you will see the status

“CPU RUNNING” indicating that the program is executed.

• To stop the DSP select Debug! Halt.

• If you want the restart the program do Debug! Restart followed by Debug! Run

Try to run the code in the intro project. To see the output from the LOG printf

commands you must enable the view of the Message Log Window. Do DSP/BIOS!

Message Log and select LOG0 in the pull-down menu.

4.3.3 CCS Debug

Single stepping for debugging the code do the following:

• Select Debug! Go Main. This will run DSP/BIOS part of the code until the start

of the “main” function. A yellow arrow will appear to the left of the start of the

“main” function in the edit window for the “intro.c” file. The arrow indicates where

the program execution has halted.

• Now you can step the code one line at the time. Three commands are available:

– Debug! Step Into will execute the marked line if it is composed of simple

instructions or if the line has a function call, will step in to the called function.

– Debug! Step Over will execute the marked line and step to the next line.

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DIGITAL SIGNAL PROCESSING LAB 92

– Debug! Step Out will conclude the execution of the present function and halt

the execution at the next line of the calling function. Try to step through the code

using the step functions. View variables if you want to know the value of a particular

variable just put the cursor on top of the variable in the source window. The value will

after a short delay pop up next to the variable. Note that local variables in functions

only have defined values when the execution is located inside the function. Variables

can also be added to the “Watch window” which enables a concurrent view of several

variables. Mark the “y” variable in source code, right-click and select Add to Watch

Window. The watch window

will then add variable “y” to the view and show its contents. Also add the “a” variable

(which is a array) to the watch window. Since “a” is an array the value of “a” is an

address. Click on the “+”. This will show the individual values of the array. The watch

window will update its contents whenever the DSP is halted. It is also possible to

change the value of a variable in the DSK from the Watch window. Simply select the

numerical value you want to change and type in a new value. CCS will send the new

value to the DSK before starting the execution.

Break points Break points can also be used to halt the execution at a particular point

in the source code. To illustrate its use consider the “intro”. Place the cursor at the

line which has the “return (sum);” instruction in the “dotp” function. Right-click and

select Toggle breakpoint. A red dot will appear at that line indicating that a breakpoint

is set at that line. Try it out. First do Debug! Restart and then Debug! Run The

execution will halt at the line and we can investigate variables etc. To resume

execution simply issue Debug! Run command again. Shortcut buttons CCS offers a

number of shortcut buttons both for the build process as well as the debugging.

Browse around with the mouse pointer to find them. The balloon help function will

show what command a particular button is associated with, i.e., move the mouse

pointer over a button and wait for the balloon help to show up. Loading the values in

a vector into Matlab In the Lab you will need to load the coefficients of a FIR filter into

Matlab for further analysis. The filter coefficients are stored in vectors. Use the

following procedure:

1. Run the DSP program on the DSK.

2. Halt the execution when it is time to upload the vector.

3. In the File menu you find File! Data! Save In the file dialog assign a name to

the data file and select ”Float” as ”Save as type”.

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DIGITAL SIGNAL PROCESSING LAB 93

4. In the next dialog enter the name of the vector in the address field. When you tab

away from it it will change to the absolute memory address of the vector. In the

length box you enter the length of the variable. Note that you need to remove the 0x

prefix if you give the length using decimals numbers. The notation with 0x in the

beginning indicates that the number is a Hexadecimal number, i.e. it is based on a

base of 16 instead of the normal base of 10 in the decimal system.

5. Open the file in an editor and remove the first line and save again

6. In Matlab use the ”load” command to load the saved file. The vector will get the

same name as the file name without the extension.

5 Debugging - limitations

The debugging features of the CCS are very handy during the code development.

However, several limitations are present.

• Since the compiler will optimize the code, all lines in the C-code cannot be used as

breakpoints. Since the architecture has several parallel processing units the final

code might execute several lines of C in one clock cycle.

