DSP LAB MANUAL

For

5TH SEMESTER ELECTRONICS AND

COMMUNICATION
ENGINEERING
 CONTENTS

PART-A. LIST OF EXPERIMENTS USING MATLAB

1. Verification of sampling theorem.
2. Impulse response of a given system.
3. Linear convolution of two given sequences.
4. Circular convolution of two given sequences.
5. Autocorrelation of given sequence and verification of its properties.
6. Cross correlation of a given sequences and verification of its properties.
7. Solving a given difference equation.
8. Computation of N-point DFT of a given sequence and to plot magnitude and
phase spectrum.
9. Linear convolution of two sequences using DFT and IDFT.
10. Circular convolution of two sequences using DFT and IDFT.
11. Design and implementation of FIR filter to meet given specifications.
12. Design and implementation of IIR filter to meet given specifications.

PART-B. LIST OF EXPERIMENTS USING DSP PROCESSOR.

1. Linear convolution of two given sequences.
2. Circular convolution of two given sequences.
3. Computation of N-point DFT of a given sequence.
4. Realization of an FIR filter (any type) to meet given specifications. The input
can be a signal from function generator/speech signal.
5. Audio application such as to plot a time and frequency display of a Microphone
plus a cosine using DSP. Read a .wav file and match with their respective
spectrograms.
6. Noise removal: Add noise above 3KHz and then remove, interference
suppression using 400Hz tone.
7. Impulse response of first order and second order system.
 EXPERIMENT NO-1

AIM: VERIFICATION OF SAMPLING THEOREM

Sampling: Is the process of converting a continuous time signal into a discrete time
signal. It is the first step in conversion from analog signal to digital signal.

Sampling theorem: Sampling theorem states that “Exact reconstruction of a continuous

time base-band signal from its samples is possible, if the signal is band-limited and the
sampling frequency is greater than twice the signal bandwidth”.
i.e. fs > 2W, where W is the signal bandwidth.

Nyquist Rate Sampling: The Nyquist rate is the minimum sampling rate required to
avoid aliasing, equal to the highest modulating frequency contained within the signal.
In other words, Nyquist rate is equal to two sided bandwidth of the signal (Upper and
lower sidebands)
To avoid aliasing, the sampling rate must exceed the Nyquist rate. i.e. fs > fN
Program:
% Nyquist Rate Sampling.
clc; % Clear screen
fs = 1400; % Sampling frequency, fs = 1400Hz
t = 0:1/fs:13/fs; % Number of samples
x = cos(2*pi*400*t)+cos(2*pi*700*t); % Input signal
xm = abs (fft(x)); % Determine the absolute FFT of the input signal
disp(‘xm’); % Displays xm on command window
disp (xm); % Displays values of xm on command window
k = 0:length(xm)-1; % Number of samples to be plot on x-axis
subplot (2,2,1); % Divide the figure window to plot the output
stem (100*k, xm); % Plot the output
xlabel ('Hz'); % Name x-axis as “Hz”
ylabel ('Magnitude'); % Name y-axis as “Magnitude”
title ('NR sampling'); % Title is “NR Sampling”

%UNDER Sampling.
fs = 1000; % Sampling frequency, fs = 1000Hz
t = 0:1/fs:9/fs; % Number of samples
x = cos(2*pi*400*t)+cos(2*pi*700*t); % Input signal
xm = abs(fft(x)); % Determine the absolute FFT of the input signal
disp(‘xm1’); % Displays xm1 on command window
disp (xm1); % Displays values of xm1 on command window
k = 0:length(xm1)-1; % Number of samples to be plot on x-axis
subplot(2,2,2); % Divide the figure window to plot the output
stem(100*k, xm1); % Plot the output
xlabel('Hz'); % Name x-axis as “Hz”
ylabel('Magnitude'); % Name y-axis as “Magnitude”
title('UNDER sampling'); % Title is “UNDER Sampling”

%OVER Sampling.
fs = 2000; % Sampling frequency, fs = 2000Hz
t = 0:1/fs:20/fs; % Number of samples
x = cos(2*pi*400*t)+cos(2*pi*700*t); % Input signal
xm = abs(fft(x)); % Determine the absolute FFT of the input signal
disp(‘xm2’); % Displays xm2 on command window
disp (xm2); % Displays values of xm2 on command window
k = 0:length(xm2)-1; % Number of samples to be plot on x-axis
subplot(2,2,3); % Divide the figure window to plot the output
stem(100*k,xm2); % Plot the output
xlabel('Hz'); % Name x-axis as “Hz”
ylabel('Magnitude'); % Name y-axis as “Magnitude”
title('OVER sampling'); % Title is “OVER Sampling”

OUTPUT :
xm
Columns 1 through 7
0.0000 0.0000 0.0000 0.0000 7.0000 0.0000 0.0000

Columns 8 through 14
14.0000 0.0000 0.0000 7.0000 0.0000 0.0000 0.0000

xm1
Columns 1 through 7
0.0000 0.0000 0.0000 5.0000 5.0000 0.0000 5.0000

Columns 8 through 10
5.0000 0.0000 0.0000

xm2
Columns 1 through 7
2.0000 2.0741 2.3496 3.1607 11.5127 0.4052 1.8998

Columns 8 through 14
8.6165 4.2355 1.2552 0.3357 0.3357 1.2552 4.2355

Columns 15 through 21
8.6165 1.8998 0.4052 11.5127 3.1607 2.3496 2.0741
 N R s a m p lin g U N D E R s a m p lin g
15 6

10 4
M agnitude

M agnitude
5 2

0 0
0 500 1000 1500 0 500 1000
Hz Hz
O V E R s a m p lin g
15

10
M agnitude

0
0 500 1000 1500 2000
Hz
 EXPERIMENT NO-2

AIM: IMPULSE RESPONSE OF A GIVEN SYSTEM

A discrete time system performs an operation on an input signal based on a predefined

criteria to produce a modified output signal. The input signal x(n) is the system
excitation, and y(n) is the system response. The transform operation is shown as,

x(n) y(n) = T[(x(n)]

T

If the input to the system is unit impulse i.e. x(n) = δ(n) then the output of the system is
known as impulse response denoted by h(n) where,

h(n) = T[δ(n)]

we know that any arbitrary sequence x(n) can be represented as a weighted sum of
discrete impulses. Now the system response is given by,
∞
y(n) = T[x(n)] = T[∑x(k) δ(n-k)] …(1)
k=-∞
For linear system (1) reduces to
∞
y(n) = ∑x(k) T[δ(n-k)] …(2)
k=-∞

The response to the shifted impulse sequence can be denoted by h(n, k) is denoted by,

h(n, k) = T[δ(n-k)] …(3)

For a time-invariant system

h(n, k) = h(n-k) …(4)

Using (4) in (3) we obtain,

T[δ(n-k)] = h(n-k) …(5)

Using (5) in (2) we get,

∞
y(n) = ∑x(k) h(n-k) …(6)
k=-∞
For a linear time-invariant system if the input sequence is x(n) and impulse response
h(n) is given, we can fine output y(n) by using eqn (6).
Which is known as convolution sum and can be represented by y(n) = x(n) * h(n)

For Example let’s find out an impulse response of a difference equation.

The general form of difference equation is,

M M
y(n) = ∑ak y(n-k) + ∑bk x(n-k) …(7)
k=1 k=0
For input x(n) = δ(n)

M
y(n) = ∑bk x(n-k) = 0 for n > M
k=0
Now (7) can be written as,

N
y(n) = ∑ak y(n-k) = 0 a0 = 1 …(8)
k=0
The solution of eqn (8) is known as homogeneous solution. The particular solution is
zero since x(n) = 0 for n > 0, that is
yp(n) = 0
Therefore we can obtain the impulse response by solving the homogeneous equation and
imposing the initial conditions to determine the arbitrary constants.

Example:
Let’s take a second order difference equation

y(n) – (1/6) y(n-1) –(1/6) y(n-2) = x(n) …(9)

For impulse response the particular solution yp(n) = 0

So, y(n) = yh(n) …(10)

Let yh(n) = λn …(11)

Substituting (11) in (9) we get,

λn - (1/6) λn-1 – (1/6) λn-2 = 0 ( x(n) = 0 for homogeneous solution)

or
λ2 - (1/6) λ – (1/6) = 0 …(12)
The roots of the characteristic equation are,
λ1 = 1/2, λ2 = -1/3

The solution yh(n) = C1 (1/2)n + C2 (-1/3)n …(13)

For impulse x(n) = δ(n); x(n) = 0 for n > 0 and x(0) = 1

Substituting the above relations in eqn (9) we have,

For n = 0,

y(0) – (1/6) y(-1) –(1/6) y(-2) = x(0) = 1

i.e. y(0) = 1. …(14)

For n = 1,

y(1) – (1/6) y(0) –(1/6) y(-1) = x(1)

y(1) – (1/6) = 0
y(1) = 1/6. …(15)

From eqn (13)

y(0) = C1 + C2 …(16)

y(1) = (1/2)C1 – (1/3)C2 …(17)

Comparing equations (14), (15), (16) and (17) we get,

C1 + C2 = 1
(1/2)C1 – (1/3)C2 = (1/6)

Solving for C1 and C2 we get,

C1 = 3/5, C2 = 2/5

Therefore, the solution

y(n) = (3/5)(1/2)n + (2/5)(-1/3)n …(18)

Now, let’s find out impulse response.

