Signal Processing Laboratory Manual BEC502 BMS Institute of Technology and Management

BMS INSTITUTE OF TECHNOLOGY AND

MANAGEMENT
Avalahalli, Doddabalapur Main Road, Yelahanka, Bengaluru:- 560 064.

DEPARTMENT OF ELECTRONICS AND COMMUNICATION

ENGINEERING

Department Vision

Be a pioneer in providing quality education in Electronics,

Communication and Allied Engineering fields to Serve as
Valuable Resource for Industry and Society.

Department Mission

1. Impart Sound Theoretical Concepts and Practical Skills

through innovative pedagogy.
2. Promote Inter-disciplinary Research.
3. Inculcate Professional Ethics.

Digital Signal Processing Laboratory Manual (BEC502)

V SEM ECE 2024-25

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BMSIT&M COLLEGE ENTRANCE ECE DEPARTMENT BLOCK

Vision of the Institute

To emerge as one of the finest technical institutions of higher learning

to develop engineering professionals who are technically
competent, ethical and environment-friendly for betterment of society.

Mission of the Institute

Accomplish stimulating learning environment through high-quality

academic instruction, innovation and industry-institute interface.

Lab Course co-ordinators (2024-25) Laboratory Instructor

1. Dr. Vijayalakshmi G V, Associate Professor, ECE Mr. Byrareddy A B

2. Mamatha K R, Assistant Professor, ECE

Contact Details: Office: 080-28561576, 080-28567187

Fax: 080-28567186

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B.E ELECTRONICS AND COMMUNICATION ENGINEERING

Choice Based Credit System (CBCS)
SEMESTER - V
Digital Signal Processing (3:0:1) 4
(Effective from the Academic Year 2024-25 for 2022 Scheme)
Course Code BEC502 CIE Marks 50

Teaching Hours/Week (L:T:P) 3:0:1 SEE Marks 50

Total Number of Contact Hours 40 Theory+8-10 lab Exam Hours 03
Course objectives:
This course will enable students to:
1. Understand the need for and importance of mathematical tools such as discrete Fourier
Transforms (DFT) and Fast Fourier Transform (FFT) to analyze the signal.
2. Apply DFT properties on various signals.
3. Use FFT algorithms to eliminate redundant calculation and enable to analysis of the
spectral properties of a signal.
4. Design analog and digital filters
5. Realize IIR and FIR filters using direct forms, cascade and parallel forms
Preamble:
Digital signal processing is continuation of Basic signal processing of 4th semester which starts
with DFT and its properties, Decimation in time FFTs. The foundations of digital filter (design of
IIR and FIR filter) design and realization are built up. Practice Problems with solutions with
applications of DSP.
Module – 1
Introduction: Introduction to DSP systems, Discrete Fourier Transforms (DFT):
Frequency domain sampling and Reconstruction of Discrete Time Signals, The Discrete
Fourier Transform Properties of the DFT: Periodicity, Linearity, shifting property, reversal, Parsvels
theorem, Correlation, Symmetry properties (statements only) and Circular Convolution. (9 Hours)
Module – 2
Linear filtering methods based on the DFT: Use of DFT in Linear Filtering, Filtering of Long Data
Sequences.
Fast-Fourier-Transform (FFT) algorithms: Efficient Computation of the DFT: Radix-2 FFT
algorithms for the computation of DFT and IDFT–decimation-in-time. (7 Hours)
Module – 3
IIR Filter Design: Analog Butterworth Filters, Frequency transformation in Analog domain, Analog
Filters using Lowpass prototype transformation. Design of IIR Digital Filters from Analog Filter:
Bilinear Transformation and Frequency Warping, Bilinear Transformation Design Procedure,
Digital Butterworth Filter Design using BLT. (8 Hours)
Module – 4
Design of FIR Filters: Characteristics features of FIR filter, Symmetric and Antisymmetric FIR filters,
Design of Linear-phase FIR filters using windows - Rectangular, Hamming and Hanning, (7 Hours)