• The processor also has memory cache which speeds up normal operations. This

means that when the DSP loads a particular memory location it will copy it into

the cache memory and then perform the operations (faster) using the cache. After

some time the cache will be written back again to the memory. This can sometimes

confuse CCS and variables in the Watch window can be erroneous.

• The third important issue is when debugging code which is dependent on external

events, e.g., interrupt driven code. When halting such a code by a break point

several features are important to consider. Firstly, since the code is stopped all real-

time deadlines will be broken so future sampled inputs will be lost. Hence a restart

from the point of the break might cause unpredicted behavior.

• Sometimes the code in the DSK which communicates with the CCS stops working.

When this happens it is necessary to power cycle, the DSK board in order to restart

the operation. Sometimes it is also necessary at the same time to restart CCS.

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DIGITAL SIGNAL PROCESSING LAB 94

PROCEDURE TO WORK ON CODE COMPOSER

STUDIO

To create the new project

Project new (File name.pjt, eg: vectors.pjt)

To create a source file

File newtype the code (save &give file name, eg: sum.c

To add source files to project

Project add files to projectsum.c

To add rts.lib file&hello.cmd:

Project add files to project rts6700.lib

Libraryfiles: rts6700.lib (path:c:\ti\c6000\cgtools\lib\rts6700.lib)

Note: select object&library in (*.o,*.1) in type of files

Project add files to projectshello.cmd

Cmd file – which is common for all non real time programs.

(path:c\ti\tutorials\dsk6711\hello.1\hello.cmd)

Note: select linker command file (*.cmd) in type of files

Compile:

To compile: projectcompile

To rebuild: projectrebuild

Which will create the final.outexecutablefile (eg.vectors.out)

Procedure to load and run program:

Load the program to DSK: file load programvectors. Out

To execute project: Debugrun

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DIGITAL SIGNAL PROCESSING LAB 95

C PROGRAMS

(USING64XX, 67XX SIMULATORS)

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DIGITAL SIGNAL PROCESSING LAB 96

1. TO VERIFY LINEAR CONVOLUTION

Aim: Write a C program for linear convolution of two discrete sequences and

compute using code composer studio V2 (using simulators 64xx, 67xx, 6711)

software.

Program:

#include<stdio.h>

#include<math.h>

main ()

{

int x[20],h[20],y[20],N1,N2,n,m;

printf ("enter the length of the sequence x (n) :");

scanf ("%d", &N1);

printf ("enter the length of the sequence h (n) :");

scanf ("%d", &N2);

printf ("enter the sequence x (n) :");

for (n=0; n<N1; n++)

scanf ("%d", &x[n]);

printf ("enter the sequence h (n) :");

for (n=0; n<N2; n++)

scanf ("%d", &h[n]);

for (n=0; n<N1+N2-1; n++)

{

if (n>N1-1)

{

y[n]=0;

for(m=n-(N1-1);m<=(N1-1);m++)

y[n] =y[n] +x[m]*h [n-m];

}

else

{

y[n] =0;

for (m=0; m<=n; m++)

y[n] =y[n] +x[m]*h [n-m];

}

}

printf ("convolution of two sequences is y (n) :");

for (n=0; n<N1+N2-1; n++)

printf ("%d\t", y[n]);

}

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Input:

Enter the length of the sequence x (n):4

Enter the length of the sequence h (n):3

Enter the sequence x (n):1 2 3 4

Enter the sequence h (n):5 6 7

Output:

Convolution of two sequences is y (n):5 16 34 52 45 28

Result: Linear convolution of two discrete sequences is computed.

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DIGITAL SIGNAL PROCESSING LAB 98

2. TO VERIFY CIRCULAR CONVOLUTION

Aim: Write a C program for circular convolution of two discrete sequences and

compute using code composer studio V2 (using simulators 64xx, 67xx, 6711)

software.