We know that,
y(n) = x(n) * h(n)
h(n) = y(n) / x(n) …(19)
Compute y(n) for various values of n
Substitute n = 0, 1, 2, 3 in eqn (18) we get,

y(0) = 1.0000
y(1) = 0.1667
y(2) = 0.1944
y(3) = 0.0602

So, y(n) = {1.0000, 0.1667, 0.1944, 0.0602}

We know that, x(n) = 1

So, the impulse response for the given difference equation using eqn (19) is
h(n) = {1.0000, 0.1667, 0.1944, 0.0602}

Program:
clc; % Clear screen
nr = [1]; % Numeartor co-efficients
dr = [1,-1/6,-1/6]; % Denominator co-efficients
h = impz(nr,dr,4); % Computes the impulse response of a given
% difference equation, returns the co-efficients
disp('Impulse response h(n):');% Displays impulse response
disp(h); % Displays impulse response on command window
subplot(2,1,1); % Divide the figure window to plot the output
stem(h); % Displays the impulse response
title('Impulse response h(n):');% Title as impulse response
x = [1]; % Input co-efficients
y = conv (h, x); % Convolution of input and impulse co-efficients to get the
% output
disp('Output y(n):') % Displays Output y(n)
disp(y); % Displays the output on command window
subplot(2,1,2); % Divide the figure window to plot the output
stem(y); % Plots the output
title('Output y(n) for given x(n)'); % Title as “Output y(n) for given x(n)”

OUTPUT:
Impulse response:
1.0000
0.1667
0.1944
0.0602
Output y(n):
1.0000
0.1667
0.1944
0.0602
 Im p u l s e r e s p o n s e
1

0 .5

0
1 1 .5 2 2 .5 3 3 .5 4

O u t p u t y ( n ) fo r g i v e n x ( n )
1

0 .5

0
1 1 .5 2 2 .5 3 3 .5 4
 EXPERIMENT NO-3

AIM: TO IMPLEMENT LINEAR CONVOLUTION OF TWO GIVEN

SEQUENCES

Theory: Convolution is an integral concatenation of two signals. It has many

applications in numerous areas of signal processing. The most popular application is the
determination of the output signal of a linear time-invariant system by convolving the
input signal with the impulse response of the system.
Note that convolving two signals is equivalent to multiplying the Fourier Transform of
the two signals.

Mathematic Formula:
The linear convolution of two continuous time signals x(t) and h(t) is defined by

For discrete time signals x(n) and h(n), is defined by

Where x(n) is the input signal and h(n) is the impulse response of the system.

In linear convolution length of output sequence is,

length(y(n)) = length(x(n)) + length(h(n)) – 1

Graphical Interpretation:

• Reflection of h(k) resulting in h(-k)

• Shifting of h(-k) resulting in h(n-k)
• Element wise multiplication of the sequences x(k) and h(n-k)
• Summation of the product sequence x(k) h(n-k) resulting in the convolution
value for y(n)

Example:

x(n) = {1, 2, 3, 1}
h(n) = {1, 1, 1}

length(y(n)) = length(x(n)) + length(y(n)) – 1

=4+3–1=6
x(k) h(k)
3
2
1 1 1 1 1

-4 -3 -2 -1 0 1 2 3 4 5 6 7 0 1 2
∞
n=0; h(-k ) y(0) = ∑ x(k) h(-k) = 1
1 k=-∞

-4 -3 -2 -1 0 1 2 3 4 5 6 7
∞
n=1; h(1-k) y(1) = ∑ x(k) h(1-k) = 1+2=3
1 k=- ∞

-4 -3 -2 -1 0 1 2 3 4 5 6 7
∞
n=2; h(2-k) y(2) =∑ x(k) h(2-k) = 1+2+3=6
1 k=-∞

-4 -3 -2 -1 0 1 2 3 4 5 6 7
∞
n=3; h(3-k) y(3) =∑ x(k) h(3-k) = 2+3+1=6
1 k=-∞

-4 -3 -2 -1 0 1 2 3 4 5 6 7
∞
n=4; h(4-k) y(4) = ∑ x(k) h(4-k) = 3+1=4
1 k=-∞

-4 -3 -2 -1 0 1 2 3 4 5 6 7
∞
n=5; h(5-k) y(5) = ∑ x(k) h(5-k) = 1
1 k=-∞

-4 -3 -2 -1 0 1 2 3 4 5 6 7
 6 6

4
3
y(n) = {1, 3, 6, 6, 4, 1}

1 1

-4 -3 -2 -1 0 1 2 3 4 5 6 7

Program:

clc; % Clear screen

x1 = input('Enter the 1st seq:'); % Get the first sequence
x2 = input('Enter the 2nd seq:'); % Get the second sequence
y = conv(x1, x2); % Convolve two signals
disp('The linear convolution of two sequences:'); % Displays the result
disp(y); % Displays the result on command window
n = 0:length(y)-1; % Defines the length for x-axis
stem(n, y); % Plots the output
xlabel('Time'); % Name the x-axis as Time
ylabel('Magnitude'); % Name the y-axis as Magnitude
title('Linear convolution'); % Title as “Linear convolution”

OUTPUT:

Enter the 1st seq:[1 2 3 1]

Enter the 2nd seq:[1 1 1]
The linear convolution of two sequences:
1 3 6 6 4 1
 Linear convolution
6

4
Magnitude

0
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
Time
 EXPERIMENT NO-4

AIM: TO IMPLEMENT CIRCULAR CONVOLUTION OF TWO GIVEN

SEQUENCES

Circular convolution:
Let x1(n) and x2(n) are finite duration sequences both of length N with DFT’s X1(k)
and X2(k). Convolution of two given sequences x1(n) and x2(n) is given by the
equation,

x3(n) = IDFT[X3(k)]

X3(k) = X1(k) X2(k)

N-1
x3(n) = ∑ x1(m) x2((n-m))N
m=0

Example:

Let’s take x1(n) = {1, 1, 2, 1} and

x2(n) = {1, 2, 3, 4}

Arrange x1(n) and x2(n) in circular fashion as shown below.

x1(m) x2(m)

1 2
x1(1) x2(1)

x1(2) x1(0)
x2(2) x2(0) 2 2 1
3 1

x1(3) x2(3)
1 4
To get x2(-m), rotate x2(m) by 4 samples in clockwise direction.

x2(-m)
4
x2(3)

x2(2) x2(0)
3 1

x2(1)
2

x3(0) = x1(m) x2(-m)

= x1(0) x2(0) + x1(1) x2(3) + x1(2) x2(2) + x1(3) x2(1)
= 1 + 4 + 6 +2
x3(0) = 13

Keep x1(m) constant and rotate x2(-m) once to compute further values.

To get x3(1) rotate x2(-m) by one sample in anti-clockwise direction

x2(1-m)
1
x2(0)

x2(3) x2(1)
4 2

x2(2)
3

x3(1) = x1(m) x2(1-m)

= x1(0) x2(1) + x1(1) x2(0) + x1(2) x2(3) + x1(3) x2(2)
 =2+1+8+3
x3(1) = 14

To get x3(2) rotate x2(1-m) by one sample in anti-clockwise direction

x2(2-m)
2
x2(1)

1 3
x2(0) x2(2)

x2(3)
4

x3(2) = x1(m) x2(2-m)

= x1(0) x2(2) + x1(1) x2(1) + x1(2) x2(0) + x1(3) x2(3)
= 3 + 2 + 2+ 4
x3(2) = 11

To get x3(3) rotate x2(2-m) by one sample in anti-clockwise direction

x2(3-m)
3
x2(2)

2 4
x2(1) x2(3)

x2(0) 1

x3(3) = x1(m) x2(3-m)

= x1(0) x2(3) + x1(1) x2(2) + x1(2) x2(1) + x1(3) x2(0)
=4+3+4+1
x3(3) = 12
The convoluted signal is,
x3(n) = {13, 14, 11, 12}

Program:
clc; % Clear screen
x1 = input('Enter 1st seq:'); % Get the first sequence
x2 = input('Enter 2nd seq:'); % Get the second sequence
n = max(length(x1),length(x2)); % Get the Maximum length out of two signals
x1 = fft(x1,n); % Compute FFT of the first sequence
x2 = fft(x2,n); % Compute FFT of the second sequence
y = x1.*x2; % Multiply the two sequences
yc = ifft(y,n); % Compute IFFT of the result
disp('circular convolution:'); % Displays circular convolution
disp(yc); % Displays the result on command window
N=0:1:n-1; % Get the length for x-axis
subplot(1,1,1); % Divide the window to plot the result
stem(N,yc); % Plot the result
xlabel('Time'); % Name the x-axis as “Time”
ylabel('Magnitude'); % Name the y-axis as “Magnitude”
title('Circular convolution'); % Title as Circular convolution

OUTPUT:
Enter 1st seq:[1 1 2 1]
Enter 2nd seq:[1 2 3 4]
Circular convolution:
13 14 11 12
 Circular convolution
15

10
Magnitude

0
0 0.5 1 1.5 2 2.5 3
Time

EXPERIMENT NO-5

AIM: AUTOCORRELATION OF A GIVEN SEQUENCE AND VERIFICATION

OF ITS PROPERTIES

Correlation: Correlation determines the degree of similarity between two signals. If the
signals are identical, then the correlation coefficient is 1; if they are totally different, the
correlation coefficient is 0, and if they are identical except that the phase is shifted by
exactly 1800(i.e. mirrored), then the correlation coefficient is -1.