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Module – 5
Realization of IIR & FIR Filters: Direct form I and Direct form II realization of an IIR filter, Cascade
realization of an IIR filter, Parallel realization of an IIR filter, Direct form I realization of FIR filter,
Lattice realization of FIR filter.
Practical component of IPCC
Experiments to be executed using MATLAB/SCILAB/OCTAVE
1.Verification of Sampling theorem
2.Implementation of DFT using built-in function FFT and user defined function.
3.To perform Circular convolution of two given signals.
4.To perform Auto and Cross Correlation of two given signals
5.To find impulse response of a system
6. IIR filter implementation
7.FIR filter implementation
8. DSP kit experiments for performing linear and circular convolution
9.DSP kit experiment to compute DFT of given signal.
10. DSP kit experiment to find impulse response of a system.
11 DSP kit experiment to add and cancel the noise

Textbooks:
1. Proakis & Monalakis, “Digital signal processing Principles Algorithms & Applications”,
4th Edition, Pearson Education, New Delhi,2007.

References:
1. Sanjit K Mitra, “Digital Signal Processing, A Computer Based Approach”, 4 th Edition, McGraw
Hill Education, 2013.
2. Oppenheim & Schaffer, “Discrete Time Signal Processing”, PHI,2003.
3. D. Ganesh Rao and Vineeth P, “Digital Signal Processing” Gejji, Cengage India Private Limited,
2017.

Course outcomes: The students will be able to:

CO1: Apply the knowledge of digital signal processing to find Fourier transforms of various
signals.
CO2: Analyze the Discrete Fourier Transform (DFT) concept to solve the various signal
processing
CO3: Design and compare different types of filter characteristics and their applications.

CO4: Work in a team, to demonstrate the usage of MATLAB/Octave in signal processing.

Alternate Assessment Tools (AATs) suggested:

Mini project related to Digital signal processing using any modern tools beyond the syllabus which
illustrates knowledge gain in the course and implementation of the same.

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Web links/e-resources:
https://onlinecourses.nptel.ac.in/noc21_ee20/
MATLAB

MATLAB (matrix laboratory) is a programming language developed by MathWorks. It

started out as a matrix programming language where linear algebra programming was
simple. It can be run both under interactive sessions and as a batch job. MATLAB is
a paradigm numerical environment and proprietary programming language developed
by Math Works.
MATLAB allows matrix manipulations, plotting of functions and data, implementation
of algorithms, creation of user interfaces, and interfacing with programs written in other
languages, including C, C++, C#, Java, Fortran and Python.

Although MATLAB is intended primarily for numerical computing, an optional toolbox uses
the MuPAD symbolic engine, allowing access to symbolic computing abilities. An additional
package, Simulink, adds graphical multi-domain simulation and model-based
design for dynamic and embedded systems.

As of 2017, MATLAB has roughly one million users across industry and academia. MATLAB
users come from various backgrounds of engineering, science, and economics. There are
various toolboxes in MATLAB like signal processing toolbox, image processing toolbox,
neural network toolbox, statistics toolbox, financial toolbox, Bioinformatics toolboxes.

The Matlab screen has 4 important windows along with Editor (where programs can be
written and stored). These are Command window where we type commands and
outputs/error messages are seen, Command history window where past commands are
remembered (we can re-run any command by clicking on the commands), Workspace
window which stores all the variables used and Current Directory window shows the files
in current directory.

In this semester students are required to register in Math works website using their official
email ID and download the MATLAB software from the following link.

https://www.mathworks.com/academia/tah-portal/bms-institute-of-technology-
31463638.html

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The installation might take minimum of 15-20 GB of your internet data as well as
same amount of system memory.

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FIG 1 MATLAB WINDOWS

FIG 2 SCREEN COMPONENTS

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EXPERIMENTS USING MATLAB/OCTAVE

EXPERIMENT NO 1
Verification of Sampling Theorem

Aim: To verify Sampling theorem for a signal of given frequency.

Theory:
● Sampling is a process of converting a continuous time signal (analog signal)
x(t) into a discrete time signal x[n], which is represented as a sequence of
numbers. (A/D converter)
● Converting back x[n] into analog (resulting in x (t ) is the process of
reconstruction. (D/A converter)
● Techniques for reconstruction-(i) ZOH (zero order hold) interpolation results
in a staircase waveform, is implemented by MATLAB plotting function
stairs (n, x), (ii) FOH (first order hold) where the adjacent samples are joined
by straight lines is implemented by MATLAB plotting function plot (n,
x),(iii) spline interpolation, etc.
● For x (t ) to be exactly the same as x(t), sampling theorem in the generation
of x(n) from x(t) is used. The sampling frequency fs determines the spacing
between samples.
● Aliasing-A high frequency signal is converted to a lower frequency, results
due to under sampling. Though it is undesirable in ADCs, it finds practical
applications in stroboscope and sampling oscilloscopes.