Program:

#include<math.h>

#define pi 3.14

main ()

{

int x[20],h[20],y[20],N1,N2,N,n,m,k;

printf ("enter the length of the sequence x (n) :");

scanf ("%d", &N1);

printf ("enter the length of the sequence h (n) :");

scanf ("%d", &N2);

printf ("enter the %d samples of sequence x (n):” N1);

for (n=0; n<N1; n++)

scanf ("%d", &x[n]);

printf ("enter the %d samples of sequence h (n):” N2);

for (n=0; n<N2; n++)

scanf ("%d", &h[n]);

if (N1>N2)

{

N=N1;

for (n=N2; n<N; n++)

h[n]=0;

}

else

{

N=N2;

for(n=N1;n<N1;n++)

y[n]=0;

}

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DIGITAL SIGNAL PROCESSING LAB 99

for(k=0;k<N;k++)

{

y[k]=0;

for (m=N-1; m>=0; m--)

{

if ((N+k-m)>=N)

{

y[k] =y[k] +x[m]*h [k-m];

}

else

y[k] =y[k] +x[m]*h [N+k-m];

}

}

printf ("\response is y (n) :");

for (n=0; n<N; n++)

printf ("\t%d", y[n]);

}

Input:

Enter the length of the sequence x (n):4

Enter the length of the sequence h (n):4

Enter the 4 samples of sequence x (n):2 1 2 1

Enter the 4 samples of sequence h (n):1 2 3 4

Output:

Response is y (n): 14 16 14 16

Result: Circular convolution of two discrete sequences is computed.

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DIGITAL SIGNAL PROCESSING LAB 100

3. DFT OF A SEQUENCE

Aim: Write a C program for DFT of a sequence and compute using code composer

studio V2 (using simulators 64xx, 67xx, 6711) software.

Program:

%DISCRETE FOURIER TRANSFORM OF THE GIVEN SEQUENCE

#include<stdio.h>

#include<math.h>

#define pi 3.14

main ()

{

int x [20], N, n, k;

float Xre[20],Xim[20],Msx[20],Psx[20];

printf ("enter the length of the sequence x (n) :");

scanf ("%d", &N);

printf ("enter the %d samples of discrete sequence x (n):” N);

for (n=0; n<N; n++)

scanf ("%d", &x[n]);

for(k=0;k<N;k++)

{

Xre[k]=0;

Xim[k]=0;

for(n=0;n<N;n++)

{

Xre[k]=Xre[k]+x[n]*cos(2*pi*n*k/N);

Xim[k]=Xim[k]+x[n]*sin(2*pi*n*k/N);

}

Msx[k]=sqrt(Xre[k]*Xre[k]+Xim[k]*Xim[k]);

Xim[k]=-Xim[k];

Psx[k]=atan(Xim[k]/Xre[k]);

}

printf ("discrete sequence is");

for (n=0; n<N; n++)

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 101

printf ("\nx(%d) =%d", n,x[n]);

printf ("\nDFT sequence is :");

for(k=0;k<N;k++)

printf("\nx(%d)=%f+i%f",k,Xre[k],Xim[k]);

printf("\nmagnitude is:");

for (k=0; k<N; k++)

printf ("\n%f",Msx[k]);

printf ("\nphase spectrum is");

for(k=0;k<n;k++)

printf("\n%f",Psx[k]);

}

Input:

Enter the length of the sequence x (n):4

Enter the 4 samples of discrete sequence x (n):1 2 3 4

Output:

Discrete sequence is:

x (0) =1

x (1) =2

x (2) =3

x (3) =4

DFT sequence is :

x (0) =10.000000+i0.000000

x (1) =-2.007959+i1.995211

x (2) =-1.999967+i-0.012741

x (3) =-1.976076+i-2.014237

Magnitude is: Phase spectrum is:

0.000000 10.000000

-0.782214 2.830683

0.006371 2.000007

0.794961 2.821707

Result: DFT of a given sequence is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 102

4. IDFT OF A SEQUENCE

Aim: Write a C program for IDFT of a sequence and compute using code composer

studio V2 (using simulators 64xx, 67xx, 6711) software.