Autocorrelation: When the same signal is compared to phase shifted copies of itself, the
procedure is known as autocorrelation.

The autocorrelation of a random signal describes the general dependence of the values of
the samples at one time on the values of the samples at another time. Consider a random
process x(t) (i.e. continuous time), its autocorrelation function is,
Where T is the period of observation.
Rxx (τ) is always real valued and an even function with a maximum value at τ = 0.

For sampled signal, the autocorrelation function is defined as,

Biased autocorrelation
….. Eqn.1

Unbiased autocorrelation

For m = 1 + 2 + …. + M+1
Where M is the number of lags.

Example: Autocorrelation of a sine wave.

Plot the autocorrelation of a sine wave with frequency of 1 Hz, sampling frequency of
200 Hz.

Note that we used the function xcorr to estimate the autocorrelation sequence, it has
double the number of samples as the signal x(n). An important point to remember when
using the function xcorr is that the origin is in the middle of the figure (here it is at lag =
1024). xcorr is a built in function in MATLAB library.

Program:

clc; % Clear screen

N = 1024; % Number of samples
f1 = 1; % Frequency of the sine wave
Fs = 200; % Sampling frequency
n = 0: N-1; % Sample index numbers

x = sin(2*pi*f1*n/Fs); % Input sine wave x(n)

t = [1:N]*(1/Fs); % Defines the length for x-axis

subplot(2,1,1); % Divide the figure window to plot the result

plot(t, x); % Plot the sine wave on the figure window
title('Sine wave of frequency 1 Hz (Fs = 200 Hz)'); % Title
xlabel('Time'); % Name x-axis as “Time”
ylabel('Amplitude'); % Name y-axis as “Amplitude”

Rxx = xcorr(x); % Estimate its autocorrelation using the function xcorr

subplot(2,1,2); % Divide the figure window
plot(Rxx); % Plot the autocorrealated signal on figure window
title('Autocorrelation function of the sine wave'); % Title
xlabel('lags'); % Name x-axis as “lags”
ylabel('Autocorrelation'); % Name y-axis “Autocorrelation”

OUTPUT:
 S in e w a ve o f fre q u e n c y 1 H z (F s = 2 0 0 H z )
1

0 .5
Amplitude

-0 .5

-1
0 1 2 3 4 5 6
Tim e , [s ]
A uto c o rre la tio n fu n c tio n o f th e s in e w a ve
1000
Autocorrelation

500

-5 0 0
0 500 1000 1500 2000 2500
lag s

We can also include our own functions to the MATLAB library by saving the file name
with .m extension.

For example we can write the autocorrelation function for the sampled signal defined in
Eqn.1, and we should save this file as autom.m

function [Rxx] = autom(x)

N = length(x);
Rxx = zeros(1,N);
for m = 1: N+1
for n = 1: N-m+1
Rxx(m) = Rxx(m) + x(n)*x(n+m-1);
end;
end;

We can call this function autom in the main program instead of the function xcorr, we
will get the output as,

OUTPUT:
 S i n e w a v e o f fr e q u e n c y 1 H z ( F s = 2 0 0 H z )
1

0 .5
A m plitude

-0 .5

-1
0 1 2 3 4 5 6
T im e , [ s ]
A u t o c o r r e l a t i o n fu n c t i o n o f t h e s i n e w a v e
1000

500
A utocorrelation

-5 0 0
0 200 400 600 800 1000 1200
la g s

EXPERIMENT NO-6
AIM: CROSS-CORRELATION OF A GIVEN SEQUENCE AND VERIFICATION
OF ITS PROPERTIES

Correlation: Correlation determines the degree of similarity between two signals. If the
signals are identical, then the correlation coefficient is 1; if they are totally different, the
correlation coefficient is 0, and if they are identical except that the phase is shifted by
exactly 1800(i.e. mirrored), then the correlation coefficient is -1.

Cross-correlation: When two independent signals are compared, the procedure is known
as cross-correlation.

The cross correlation function measures the dependence of the value of one signal on
another signal. For two WSS (Wide Sense Stationary) processes x(t) and y(t) it is
described by,

Or

Where T is the observation time.

For sampled signals, cross-correlation is defined as,

For m = 1 + 2 + ….. + N+1

Where N is the record length (i.e. number of samples).

Example:
Plot the cross correlation for the following signal,
x(n) = sin(2πf1t) with f1 = 1 Hz
y(n) = x(n) + w(n)

Where w(n) is a zeroes mean, unit variance of Gaussian random process.

Program:

clc; % Clear screen

N = 1024; % Number of samples
f1 = 1; % Frequency of the sine wave
Fs = 200; % Sampling frequency
n = 0: N-1; % Sample index numbers

x = sin(2*pi*f1*n/Fs); % Input sine wave x(n)

y = x + 10*randn(1, N); % Add noise to the input signal
t = [1:N]*(1/Fs); % Defines the length for x-axis

subplot(3,1,1); % Divide the figure window to plot the result

plot(x); % Plot the sine wave
title(‘Pure sine wave'); % Title as “Pure sine wave”
xlabel('Time'); % Name x-axis as “Time”
ylabel('Amplitude'); % Name y-axis as “Amplitude”

subplot(3,1,2); % Divide the figure window to plot the result

plot(y); % Plot the sine wave with noise
title(‘Pure sine wave + Noise'); % Title as “Pure sine wave + Noise”
xlabel('Time'); % Name x-axis as “Time”
ylabel('Amplitude'); % Name y-axis as “Amplitude”

Rxy = xcorr(x,y); % Estimate its cross-correlation using the function xcorr

subplot(3,1,3); % Divide the figure window to plot the result
plot(Rxy); % Plot the cross-correlated signal on figure window
title('Cross-correlation Rxy’); % Title as “Cross-correlation Rxy”
xlabel('lags'); % Name x-axis as “lags”
ylabel('Cross-correlation'); % Name y-axis “Cross-correlation”

OUTPUT:
 P u re s in e w a ve
1
A m plitude
0

-1
0 200 400 600 800 1000 1200
T im e
P u re s in e w a ve + N o is e
50
A m plitude

-5 0
0 200 400 600 800 1000 1200
T im e
C ro s s -c o rre la t io n R x y
1000
C ros s -c orrelation

-1 0 0 0
0 500 1000 1500 2000 2500
la g s

EXPERIMENT NO-7
AIM: TO SOLVE A GIVEN DIFFERENCE EQUATION

A General Nth order Difference equations looks like,

y[n] + a1y[n-1] + a2y[n-2] + …+ aNy[n-N) =

b0x[n] + b1x[n-1] + …. + bMx[n-M]

Here y[n] is “Most advanced” output sample and y[n-m] is “Least advanced” ouput
sample.
The difference between these two index values is the order of the difference equation.
Here we have: n-(n-N) = N

We can rewrite the above equation as,

y[n] + ∑ai y[n-i] = ∑bi x[n-i]

y[n] becomes,

y[n] = -∑ai y[n-i] + ∑bi x[n-i]

Example:

y[n+2] – 1.5y[n+1] + y[n] = 2x[n]

In general we start with the “Most advanced” output sample. Here it is y[n+2].
So, here we need to subtract 2 from each sample argument. We get

y[n] – 1.5y[n-1] + y[n-2] = 2x[n-2]

 y[n] = 1.5y[n-1] - y[n-2] + 2x[n-2]

Lets assume our input x[n] = u[n] = 0 x<0

1 x≥0

In our example we have taken x[n] = u[n] = 0 x<0

1 0≤x<10

We need N past values to solve Nth order difference equation.

y[-2] = 1
y[-1] = 2
Compute y[n] for various values of n
y[0] = 1.5y[-1] - y[-2] + 2x[-2]
= 1.5*2 – 1 + 2*0
y[0] = 2

y[1] = 1.5y[0] - y[-1] + 2x[-1]

= 1.5*2 - 2 + 2*0
y[1] = 1

y[2] = 1.5y[1] - y[0] + 2x[0]

= 1.5*1 – 2 + 2*1
y[2] = 1.5

And so on…

Program:
clc; % Clear screen
x = [0 0 ones(1, 10)]; % Input x[n] = u[n]
y_past = [1 2]; % Past values to solve difference equation
y(1) = y_past(1); % y[1] <= y[-2]
y(2) = y_past(2); % y[2] <= y[-1]

for k = 3:(length(x)+2) % Compute y[n] for various values of n

y(k) = 1.5*y(k-1) - y(k-2) + 2*x(k-2);
end % End of for loop

disp(‘Solution for the given difference equation:’);

disp(y); % Display the result on command window
subplot(1,1,1); % Divide the window to plot the result
stem (-2:(length(y)-3),y); % Plot the result
xlabel(‘Input x[n]’); % Name x-axis as Input x[n]
ylabel(‘Output y[n]’); % Name y-axis as Output y[n]
title(‘Difference equation’); % Title as “Difference equation”

OUTPUT:

Solution for the given difference equation:

Columns 1 through 7
1.0000 2.0000 2.0000 1.0000 1.5000 3.2500 5.3750

Columns 8 through 14
6.8125 6.8438 5.4531 3.3359 1.5508 0.9902 1.9346
 D iffe r e n c e e q u a t io n
8

6
O utp ut y [n]

0
-2 0 2 4 6 8 1 0 1 2
In p u t x [ n ]
 EXPERIMENT NO-8

AIM: TO COMPUTE N-POINT DFT OF A GIVEN SEQUENCE AND TO PLOT

MAGNITUDE AND PHASE SPECTRUM.