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Algorithm:
1. Input the desired frequency fmax (for which sampling theorem is to be
verified).
2. Generate an analog signal xt of frequency fmax for comparison.
3. Generate over sampled, Nyquist & Under sampled discrete time signals.
4. Plot the waveforms and hence prove sampling theorem.

MATLAB Implementation:

5. Step 1: MATLAB can generate only discrete time signals. For an

approximate analog signal xt, choose the spacing between the samples to be
very small (≈0), say 50µs = 0.00005. Next choose the time duration, say xt
exists for 0.05seconds.(final in program) (for low frequency say <1000 Hz
choose 0.05 sec, for higher choose 0.01 secs or lesser as appropriate).
Now begin with the vector that represents the time base
t = 0:0.00005:0.05;
The colon (:) operator in MATLAB creates a vector, in the above case a time vector
running from 0 to 0.05 in steps of 0.00005. The semicolon (;) tells MATLAB not
to display the result.
Given t, the analog signal xt of frequency fd is generated as
(sin(ωt)=sin(2πft))-
xt=sin(2*pi*fmax*t);
pi is recognized as 3.14 by MATLAB.

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Step 2: To illustrate oversampling condition, choose sampling frequency

fs0=2.2*fd. For this sampling rate T0=1/fs0, generate the time vector as n1 =
0:T0:0.05; & over sampled discrete time signal x1=sin(2*pi*fmax*n1);
[Alternately let n1 be a fixed number of samples, say n=0:10; &
x1=sin(2*pi*n*fd/fs0);]
The discrete signal is converted back to analog signal (reconstructed) using the
MATLAB plot function (FOH). In one plot both the analog signal (approximate
one in blue color) and the reconstructed signal (in red) are plotted for comparison

Step 3: Repeat step 2 for different sampling frequencies, i.e., fs=1.3*fd&fs=2*fd

for under sampling and Nyquist sampling conditions.

MATLAB Program:

tfinal=0.05;
t=0:0.00005: tfinal;
fmax=input('Enter analog frequency');

%Define analog signal for comparison

xt=sin(2*pi*fmax*t);

%Simulate condition for under sampling i.e., fs1<2*fmax

fs1=1.3*fmax;
%define the time vector
n1=0:1/fs1: tfinal;
%Generate the under sampled signal

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xn1=sin(2*pi*n1*fmax);
%plot the analog & sampled signals
subplot(3,1,1);
plot(t,xt,'b',n1,xn1,'r*-');
title('Under sampling plot');

%Condition forNyquistplot
fs2=2*fmax;
n2=0:1/fs2:tfinal;
xn2=sin(2*pi*fmax*n2);
subplot(3,1,2);
plot(t,xt,'b',n2,xn2,'r*-');
title('Nyquist plot');
%Condition for Oversampling
fs3=10*fmax;
n3=0:1/fs3:tfinal;
xn3=sin(2*pi*fmax*n3);
subplot(3,1,3);
plot(t,xt,'b',n3,xn3,'r*-');
title('Oversampling plot');
xlabel('time');
ylabel('amplitude');
legend('analog','discrete')

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Result:
Enter analog frequency 200

Output Waveforms
Plots of a sampled sine wave of 200Hz

Students are expected to use interp1q () function to connect the samples in the
sampled signal.

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Plots of a sampled cosine wave of 200Hz

Inference:
1. From the under-sampling plot observe the aliasing effect. The analog signal
is of 200Hz (T=0.005s). The reconstructed (from under sampled plot) is of a
lower frequency. The alias frequency is computed as fmax-fs1 = 200-1.3*200
= 200-260= -60Hz
This is verified from the plot. The minus sign results in a 180˚ phase
shift.
(For example: 3kHz& 6kHz sampled at 5kHz result in aliases of -2kHz (3k-
5k) & 1kHz (6k-5k) respectively)
2. Sampling at the Nyquist rate results in samples sin(πn) which are identically
zero, i.e., we are sampling at the zero crossing points and hence the signal

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component is completely missed. This can be avoided by adding a small

phase shift to the sinusoid. The above problem is not seen in cosine
waveforms (except cos(90n)). A simple remedy is to sample the analog
signal at a rate higher than the Nyquist rate. The Fig1.2 shows the result due
to a cosine signal (x1=cos(2*pi*fmax*n1);
3. The over sampled plot shows a reconstructed signal almost similar to that of
the analog signal. Using low pass filtering the wave form can be further
smoothened.