Program:

% Inverse Discrete Foureir Transform

#include<stdio.h>

#include<math.h>

#define pi 3.1428

main()

{

int j, k, N, n;

float XKr[10],XKi[10],xni[10],xnr[10],t;

printf (" enter seq length: \n");

scanf ("%d", &N);

printf ("enter the real & imaginary terms of the sequence: \n");

for (j=0; j<N; j++)

{

scanf ("%f%f", &XKr[j], &XKi[j]);

}

for (n=0; n<N; n++)

{

xnr[n] =0; xni[n] =0;

for(k=0;k<N;k++)

{

t=2*pi*k*n/N;

xnr[n]=( xnr[n]+XKr[k]*cos(t) - XKi[k]*sin(t)) ;

xni[n]=( xni[n]+XKr[k]*sin(t) + XKi[k]*cos(t)) ;

}

xnr[n]=xnr[n]/N;

xni[n]=xni[n]/N;

}

printf("IDFT seq:");

for(j=0;j<N;j++)

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 103

{

printf("%f + (%f) i\n",xnr[j],xni[j]);

}

}

Input:

Enter seq length: 4

Enter the real & imaginary terms of the sequence:

3

0

0

-1

1

0

0

1

Output:

IDFT seq: 1.000000 + (0.000000) i

1.000000 + (0.000302) i

1.000603 + (0.000605) i

0.000006 + (-0.002717) i

Result: IDFT of a given sequence is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 104

5. N - POINT DISCRETE FOUREIR TRANSFORM

Aim: Write a C program for N - Point Discrete Foureir Transform of a sequence

and compute using code composer studio V2( using simulators 64xx,67xx,6711)

software.

Program:

% N - Point Discrete Foureir Transform

#include<stdio.h>

#include<math.h>

#define pi 3.142857

main ()

{

int j, k, n, N, l;

float XKr[10],XKi[10],xni[10],xnr[10],t;

printf (" enter seq length: \n");

scanf ("%d", &l);

printf ("enter the real & imaginary terms of the sequence: \n");

for (j=0; j<l; j++)

{

scanf ("%f%f", &xnr[j], &xni[j]);

}

printf (" enter required length: \n");

scanf ("%d", &N);

if (l<N)

{

for(j=l;j<N;j++)

xnr[j]=xni[j]=0;

}

for(k=0;k<N;k++)

{

XKr[k]=0; XKi[k]=0;

for(n=0;n<N;n++)

{

t=2*pi*k*n/N;

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 105

XKr[k]=( XKr[k]+xnr[n]*cos(t) + xni[n]*sin(t)) ;

XKi[k]=( XKi[k]+ xni[n]*cos(t) - xnr[n]*sin(t)) ;

}

}

printf (" The %d Point DFT seq:” N);

for(j=0;j<N;j++)

{

printf("%3.3f + (%3.3f) i\n",XKr[j],XKi[j]);

}

}

Input:

Enter seq length:

3

Enter the real & imaginary terms of the sequence:

1

0

1

0

1

0

enter required length:

4

Out put:

The 4 Point DFT seq:3.000 + (0.000) i

-0.001 + (-0.999) i

1.000 + (-0.001) i

0.002 + (1.004) i

Result: 4 - Point Discrete Foureir Transform of a sequence is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 106

6. TO GENERATE SUM OF SINUSOIDAL SIGNALS

Aim: Write a C program to generate sum of sinusoidal signals and compute using

code composer studio V2 (using simulators 64xx, 67xx, 6711) software.