Discrete Fourier Transform: The Discrete Fourier Transform is a powerful

computation tool which allows us to evaluate the Fourier Transform X(ejω) on a digital
computer or specially designed digital hardware. Since X(ejω) is continuous and
periodic, the DFT is obtained by sampling one period of the Fourier Transform at a finite
number of frequency points. Apart from determining the frequency content of a signal,
DFT is used to perform linear filtering operations in the frequency domain.

The sequence of N complex numbers x0,..., xN−1 is transformed into the sequence of N
complex numbers X0, ..., XN−1 by the DFT according to the formula:

N-1
X(k) = ∑x(n)e-j2πnk/N k = 0,1, …. N-1
n=0

Example:
Lets assume the input sequence x[n] = [1 1 0 0]

We have,
N-1
X(k) = ∑x(n)e-j2πnk/N k = 0,1, …. N-1
n=0

For k = 0,
3
X(0) = ∑x(n) = x(0) + x(1) + x(2) + x(3)
n=0
X(0) = 1+1+0+0 = 2

For k = 1,
3
X(1) = ∑x(n)e-jπn/2 = x(0) + x(1) e-jπ/2 + x(2) e-jπ + x(3) e-j3π/2
n=0
= 1 + cos(π/2) - jsin(π/2)
X(1) = 1 – j
For k = 2,
3
X(2) = ∑x(n)e-jπn = x(0) + x(1) e-jπ + x(2) e-j2π + x(3) e-j3π
n=0
= 1 + cos π – jsin π
X(2) = 1-1 = 0

For k = 3,
3
X(3) = ∑x(n)e-j3nπ/2 = x(0) + x(1) e-j3π/2 + x(2) e-j3π + x(3) e-j9π/2
n=0
= 1 + cos(3π/2) - jsin(3π/2)
X(3) = 1 + j

The DFT of the given sequence is,

X(k) = { 2, 1-j, 0, 1+j }

Program:
clc; % Clear screen
x1 = input('Enter the sequence:'); % Get the input sequence
n = input('Enter the length:'); % Get the value of N
m = fft(x1,n); % Computes the DFT using FFT algorithm
disp('N-point DFT of a given sequence:'); % Display the results
disp(m); % Displays the result on command window
N = 0:1:n-1; % Decides the length to plot the results

subplot(2,2,1); % Divide the figure window to plot the

% results
stem(N,m); % Plots the magnitude spectrum
xlabel('Length'); % Name x-axis as “Length”
ylabel('Magnitude of X(k)'); % Name y-axis as “Magnitude of X(k)”
title('Magnitude spectrum:'); % Title as “Magnitude spectrum”
an = angle(m); % Get the angle of the output sequence X(k)

subplot(2,2,2); % Divide the figure window to plot the

% results
stem(N, an); % Plots the phase spectrum
xlabel('Length'); % Name x-axis as “Length”
ylabel('Phase of X(k)'); % Name y-axis as “Phase of X(k)”
title('Phase spectrum:'); % Title as “Phase spectrum”
OUTPUT:

Enter the sequence: [1 1 0 0]

Enter the length: 4
N-point DFT of a given sequence:
Columns 1 through 3

2.0000 1.0000 - 1.0000i 0

Column 4

1.0000 + 1.0000i

Magnitude spectrum: Phase spectrum:

2 1
Magnitude of X(k)

1.5 0.5
Phase of X(k)

1 0

0.5 -0.5

0 -1
0 1 2 3 0 1 2 3
Length Length
 EXPERIMENT NO-9

AIM: LINEAR CONVOLUTION OF TWO GIVEN SEQUENCES USING DFT

AND IDFT

Theory: Convolution is an integral concatenation of two signals. It has many

applications in numerous areas of signal processing. The most popular application is the
determination of the output signal of a linear time-invariant system by convolving the
input signal with the impulse response of the system.
Note that convolving two signals is equivalent to multiplying the Fourier Transform of
the two signals.

Mathematic Formula:
The linear convolution of two continuous time signals x(t) and h(t) is defined by

For discrete time signals x(n) and h(n), is defined by

Where x(n) is the input signal and h(n) is the impulse response of the system.

Example:
x1(n) = {1, 1, 2}
x2(n) = {1, 2}

For linear convolution,

Length N = Length(x1) + Length(x2) - 1
N=3+2–1=4

Convolution of two sequences x1(n) and x2(n) is,

x3(n) = IDFT[X3(k)]
x3(n) = IDFT[X1(k) X2(k)]

Where,
X1(k) = DFT [x1(n)]

X2(k) = DFT [x2(n)]

 N-1
X1(k) = ∑ x1(n)e-j2πkn/N k= 0, 1, 2, … , N-1
n=0

Given x1(n) = {1, 1, 2} and N=4

3
X1(0) = ∑ x1(n) = 1 + 1 + 2 = 4
n=0

5
X1(1) = ∑ x1(n)e-jπn/2 = 1 – j – 2 = -1 - j
n=0

5
X1(2) = ∑ x1(n)e-jπn = 1 – 1 + 2 = 2
n=0

5
X1(3) = ∑ x1(n)e-j3πn/2 = 1 + j – 2 = -1 + j
n=0

X1(k) = {4, -1-j, 2, -1+j}

Now,
N-1
X2(k) = ∑ x2(n)e-j2πkn/N k= 0, 1, 2, … , N-1
n=0

Given x2(n) = {1, 2} and N=4

3
X2(0) = ∑ x2(n) = 1 + 2 = 3
n=0
 3
X2(1) = ∑ x2(n) e-jπn/2 = 1 + 2(-j) = 1 - j2
n=0

3
X2(2) = ∑ x2(n) e-jπn = 1 + 2(-1) = -1
n=0

3
X2(3) = ∑ x2(n) e-j3πn/2 = 1 + 2(j) = 1 + j2
n=0

X2(k) = {3, 1-j2, -1, 1+j2}

We know that,
X3(k) = X1(k) X2(k)
X3(k) = {12, -3+j, -2, -3-j}

Convolution of two given sequences is,

x3(n) = IDFT[X3(k)]

N-1
x3(n) = (1/N) ∑ X3(k)ej2πkn/N n = 0, 1, 2, …. , N-1
k=0

3
x3(0) = (1/4) ∑ X3(k) = (1/4) [12 – 3 +j -2 -3 –j] = 1
k=0

3
x3(1) = (1/4) ∑ X3(k)ejπk/2
k=0
x3(1) = (1/4) [12 + (-3 + j) j + (-2) (-1) + (-3 - j) (-j)] = 3

3
x3(2) = (1/4) ∑ X3(k)ejπk
k=0

x3(2) = (1/4) [12 + (-3 + j) (-1) + (-2) (1) + (-3 - j) (-1)] = 4

 3
x3(3) = (1/4) ∑ X3(k)ej3πk/2
k=0

x3(3) = (1/4) [12 + (-3 + j) (-j) + (-2) (-1) + (-3 - j) (j)] = 4

Convoluted sequence of two given sequences is,

x3(n) = {1, 3, 4, 4}

Program:
clc; % Clear screen
x1 = input('Enter the 1st seq:'); % Get the first sequence
x2 = input('Enter the 2nd seq:'); % Get the second sequence
n = length(x1) + length(x2)-1; % Get the length of the sequence
x1 = fft(x1,n); % Compute the DFT of x1 using FFT algorithm
x2 = fft(x2,n); % Compute the DFT of x2 using FFT algorithm
y = x1.*x2; % Multiply two DFT’s
yc = ifft(y,n); % Compute IDFT using IFFT algorithm
disp('Linear convolution using DFT and IDFT:'); % Display Linear convolution
disp(yc); % Displays the result on command window
N = 0:1:n-1; % Defines the length of x-axis to plot the result
subplot(1,1,1); % Divide the window to plot the result
stem(N,yc); % Plots the result
xlabel('Time'); % Name the x-axis as Time
ylabel('Magnitude'); % Name the y-axis as Magnitude
title('Linear convolution using DFT and IDFT:'); % Title

OUTPUT:
Enter the 1st seq: [1 1 2]
Enter the 2nd seq: [1 2]
Linear convolution using DFT and IDFT:
1 3 4 4
 Linear convolution using DFT and IDFT:
4

3.5

2.5
Magnitude

1.5

0.5

0
0 0.5 1 1.5 2 2.5 3
Time
 EXPERIMENT NO-10

AIM: TO IMPLEMENT CIRCULAR CONVOLUTION OF TWO GIVEN

SEQUENCES USING DFT AND IDFT

Circular convolution:
Let x1(n) and x2(n) are finite duration sequences both of length N with DFT’s X1(k)
and X2(k). Convolution of two given sequences x1(n) and x2(n) is given by,

x3(n) = IDFT[X3(k)]
x3(n) = IDFT[X1(k) X2(k)]

Where,
X1(k) = DFT [x1(n)]
X2(k) = DFT [x2(n)]

Example:
x1(n) = {1, 1, 2, 1}
x2(n) = {1, 2, 3, 4}

N-1
X1(k) = ∑ x1(n)e-j2πkn/N k= 0, 1, 2, … , N-1
n=0

Given x1(n) = {1, 1, 2, 1} and N=4

3
X1(0) = ∑ x1(n) = 1 + 1 + 2 + 1 = 5
n=0

3
X1(1) = ∑ x1(n)e-jπn/2 = 1 – j – 2 + j = -1
n=0

3
X1(2) = ∑ x1(n)e-jπn = 1 – 1 + 2 - 1 = 1
n=0

3
X1(3) = ∑ x1(n)e-j3πn/2 = 1 + j - 2 - j = -1
n=0
X1(k) = {5, -1, 1, -1}

Now,

N-1
X2(k) = ∑ x2(n)e-j2πkn/N k= 0, 1, 2, … , N-1
n=0

Given x2(n) = {1, 2, 3, 4} and N=4

3
X2(0) = ∑ x2(n) = 1 + 2 + 3 + 4 = 10
n=0

3
X2(1) = ∑ x2(n) e-jπn/2 = 1 + 2(-j) + 3(-1) + 4(j) = -2 + j2
n=0

3
X2(2) = ∑ x2(n) e-jπn = 1 + 2(-1) + 3(1) + 4(-1) = -2
n=0

3
X2(3) = ∑ x2(n) e-j3πn/2 = 1 + 2(j) + 3(-1) + 4(-j) = -2 – j2
n=0

X2(k) = {10, -2+j2, -2, -2-j2}

We know that,
X3(k) = X1(k) X2(k)
X3(k) = {50, 2 – j2, -2, 2 + j2}

Convolution of two given sequences is,

x3(n) = IDFT[X3(k)]

N-1
x3(n) = (1/N) ∑ X3(k)ej2πkn/N n = 0, 1, 2, …. , N-1
k=0
 3
x3(0) = (1/4) ∑ X3(k) = (1/4) [50 + 2 - j2 – 2 + 2 + j2] = 13
k=0

3
x3(1) = (1/4) ∑ X3(k)ejπk/2
k=0
x3(1) = (1/4) [50 + (2 - j2) j + (-2) (-1) + (2 + j2) (-j)] = 14

3
x3(2) = (1/4) ∑ X3(k)ejπk
k=0

x3(2) = (1/4) [50 + (2 - j2) (-1) + (-2) (1) + (2 + j2) (-1)] = 11

3
x3(3) = (1/4) ∑ X3(k)ej3πk/2
k=0
x3(3) = (1/4) [50 + (2 - j2) (-j) + (-2) (-1) + (2 + j2) (j)] = 12

Convoluted sequence of two given sequences is,

x3(n) = {13, 14, 11, 12}

Program:
clc; % Clear screen
x1 = input('Enter 1st sequence:'); % Get the first sequence
x2 = input('Enter 2nd sequence:'); % Get the second sequence
n = max(length(x1), length(x2)); % Get the maximum length of the two sequences
x1 = fft(x1,n); % Compute the DFT of the x1 using FFT algorithm
x2 = fft(x2,n); % Compute the DFT of the x2 using FFT algorithm
y = x1.*x2; % Multiply two DFT’s
yc = ifft(y,n); % Compute the IDFT of mutliplied sequence to get
% the convoluted sequence
disp('Circular convolution using DFT and IDFT:'); % Displays Circular convolution
disp(yc); % Displays the result on command window
N = 0:1:n-1; % Defines the length of x-axis to plot the result
subplot(1,1,1); % Divide the window to plot the result
stem(N,yc); % Plots the results
xlabel('Time'); % Name the x-axis as “Time”
ylabel('Magnitude'); % Name the y-axis as “Magnitude”
title('Circular convolution using DFT and IDFT:'); % Title
OUTPUT:

Enter 1st sequence:[1 1 2 1]

Enter 2nd sequence:[1 2 3 4]
Circular convolution using DFT and IDFT:
13 14 11 12

C ir c u la r c o n v o lu t io n u s in g D F T a n d ID F T :
1 5

1 0
M a g n itu d e

0
0 0 .5 1 1 .5 2 2 .5 3
T im e
 EXPERIMENT NO-11

AIM: DESIGN AND IMPLEMENTATION OF FIR FILTER TO MEET GIVEN

SPECIFICATIONS (LOW PASS FILTER USING HAMMING WINDOW)

Finite Impulse Response (FIR) Filter: The FIR filters are of non-recursive type,
whereby the present output sample is depending on the present input sample and previous
input samples.
The transfer function of a FIR causal filter is given by,
N-1
H(z) = ∑ h(n)z-n
n=0

Where h(n) is the impulse response of the filter.

The Fourier transform of h(n) is

N-1
H(ejw) = ∑h(n)e-jwn
n=0

In the design of FIR filters most commonly used approach is using windows.
The desired frequency response Hd(ejw) of a filter is periodic in frequency and can be
expanded in Fourier series. The resultant series is given by,
π
hd(n) = (1/2π) ∫ H(ejw)ejwn dw
-π
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.
The Fourier coefficients of the filter are modified by multiplying the infinite impulse
response with a finite weighing sequence w(n) called a window.

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

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

After multiplying w(n) with hd(n), we get a finite duration sequence h(n) that satisfies
the desired magnitude response,

h(n) = hd(n) w(n) for |n| ≤ [(N-1)/2]

=0 for |n| > [(N-1)/2]
The frequency response H(ejw) of the filter can be obtained by convolution of Hd(ejw)
and W(ejw) is given by,
π
H(ejw) = (1/2π) ∫ Hd(ejθ) W(ej(w-θ) dθ
-π

H(ejw) = Hd(ejw) * W(ejw)

Example:
Here we design a lowpass filter using hamming window.
Hamming window function is given by,

wH(n) = 0.54 + 0.46 cos ((2πn)/(N-1)) for –(N-1)/2 ≤ n ≤ (N-1)/2

=0 otherwise

The frequency response of Hamming window is,

WH(ejw ) = 0.54[(sin(wN/2))/(sin(w/2))]
+ 0.23[sin (wN/2 – πN/N – 1)/sin (w/2 – π/N -1)]
+ 0.23[sin (wN/2 + πN/N – 1)/sin (w/2 + π/N – 1)]

Program:

clc; % Clear screen

wp = input('Pass band edge freq:'); % Get Passband edge frequency
ws = input('Stop band edge freq:'); % Get Stopband edge frequency
tw = ws-wp; % Subtract PB frequency from SB frequency

N = ceil(6.6*pi/tw)+1; % Compute length

wn = (hamming(N)); % Returns N-point symmetric hamming window in
% column vector
B = fir1(N-1,wp,wn); % Designs (N-1)th order lowpass FIR filter and
% returns filter coefficients in length N in vector B
disp('Impulse response coeff='); % Displays Impulse response coefficients
disp(B); % Displays the coefficients on command window
[H,w] = freqz(B,1,256); % Digital filter frequency response. This function
% returns the N-point complex frequency response
% vector H and the N-point frequency vector w in
% radians/sample of the filter.
Mag = 20*log10(abs(H)); % Get the magnitude
plot(w/pi*pi, Mag); % Plot the Magnitude spectrum
xlabel(‘Frequency in radians in terms of pi');% Name x-axis
ylabel('Gain in db'); % Name y-axis as “Gain in db”
OUTPUT:

Pass band edge freq: 0.05*pi

Stop band edge freq: 0.4*pi
Impulse response coeff=
Columns 1 through 7

-0.0027 -0.0035 -0.0041 -0.0010 0.0105 0.0333 0.0666

Columns 8 through 14

0.1046 0.1383 0.1580 0.1580 0.1383 0.1046 0.0666

Columns 15 through 20

0.0333 0.0105 -0.0010 -0.0041 -0.0035 -0.0027

-1 0

-2 0

-3 0

-4 0
Gain in db

-5 0

-6 0

-7 0

-8 0

-9 0

-1 0 0
0 0 .5 1 1.5 2 2.5 3 3 .5
F re q u e n c y in ra d ia n s in t e rm s o f p i
 EXPERIMENT NO-12

AIM: DESIGN AND IMPLEMENTATION OF IIR FILTER TO MEET GIVEN

SPECIFICATIONS

Basically digital filter is a linear time-invariant discrete time system.

Infinite Impulse Response(IIR) filter: IIR filters are of recursive type, whereby the
present output sample depends on the present input, past input samples and output
samples.
The impulse response h(n) for a realizable filter is,
h(n) = 0 for n≤0

And for stability, it must satisfy the condition,

∞
∑ | h(n) | < ∞
n=0

Example:
Let’s design an analog Butterworth lowpass filter.
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 Ωc = 1rad/sec for the value of N.
4. calculate the value of cutoff frequency Ωc
5. find the transfer function Ha(s) for the above value of Ωc by substituting
s→ (s/ Ωc) in H(s).