Program:

%SUM OF TWO SINUSOIDALS

#include<stdio.h>

#include<math.h>

main ()

{

int w1, w2, t;

float a, b, c;

printf ("enter the w1 value:");

scanf ("%d", &w1);

printf ("enter the w2 value:");

scanf ("%d", &w2);

printf ("sum of two sinusoidal :");

for(t=-5;t<5;t++)

{

b=sin(w1*t);

c=sin (w2*t);

a=b+c;

printf ("\n%f", a);

}

}

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 107

Input:

Enter the w1 value: 5

Enter the w2 value: 5

Output:

Sum of two sinusoidal:

0.264704

-1.825891

-1.300576

1.088042

1.917849

0.000000

-1.917849

-1.088042

1.300576

1.825891

Result: samples of sum of sinusoidal signals are generated.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 108

PROGRAMS USING TMS 320C 6711 DSK

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 109

1. TO VERIFY LINEAR CONVOLUTION

Aim: Write a C program for linear convolution of two discrete sequences and

compute using TMS 320C 6711 DSK.

Program:

#include<stdio.h>

#include<math.h>

main ()

{

int x[20],h[20],y[20],N1,N2,n,m;

printf ("enter the length of the sequence x (n) :");

scanf ("%d", &N1);

printf ("enter the length of the sequence h (n) :");

scanf ("%d", &N2);

printf ("enter the sequence x (n) :");

for (n=0; n<N1; n++)

scanf ("%d", &x[n]);

printf ("enter the sequence h (n) :");

for (n=0; n<N2; n++)

scanf ("%d", &h[n]);

for (n=0; n<N1+N2-1; n++)

{

if (n>N1-1)

{

y[n]=0;

for(m=n-(N1-1);m<=(N1-1);m++)

y[n] =y[n] +x[m]*h [n-m];

}

else

{

y[n] =0;

for (m=0; m<=n; m++)

y[n] =y[n] +x[m]*h [n-m];

}

}

printf ("convolution of two sequences is y (n) :");

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 110

for (n=0; n<N1+N2-1; n++)

printf ("%d\t", y[n]);

}

Input:

Enter the length of the sequence x (n):4

Enter the length of the sequence h (n):3

Enter the sequence x (n):1 2 3 4

Enter the sequence h (n):5 6 7

Output:

Convolution of two sequences is y (n):5 16 34 52 45 28

Result: Linear convolution of two discrete sequences is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 111

2. TO VERIFY CIRCULAR CONVOLUTION

Aim: Write a C program for circular convolution of two discrete sequences and

compute using using TMS 320C 6711 DSK.

Program:

#include<math.h>

#define pi 3.14

main ()

{

int x[20],h[20],y[20],N1,N2,N,n,m,k;

printf ("enter the length of the sequence x (n) :");

scanf ("%d", &N1);

printf ("enter the length of the sequence h (n) :");

scanf ("%d", &N2);

printf ("enter the %d samples of sequence x (n):", N1);

for (n=0; n<N1; n++)

scanf ("%d", &x[n]);

printf ("enter the %d samples of sequence h (n):” N2);

for (n=0; n<N2; n++)

scanf ("%d", &h[n]);

if (N1>N2)

{

N=N1;

for (n=N2; n<N; n++)

h[n]=0;

}

else

{

N=N2;

for(n=N1;n<N1;n++)

y[n]=0;

}

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 112

for(k=0;k<N;k++)

{

y[k]=0;

for (m=N-1; m>=0; m--)

{

if ((N+k-m)>=N)

{

y[k] =y[k] +x[m]*h [k-m];

}

else

y[k] =y[k] +x[m]*h [N+k-m];

}

}

printf ("\response is y (n) :");

for (n=0; n<N; n++)

printf ("\t%d", y[n]);

}

Input:

Enter the length of the sequence x (n):4

Enter the length of the sequence h (n):4

Enter the 4 samples of sequence x (n):2 1 2 1

Enter the 4 samples of sequence h (n):1 2 3 4

Output:

Response is y (n): 14 16 14 16

Result: Circular convolution of two discrete sequences is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 113