Program:

clc; % Clear screen

Pa = input('Passband attenuation in DB:'); % Get the passband attenuation
Sa = input('Stopband attenuation in DB:'); % Get the stopband attenuation
Fpb = input('Passband edge frequency in Hz:'); % Get Passband edge frequency
Fsb = input('Stopband edge frequency in Hz:'); % Get Stopband edge frequency
fs = input('Sampling frequency:'); % Get Sampling frequency

wp = 2*Fpb/fs; % Convert PB edge frequency in Hz to radians

ws = 2*Fsb/fs; % Convert SB edge frequency in Hz to radians

[N,wn] = buttord(wp,ws,Pa,Sa); % Find cutoff frequency and order of the filter

disp('Order:'); % Display the Order
disp(N); % Display order N on command window
disp('Cutoff frequency:'); % Display Cutoff frequency
disp(wn); % Display Cutoff frequency on command window
[b,a] = butter(N,wn); % Get Numerator and denominator coefficients of
% the filter
disp('b='); disp(b); % Display the numerator coefficients on command
% window
disp('a='); disp(a); % Display the denominator coefficients on
% command window
w = 0:0.01:pi; % Defines length for x-axis to plot the result
[h,om] = freqz(b,a,w,'whole'); % Frequency response of the filter
m = 20*log10(abs(h)); % Absolute value of the frequency response vector
an = angle(h); % Get the phase of the frequency response vector

subplot(2,1,1); % Divide the figure window to plot the frequency

% response
plot(om/pi,m); % Plot the response
xlabel('Normalised frequency:'); % Name x-axis as “Normalised frequency”
ylabel('Gain in db:'); % Name y-axis as “ Gain in db”
title(‘Frequency Response:’); % Title as Frequency Response
subplot(2,1,2); % Divide the figure window
plot(om/pi,an); % Plot the Phase spectrum
xlabel('Normalised frequency:'); % Name x-axis as “Normalised frequency”
ylabel('Phase in radians:'); % Name y-axis as Phase in radians
title(‘Phase spectrum’); % Title as Phase spectrum

OUTPUT:

assband attenuation in DB:4

Stopband attenuation in DB:30
Passband edge frequency in Hz:400
Stopband edge frequency in Hz:800
Sampling frequency:2000
Order:
3

Cutoff frequency:
0.4914

b=
0.1600 0.4800 0.4800 0.1600

a=
1.0000 -0.0494 0.3340 -0.0045
Procedure to Setup Emulator:

Setup CCStudio v3.1.lnk

1.Open the Setup CCStudio v3.1
2. Select Create Board (Marked in Circle, can witness in the below Figure)

Chose this option

3. Right Click on the TI XDS510 emulator and select add to system.. Enter,
(After selecting that options a Connection Properties window will be opened as shown in
step 4).
4. Provide Connection Name as ChipMax_6713 and click on Next.

5. Choose the option Falling edge is JTAG Standard in TMS/TDO Output Timing.
6. Right Click on TMS320C6710 and Select Add to System…Enter.
7. a) Provide Processor Name as TMS320C6713_0
b) Select GEL File, Click on browse icon and select DSP621x_671x.gel.
c) Select N/A in Master/Slave.
d) Click on Ok.
8. Click on Save and quit (highlighted by Circle).

9. Click on Yes to start Code Composer Studio.

10. Go to Debug and select the option connect.
11. Now Target is connected
 Experiment 1: Linear Convolution
Procedure to create new Project:
1. To create project, Go to Project and Select New.

2. Give project name and click on finish.

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).

3. Click on File New Source File, To write the Source Code.
Aim: Linear Convolution of the two given sequences

Mathematical Formula:
The linear convolution of two continuous time signals x(t) and h(t) is defined by

For discrete time signals x(n) and h(n), is defined by

Where x(n) is the input signal and h(n) is the impulse response of the system.

In linear convolution length of output sequence is,

Length (y(n)) = length(x(n)) + length(h(n)) – 1

Program:
#include<stdio.h>
main()
{ int m=4; /*Lenght of i/p samples sequence*/
int n=4; /*Lenght of impulse response Co-efficients */
int i=0,j;
int x[10]={1,2,3,4,0,0,0,0}; /*Input Signal Samples*/
int h[10]={1,2,3,4,0,0,0,0}; /*Impulse Response Co-efficients*/
/*At the end of input sequences pad 'M' and 'N' no. of zero's*/
int *y;
y=(int *)0x0000100;
for(i=0;i<m+n-1;i++)
{
y[i]=0;
for(j=0;j<=i;j++)
y[i]+=x[j]*h[i-j];
}
for(i=0;i<m+n-1;i++)
printf("%d\n",y[i]);
}

Output:
1, 4, 10, 20, 25, 24, 16.
4. Enter the source code and save the file with “.C” extension.

5. Right click on source, Select add files to project .. and Choose “.C “ file Saved before.
6. Right Click on libraries and select add files to Project.. and choose
C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib and click open.
7. a)Go to Project to Compile .
b) Go to Project to Build.
c) Go to Project to Rebuild All.

Here it
shows
error if
any
8. Go to file and load program and load “.out” file into the board..
9. Go to Debug and click on run to run the program.
10. Observe the output in output window.
11. To see the Graph go to View and select time/frequency in the Graph, And give the
correct Start address provided in the program, Display data can be taken as per user.
12. Green line is to choose the point, Value at the point can be seen (Highlighted by
circle at the left corner).
 Experiment 2: Circular Convolution
Procedure to create new Project:
1. To create project, go to Project and Select New.

2. Give project name and click on finish.

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).

3. Click on File New Source File, to write the Source Code.
AIM: To implement circular convolution of two sequences.

Circular Convolution:
Let x1(n) and x2(n) are finite duration sequences both of length N with DFT’s X1(k) and
X2(k). Convolution of two given sequences x1(n) and x2(n) is given by the equation,

x3(n) = IDFT[X3(k)]

X3(k) = X1(k) X2(k)

N-1

x3(n) = ∑ x (m) x ((n-m))

1 2 N
m=0

Program:

#include<stdio.h>
int m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],a[30];
void main()
{
int *y;
y=(int *)0x0000100;
printf(" enter the length of the first sequence\n");
scanf("%d",&m);
printf(" enter the length of the second sequence\n");
scanf("%d",&n);
printf(" enter the first sequence\n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf(" enter the second sequence\n");
for(j=0;j<n;j++)
scanf("%d",&h[j]);
if(m-n!=0) /*If length of both sequences are not equal*/
{
if(m>n) /* Pad the smaller sequence with zero*/
{
for(i=n;i<m;i++)
h[i]=0;
n=m;
}
for(i=m;i<n;i++)
x[i]=0;
m=n;
}
y[0]=0;
a[0]=h[0];
for(j=1;j<n;j++) /*folding h(n) to h(-n)*/
a[j]=h[n-j];
/*Circular convolution*/
for(i=0;i<n;i++)
y[0]+=x[i]*a[i];
for(k=1;k<n;k++)
{
y[k]=0;
/*circular shift*/
for(j=1;j<n;j++)
x2[j]=a[j-1];
x2[0]=a[n-1];
for(i=0;i<n;i++)
{
a[i]=x2[i];
y[k]+=x[i]*x2[i];
}
}
/*displaying the result*/
printf(" the circular convolution is\n");
for(i=0;i<n;i++)
printf("%d ",y[i]);
}

Output:

enter the length of the first sequence

4
enter the length of the second sequence
4
enter the first sequence
4321
enter the second sequence
1111
the circular convolution is
10 10 10 10
4. Enter the source code and save the file with “.C” extension.

5. Right click on source, Select add files to project .. and Choose “.C “ file Saved before.
6. Right Click on libraries and select add files to Project.. and choose
C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib and click open.
7. a) Go to Project to Compile .
b) Go to Project to Build.
c) Go to Project to Rebuild All.

In case of
any errors
or
warnings
it will be
displayed
here
8. Go to file and load program and load “.out” file into the board..
9. Go to Debug and click on run to run the program.
10. Enter the input data to calculate the circular convolution.
The corresponding output will be shown on the output window as shown below
11. To see the Graph go to View and select time/frequency in the Graph, and give the
correct Start address provided in the program, Display data can be taken as per user.
12. 12. Green line is to choose the point, Value at the point can be seen (Highlighted by
circle at the left corner).
 Experiment 3: N-Point DFT
Procedure to create new Project:
1. To create project, Go to Project and Select New.

2. Give project name and click on finish.

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).

3. Click on File New Source File, To write the Source Code.
AIM: TO COMPUTE N-POINT DFT OF A GIVEN SEQUENCE AND TO PLOT
MAGNITUDE AND PHASE SPECTRUM.

Discrete Fourier Transform: The Discrete Fourier Transform is a powerful

computation tool which allows us to evaluate the Fourier Transform X(ejω) on a digital
computer or specially designed digital hardware. Since X(ejω) is continuous and periodic,
the DFT is obtained by sampling one period of the Fourier Transform at a finite number
of frequency points. Apart from determining the frequency content of a signal, DFT is
used to perform linear filtering operations in the frequency domain.

The sequence of N complex numbers x0,..., xN−1 is transformed into the sequence of N
complex numbers X0, ..., XN−1 by the DFT according to the formula:

N-1
X(k) = ∑x(n)e -j2πnk/N
k = 0,1, …. N-1
n=0

Program:

//DFT of N-point from lookup table. Output from watch window

#include <stdio.h>
#include <math.h>
#define N 4 //number of data values

float pi = 3.1416;
short x[N] = {1,1,0,0}; //1-cycle cosine
float out[2] = {0,0}; //initialize Re and Im results

void dft(short *x, short k, float *out) //DFT function

{
float sumRe = 0; //initialize real component
float sumIm = 0; //initialize imaginary component
int i = 0;
float cs = 0; //initialize cosine component
float sn = 0; //initialize sine component

for (i = 0; i < N; i++) //for N-point DFT

{
cs = cos(2*pi*(k)*i/N); //real component
sn = sin(2*pi*(k)*i/N); //imaginary component
sumRe = sumRe + x[i]*cs; //sum of real components
sumIm = sumIm - x[i]*sn; //sum of imaginary components
}
out[0] = sumRe; //sum of real components
out[1] = sumIm; //sum of imaginary components
printf("%f %f\n",out[0],out[1]);
}

void main()
{
int j;
for (j = 0; j < N; j++)
dft(x, j, out); //call DFT function
}

Output:
2.000000 0.000000
0.999996 -1.000000
0.000000 0.000007
1.000011 1.000000
4. Enter the source code and save the file with “.C” extension.
5. Right click on source, Select add files to project .. and Choose “.C “ file Saved before.
6. Right Click on libraries and select add files to Project.. and choose
C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib and click open.
7. a) Go to Project to Compile .
b) Go to Project to Build.
c) Go to Project to Rebuild All.

It will shows
warnings and
errors if any

Here it
shows
error if
any
8. Go to file and load program and load “.out” file into the board..
9. Go to Debug and click on run to run the program.
10. Observe the output in output window.

Output will
shown here
11. To see the Graph go to View and select time/frequency in the Graph, And give the
correct Start address provided in the program, Display data can be taken as per user.
12. Green line is to choose the point, Value at the point can be seen (Highlighted by
circle at the left corner).
 Experiment 4: FIR FILTER
Procedure to create new Project:
1. To create project, Go to Project and Select New.

2. Give project name and click on finish.

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).
3. Click on File New Source File, To write the Source Code.
AIM: Realization of FIR filter (any type) to meet given specifications. The input can be a
signal from Function Generator/Speech signal
Finite Impulse Response (FIR) Filter: The FIR filters are of non-recursive type,
whereby the present output sample is depending on the present input sample and previous
input samples.
The transfer function of a FIR causal filter is given by,
N-1
H(z) = ∑ h(n)z-n
n=0
Where h(n) is the impulse response of the filter.
The Fourier transform of h(n) is
N-1
H(ejw) = ∑h(n)e-jwn
n=0
In the design of FIR filters most commonly used approach is using windows.
The desired frequency response Hd(ejw) of a filter is periodic in frequency and can be
expanded in Fourier series. The resultant series is given by, π
hd(n) = (1/2π) ∫ H(ejw)ejwn dw
-π
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.
The Fourier coefficients of the filter are modified by multiplying the infinite impulse
response with a finite weighing sequence w(n) called a window.

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

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

After multiplying w(n) with hd(n), we get a finite duration sequence h(n) that satisfies
the desired magnitude response,

h(n) = hd(n) w(n) for |n| ≤ [(N-1)/2]

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

The frequency response H(ejw) of the filter can be obtained by convolution of Hd(ejw)
and W(ejw) is given by,
π
H(ejw) = (1/2π) ∫ Hd(ejθ) W(ej(w-θ) dθ
-π
H(e ) = Hd(e ) * W(ejw)
jw jw

Program:
#include <stdio.h>
#include "c6713dsk.h"
#include "master.h"
#include "aic23cfg.h"
#include "dsk6713_aic23.h"
#include <std.h>
#include <swi.h>
#include <log.h>
#include <c6x.h>
#include <csl.h>
#include <csl_mcbsp.h>

/* Length of sine wave table */

#define SINE_TABLE_SIZE 48

// Delta
/*float filter_Coeff[] ={0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,
0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,
0.00,0.00,0.00,0.00,0.00,0.00,1.00,0.00};*/

// Low Pass Filter

/*float filter_Coeff[] ={2.715297634171146e-003, 2.315819802942924e-004,-
1.244493581373643e-002, 7.244364917221401e-003,1.207341716154354e-002,
2.734134166585232e-003,
-1.941706440790678e-002, -1.729098218843226e-002,1.773008730568675e-002,
4.091495174059349e-002,2.113436751136944e-003, -6.788468549771730e-002,
-6.059440700570791e-002, 8.970313256448266e-002,3.014572949374625e-001,
4.019009454299968e-001,3.014572949374625e-001, 8.970313256448266e-002,
-6.059440700570791e-002, -6.788468549771730e-002,2.113436751136944e-003,
4.091495174059349e-002,1.773008730568675e-002, -1.729098218843226e-002,
-1.941706440790678e-002, 2.734134166585232e-003,1.207341716154354e-002,
7.244364917221401e-003,-1.244493581373643e-002, 2.315819802942924e-004,
2.715297634171146e-003};*/

// High Pass Filter

float filter_Coeff[] ={3.294316420702696e-004,3.800020076486443e-
003,9.822200806739014e-003,1.517265313889167e-002,
1.323547007544908e-002,2.635896986048919e-004,-1.808215737734512e-002,-
2.666833013269758e-002,
-1.155354962270025e-002,2.448211866656400e-002,5.534101055783895e-
002,4.424359087198896e-002,
-2.922329551555757e-002,-1.473332022689261e-001,-2.574625659073934e-
001,6.976203109125493e-001,
-2.574625659073934e-001,-1.473332022689261e-001,-2.922329551555757e-
002,4.424359087198896e-002,
5.534101055783895e-002,2.448211866656400e-002,-1.155354962270025e-002,-
2.666833013269758e-002,
-1.808215737734512e-002,2.635896986048919e-004,1.323547007544908e-
002,1.517265313889167e-002,
9.822200806739014e-003,3.800020076486443e-003,3.294316420702696e-004};

// Pre-generated sine wave data, 16-bit signed samples

int sinetable[SINE_TABLE_SIZE] = {
0x0000, 0x10b4, 0x2120, 0x30fb, 0x3fff, 0x4dea, 0x5a81, 0x658b,
0x6ed8, 0x763f, 0x7ba1, 0x7ee5, 0x7ffd, 0x7ee5, 0x7ba1, 0x76ef,
0x6ed8, 0x658b, 0x5a81, 0x4dea, 0x3fff, 0x30fb, 0x2120, 0x10b4,
0x0000, 0xef4c, 0xdee0, 0xcf06, 0xc002, 0xb216, 0xa57f, 0x9a75,
0x9128, 0x89c1, 0x845f, 0x811b, 0x8002, 0x811b, 0x845f, 0x89c1,
0x9128, 0x9a76, 0xa57f, 0xb216, 0xc002, 0xcf06, 0xdee0, 0xef4c
};
DSK6713_AIC23_Config config = { \
0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Left line input channel volume */ \
0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume */\
0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */ \
0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */ \
// 0x0014 micin with 0dB boost
0x0014, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */ \
0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */ \
0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */ \
0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \

0x008c, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */ \

0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \
};

DSK6713_AIC23_CodecHandle hCodec;
Int32 InitWait =1;
Int32 data;
Int32 Logger[1024];
Int32 LoggerIndex =0;

float in_buffer[100];

main(){
int i;
LED=0x0;

// Filter Initialization
for( i = 0 ; i < 100; i++ ) in_buffer[i]=0.0;

// Initialize codec
hCodec = DSK6713_AIC23_openCodec(0, &config);
IRQ_globalEnable();
IRQ_enable(IRQ_EVT_RINT1);
IRQ_enable(IRQ_EVT_XINT1);
}

void led(){
static int cc = 1;
LED = cc;

// To Shift Pattern
if (cc == 0x03) cc = 0x05;
else if (cc == 0x05) cc = 0x06;
else if (cc == 0x06) cc = 0x03;
else cc = 0x03;
//To count Binary
//cc++; if (cc>0x07) cc = 0x00;

// TO Toggle LED
// *cc ^= 0x07; if ((cc !=0x00) && (cc !=0x07)) cc = 0x07;
}

setpll200M(){

void read(){
int i = 0;
float result;
data=MCBSP_read(DSK6713_AIC23_DATAHANDLE);

if(data>=32768) data= data|0xffff0000;

for( i = 0; i <= 29; i++) in_buffer[i] = in_buffer[i+1];
in_buffer[30]=((float)(data))/16;

result = 0;
for( i = 0 ; i <= 30; i++ ) result += filter_Coeff[i] * in_buffer[i];
data = (Int32)(result*512);
//data = (Int32)(in_buffer[30]*16);
Logger[LoggerIndex++] = data;
if (LoggerIndex == 1024)
LoggerIndex = 0;
}

void write(){
if (InitWait<1000){
InitWait++;
MCBSP_write(DSK6713_AIC23_DATAHANDLE, 0);
MCBSP_write(DSK6713_AIC23_DATAHANDLE, 0);
}
else{
MCBSP_write(DSK6713_AIC23_DATAHANDLE, data);
MCBSP_write(DSK6713_AIC23_DATAHANDLE, data);
}
}

Output:

4. Enter the source code and save the file with main.c extension.
5. Right click on source, Select add files to project and Choose main.c file Saved before.

6. Add the other supporting .c files which configure the audio codec.
7. 7. a) Go to Project to Compile.
b) Go to Project to Build.
c) Go to Project to Rebuild All.
 It will shows
warnings and
errors if any

8. Go to file and load program and load “.out” file into the board.

9. Go to Debug and click on run to run the program.

10. To see the Graph go to View and select time/frequency in the Graph and give the
correct Start address provided in the program, Display data can be taken as per user.
11. Green line is to choose the point, Value at the point can be seen (Highlighted by
circle at the left corner).

12. In the graph, chose FFT magnitude as display type we will get Graph B
Graph 2:FFT magnitude of FIR filter.

13. The impulse response of FIR filters.

 Experiment 6: Noise Removal
Procedure to create new Project:
1. To create project, Go to Project and Select New.

2. Give project name and click on finish.

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).
3. Click on File New Source File, To write the Source Code.
AIM: Noise removal in a given mixed signal.
Noise removal: In the real time systems every signal will get corrupted by the noise. To
analyze the noise we need to add the noise, a noise will be generated and added to the
signal. A noise of 3KHz is added to a tone of 400KHz then the noise is removed by using
an high pass filter.