3. FINDING DFT OF A SEQUENCE

Aim: Write a C program for DFT of a sequence and compute using TMS 320C

6711 DSK.

Program:

%DISCRETE FOURIER TRANSFORM OF THE GIVEN SEQUENCE

#include<stdio.h>

#include<math.h>

#define pi 3.14

main ()

{

int x [20], N, n, k;

float Xre[20],Xim[20],Msx[20],Psx[20];

printf ("enter the length of the sequence x (n) :");

scanf ("%d", &N);

printf ("enter the %d samples of discrete sequence x (n):” N);

for (n=0; n<N; n++)

scanf ("%d", &x[n]);

for(k=0;k<N;k++)

{

Xre[k]=0;

Xim[k]=0;

for(n=0;n<N;n++)

{

Xre[k]=Xre[k]+x[n]*cos(2*pi*n*k/N);

Xim[k]=Xim[k]+x[n]*sin(2*pi*n*k/N);

}

Msx[k]=sqrt(Xre[k]*Xre[k]+Xim[k]*Xim[k]);

Xim[k]=-Xim[k];

Psx[k]=atan(Xim[k]/Xre[k]);

}

printf ("discrete sequence is:");

for (n=0; n<N; n++)

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 114

printf ("\nx (%d) =%d", n, x[n]);

printf ("\nDFT sequence is :");

for(k=0;k<N;k++)

printf("\nx(%d)=%f+i%f",k,Xre[k],Xim[k]);

printf ("\nmagnitude is:");

for (k=0; k<N; k++)

printf ("\n%f", Msx[k]);

printf ("\nphase spectrum is:");

for(k=0;k<n;k++)

printf("\n%f",Psx[k]);

}

Input:

Enter the length of the sequence x (n):4

Enter the 4 samples of discrete sequence x (n):1 2 3 4

Output:

Discrete sequence is:

x (0) =1

x (1) =2

x (2) =3

x (3) =4

DFT sequence is :

x (0) =10.000000+i0.000000

x (1) =-2.007959+i1.995211

x (2) =-1.999967+i-0.012741

x (3) =-1.976076+i-2.014237

Magnitude is: Phase spectrum is:

0.000000 10.000000

-0.782214 2.830683

0.006371 2.000007

0.794961 2.821707

Result: DFT of a given sequence is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 115

4. FINDING IDFT OF A SEQUENCE

Aim: Write a C program for IDFT of a sequence and compute using TMS 320C

6711 DSK.

Program:

% Inverse Discrete Foureir Transform

#include<stdio.h>

#include<math.h>

#define pi 3.1428

main ()

{

int j, k, N, n;

float XKr[10],XKi[10],xni[10],xnr[10],t;

printf (" enter seq length: \n");

scanf ("%d", &N);

printf ("enter the real & imaginary terms of the sequence: \n");

for (j=0; j<N; j++)

{

scanf ("%f%f", &XKr[j], &XKi[j]);

}

for (n=0; n<N; n++)

{

xnr[n] =0; xni[n] =0;

for(k=0;k<N;k++)

{

t=2*pi*k*n/N;

xnr[n]=( xnr[n]+XKr[k]*cos(t) - XKi[k]*sin(t)) ;

xni[n]=( xni[n]+XKr[k]*sin(t) + XKi[k]*cos(t)) ;

}

xnr[n]=xnr[n]/N;

xni[n]=xni[n]/N;

}

printf("IDFT seq:");

for(j=0;j<N;j++)

{

printf("%f + (%f) i\n",xnr[j],xni[j]);

}

}

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 116

Input:

Enter seq length:

4

Enter the real & imaginary terms of the sequence:

3

0

0

-1

1

0

0

1

Output:

IDFT seq: 1.000000 + (0.000000) i

1.000000 + (0.000302) i

1.000603 + (0.000605) i

0.000006 + (-0.002717) i

Result: IDFT of a given sequence is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 117

5. N - POINT DISCRETE FOUREIR TRANSFORM

Aim: Write a C program for N - Point Discrete Foureir Transform of a sequence and

compute using TMS 320C 6711 DSK.