Program:
#include <stdio.h>
#include "c6713dsk.h"
#include "master.h"
#include "aic23cfg.h"
#include "dsk6713_aic23.h"
#include <std.h>
#include <swi.h>
#include <log.h>
#include <c6x.h>
#include <csl.h>
#include <csl_mcbsp.h>

/* Length of sine wave table */

#define SINE_TABLE_SIZE 48

// Delta
/*float filter_Coeff[] ={0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,
0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,0.00,
0.00,0.00,0.00,0.00,0.00,0.00,1.00,0.00};*/

// Low Pass Filter

/*float filter_Coeff[] ={2.715297634171146e-003, 2.315819802942924e-004,-
1.244493581373643e-002, 7.244364917221401e-003,1.207341716154354e-002,
2.734134166585232e-003,
-1.941706440790678e-002, -1.729098218843226e-002,1.773008730568675e-002,
4.091495174059349e-002,2.113436751136944e-003, -6.788468549771730e-002,
-6.059440700570791e-002, 8.970313256448266e-002,3.014572949374625e-001,
4.019009454299968e-001,3.014572949374625e-001, 8.970313256448266e-002,
-6.059440700570791e-002, -6.788468549771730e-002,2.113436751136944e-003,
4.091495174059349e-002,1.773008730568675e-002, -1.729098218843226e-002,
-1.941706440790678e-002, 2.734134166585232e-003,1.207341716154354e-002,
7.244364917221401e-003,-1.244493581373643e-002, 2.315819802942924e-004,
2.715297634171146e-003};*/

// High Pass Filter

float filter_Coeff[] ={3.294316420702696e-004,3.800020076486443e-
003,9.822200806739014e-003,1.517265313889167e-002,
1.323547007544908e-002,2.635896986048919e-004,-1.808215737734512e-002,-
2.666833013269758e-002,
-1.155354962270025e-002,2.448211866656400e-002,5.534101055783895e-
002,4.424359087198896e-002,
-2.922329551555757e-002,-1.473332022689261e-001,-2.574625659073934e-
001,6.976203109125493e-001,
-2.574625659073934e-001,-1.473332022689261e-001,-2.922329551555757e-
002,4.424359087198896e-002,
5.534101055783895e-002,2.448211866656400e-002,-1.155354962270025e-002,-
2.666833013269758e-002,
-1.808215737734512e-002,2.635896986048919e-004,1.323547007544908e-
002,1.517265313889167e-002,
9.822200806739014e-003,3.800020076486443e-003,3.294316420702696e-004};

// Pre-generated sine wave data, 16-bit signed samples

int sinetable[SINE_TABLE_SIZE] = {
0x0000, 0x10b4, 0x2120, 0x30fb, 0x3fff, 0x4dea, 0x5a81, 0x658b,
0x6ed8, 0x763f, 0x7ba1, 0x7ee5, 0x7ffd, 0x7ee5, 0x7ba1, 0x76ef,
0x6ed8, 0x658b, 0x5a81, 0x4dea, 0x3fff, 0x30fb, 0x2120, 0x10b4,
0x0000, 0xef4c, 0xdee0, 0xcf06, 0xc002, 0xb216, 0xa57f, 0x9a75,
0x9128, 0x89c1, 0x845f, 0x811b, 0x8002, 0x811b, 0x845f, 0x89c1,
0x9128, 0x9a76, 0xa57f, 0xb216, 0xc002, 0xcf06, 0xdee0, 0xef4c
};
DSK6713_AIC23_Config config = { \
0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Left line input channel volume */ \
0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume */\
0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */ \
0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */ \
// 0x0014 micin with 0dB boost
0x0014, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */ \
0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */ \
0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */ \
0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \

0x008c, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */ \

0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \
};

DSK6713_AIC23_CodecHandle hCodec;
Int32 InitWait =1;
Int32 data;
Int32 Logger[1024];
Int32 LoggerIndex =0;

float in_buffer[100];

main(){
int i;
LED=0x0;

// Filter Initialization
for( i = 0 ; i < 100; i++ ) in_buffer[i]=0.0;

// Initialize codec
hCodec = DSK6713_AIC23_openCodec(0, &config);
IRQ_globalEnable();
IRQ_enable(IRQ_EVT_RINT1);
IRQ_enable(IRQ_EVT_XINT1);
}

void led(){
static int cc = 1;
LED = cc;

// To Shift Pattern
if (cc == 0x03) cc = 0x05;
else if (cc == 0x05) cc = 0x06;
else if (cc == 0x06) cc = 0x03;
else cc = 0x03;
//To count Binary
//cc++; if (cc>0x07) cc = 0x00;

// TO Toggle LED
// *cc ^= 0x07; if ((cc !=0x00) && (cc !=0x07)) cc = 0x07;
}

setpll200M(){

void read(){
int i = 0;
float result;
data=MCBSP_read(DSK6713_AIC23_DATAHANDLE);

if(data>=32768) data= data|0xffff0000;

for( i = 0; i <= 29; i++) in_buffer[i] = in_buffer[i+1];
in_buffer[30]=((float)(data))/16;

result = 0;
for( i = 0 ; i <= 30; i++ ) result += filter_Coeff[i] * in_buffer[i];
data = (Int32)(result*512);
//data = (Int32)(in_buffer[30]*16);
Logger[LoggerIndex++] = data;
if (LoggerIndex == 1024)
LoggerIndex = 0;
}

void write(){
if (InitWait<1000){
InitWait++;
MCBSP_write(DSK6713_AIC23_DATAHANDLE, 0);
MCBSP_write(DSK6713_AIC23_DATAHANDLE, 0);
}
else{
MCBSP_write(DSK6713_AIC23_DATAHANDLE, data);
MCBSP_write(DSK6713_AIC23_DATAHANDLE, data);
}
}

Output:
4. Enter the source code and save the file with main.c extension.

5. Right click on source, Select add files to project and Choose main.c file Saved before.
6. Add the other supporting .c files which configure the audio codec.

7. 7. a) Go to Project to Compile.
b) Go to Project to Build.
c) Go to Project to Rebuild All.
 It will shows
warnings and
errors if any

8. Go to file and load program and load “.out” file into the board.
9. Go to Debug and click on run to run the program.
10. To see the Graph go to View and select time/frequency in the Graph and give the
correct Start address provided in the program, Display data can be taken as per user.

11. Green line is to choose the point, Value at the point can be seen (Highlighted by
circle at the left corner).
 Experiment 7: Impulse Response
Procedure to create new Project:
1. To create project, Go to Project and Select New.

2. Give project name and click on finish.

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).

3. Click on File New Source File, To write the Source Code.

AIM: To find Impulse response of a first order and second order system.
A discrete time system performs an operation on an input signal based on predefined
criteria to produce a modified output signal. The input signal x(n) is the system
excitation, and y(n) is the system response. The transform operation is shown as,

x(n) y(n) = T[(x(n)]

T

The convolution sum can be represented by, y(n) = x(n) * h(n)

For Example let’s find out an impulse response of a difference equation.

The general form of difference equation is,

M M
y(n) = ∑ak y(n-k) + ∑bk x(n-k)
k=1 k=0

Find out the impulse response of second order difference

equation.
Program:
#include<stdio.h>
#define Order 2
#define Len 5
float h[Len] = {0.0,0.0,0.0,0.0,0.0},sum;
void main()
{
int j, k;
float a[Order+1] = {0.1311, 0.2622, 0.1311};
float b[Order+1] = {1, -0.7478, 0.2722};

for(j=0; j<Len; j++)

{
sum = 0.0;
for(k=1; k<=Order; k++)
{
if ((j-k) >= 0)
sum = sum+(b[k]*h[j-k]);
}
if (j <= Order)
h[j] = a[j]-sum;
else
h[j] = -sum;
printf (" %f ",h[j]);
}
}

Output:
0.131100 0.360237 0.364799 0.174741 0.031373

For first order difference equation.

Program:
#include<stdio.h>
#define Order 1
#define Len 5
float h[Len] = {0.0,0.0,0.0,0.0,0.0},sum;

void main()
{
int j, k;
float a[Order+1] = {0.1311, 0.2622};
float b[Order+1] = {1, -0.7478};
for(j=0; j<Len; j++)
{
sum = 0.0;
for(k=1; k<=Order; k++)
{
if((j-k)>=0)
sum = sum+(b[k]*h[j-k]);
}
if(j<=Order)
h[j] = a[j]-sum;
else
h[j] = -sum;
printf("%f ", j, h[j]);
}
}

Output:
0.131100 0.360237 0.269385 0.201446 0.150641

4. Enter the source code and save the file with “.C” extension.
5. Right click on source, Select add files to project .. and Choose “.C “ file Saved before.
6. Right Click on libraries and select add files to Project.. and choose
C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib and click open.
7. a) Go to Project to Compile.
b) Go to Project to Build.
 c) Go to Project to Rebuild All.

It shows errors
and warnings
here.

8. Go to file and load program and load “.out” file into the board..
9. Go to Debug and click on run to run the program.
10. Observe the output in output window.
Output will
display
here
11. To see the Graph go to View and select time/frequency in the Graph, and give the
correct Start address or the output variable name provided in the program, Display data
can be taken as per user. Select DSP data type as 32-bit floating point.
12. Green line is to choose the point, Value at the point can be seen (Highlighted by
circle at the left corner).