Program:

% N - Point Discrete Foureir Transform

#include<stdio.h>

#include<math.h>

#define pi 3.142857

main ()

{

int j, k, n, N, l;

float XKr[10],XKi[10],xni[10],xnr[10],t;

printf (" enter seq length: \n");

scanf ("%d", &l);

printf ("enter the real & imaginary terms of the sequence: \n");

for (j=0; j<l; j++)

{

scanf ("%f%f", &xnr[j], &xni[j]);

}

Printf (" enter required length: \n");

Scanf ("%d", &N);

if (l<N)

{

for(j=l;j<N;j++)

xnr[j]=xni[j]=0;

}

for(k=0;k<N;k++)

{

XKr[k]=0; XKi[k]=0;

for(n=0;n<N;n++)

{

[

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 118

t=2*pi*k*n/N;

XKr[k]=( XKr[k]+xnr[n]*cos(t) + xni[n]*sin(t)) ;

XKi[k]=( XKi[k]+ xni[n]*cos(t) - xnr[n]*sin(t)) ;

}

}

Printf (" The %d Point DFT seq:” N);

for(j=0;j<N;j++)

{

printf("%3.3f + (%3.3f) i\n",XKr[j],XKi[j]);

}

}

Input:

Enter seq length:

3

Enter the real & imaginary terms of the sequence:

1

0

1

0

1

0

Enter required length:

4

Out put:

The 4 Point DFT seq: 3.000 + (0.000) i

-0.001 + (-0.999) i

1.000 + (-0.001) i

0.002 + (1.004) i

Result: 4 - Point Discrete Foureir Transform of a sequence is computed.

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 119

6. TO GENERATE SUM OF SINUSOIDAL SIGNALS

Aim: Write a C program to generate sum of sinusoidal signals and compute using

TMS 320C 6711 DSK.

Program:

%SUM OF TWO SINUSOIDALS

#include<stdio.h>

#include<math.h>

main ()

{

int w1, w2, t;

float a, b, c;

printf ("enter the w1 value :");

scanf ("%d", &w1);

printf ("enter the w2 value :");

scanf ("%d", &w2);

printf ("sum of two sinusoidal :");

for(t=-5;t<5;t++)

{

b=sin(w1*t);

c=sin (w2*t);

a=b+c;

printf ("\n%f", a);

}

}

ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB 120

Input:

Enter the w1 value: 5

Enter the w2 value: 5

Output:

Sum of two sinusoidal:

0.264704

-1.825891

-1.300576

1.088042

1.917849

0.000000

-1.917849

-1.088042

1.300576

1.825891

Result: Samples of sum of sinusoidal signals are obtained.

ELECTRONICS & COMMUNICATION ENGINEERING

121

REFERENCES

1. Rabiner, Lawrence R. & Gold, Bernard, Theory and application of digital

signal processing, Prentice-Hall (1975).

2. Proakis, J.G. and Manolakis, D.G., Digital signal processing: principles,

algorithms and applications, Macmillan (1996).

3. A. V. Oppenheim and R. W. Schafer, Digital Signal Processing, Prentice-Hall,

(1975).

4. Sanjit K.Mitra, Digital Signal Processing, A Computer based approach, Tata

McGraw-Hill, (1998).

5. Vinay K. Ingle and John G. Proakis, Digital Signal Processing using

MATLAB, BookWare Companion Series,(2001).

6. Rudra Pratap, Getting Started With MATLAB: A Quick Introduction for

Scientists and Engineers, Oxford University Press.

7. Avtar Singh and S.Srinivasan, Digital Signal Processing Implementations,

Penram International Publishing (India) Pvt. Ltd.

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