Open ended Experiments:

Generate sine/cosine waveforms of multiple frequencies, say

x1=sin(2*pi*fd1*n1)+ sin(2*pi*fd2*n2); where fd1, fd2 are 2 different
frequencies

VIVA Questions
1. Define plot, stem, and subplot in MATLAB.
2. List all the functions used in the above program.
3. Briefly explain sampling theorem.
4. What is aliasing?

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EXPERIMENT NO 2
Computation of N point DFT of a Given Sequence and to Plot Magnitude
and Phase Spectrum.

Aim: To compute DFT of the given sequence.

Theory:
● Discrete Fourier Transform (DFT) is used for performing frequency analysis
of discrete time signals. DFT gives a discrete frequency domain
representation whereas the other transforms are continuous in frequency
domain.
● The N point DFT of discrete time signal x[n] is given by the equation
N-1 − j2π kn
X ( k )= ∑ x [n ]e N
; k= 0,1,2, .. .. N-1
n= 0

Where N is chosen such that N≥ L , where L=length of x[n].

● The inverse DFT allows us to recover the sequence x[n] from the frequency
samples.
N−1 j2π kn
1
x [n ]=
N
∑ x [n ]e N
; n= 0,1,2, .. . .N-1
k= 0

● X(k) is a complex number (remember ejw=cosw + jsinw). It has both

magnitude and phase which are plotted versus k. These plots are magnitude
and phase spectrum of x[n]. The ‘k’ gives us the frequency information.
● Here k=N in the frequency domain corresponds to sampling frequency (fs).
Increasing N increases the frequency resolution, i.e., it improves the spectral
characteristics of the sequence. For example, if fs=8 kHz and N=8 point
DFT, then in the resulting spectrum, k=1 corresponds to 1 kHz frequency.
For the same fs and x[n], if N=80 point DFT is computed, then in the

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resulting spectrum, k=1 corresponds to 100Hz frequency. Hence, the

resolution in frequency is increased.
● Since N≥ L , increasing N to 8 from 80 for the same x[n] implies x[n] is still
the same sequence (<8), the rest of x[n] is padded with zeros. This implies
that there is no further information in time domain, but the resulting spectrum
has higher frequency resolution. This spectrum is known as ‘high density
spectrum’ (resulting from zero padding x[n]). Instead of zero padding, for
higher N, if more number of points of x[n] are taken (more data in time
domain), then the resulting spectrum is called a ‘high resolution spectrum’.
Algorithm:
1. Input the sequence for which DFT is to be computed.
2. Input the length of the DFT required (say 4, 8, >length of the sequence).
3. Compute the DFT using the ‘fft’ command.
4. Plot the magnitude & phase spectra.

MATLAB Implementation:
MATLAB has an inbuilt function ‘FFT’ which computes the Discrete Fourier
transform.
FFT(X) is the discrete Fourier transform (DFT) of vector X. For length N input
vector x, the DFT is a length N vector X, with elements N. FFT(X,N) is the N-point
FFT, padded with zeros if X has less than N points and truncated if it has more.
The magnitude spectrum is computed using the function ABS Absolute value.
ABS(X) is the absolute value of the elements of X. When X is complex, ABS(X)
is the complex modulus (magnitude) of the elements of X.
The phase spectrum is computed using the function ANGLE Phase angle.
ANGLE (H) returns the phase angles, in radians, of a matrix with complex
elements.

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Matlab Program:
N-point FFT and IFFT of a given sequence

clc;
clear all;
close all;
x=input ('Enter the sequence x (n)');
N=length(x);
y=fft(x,N);
disp(y);
mag=abs(y);
phase=angle(y);
subplot(2,2,1);
stem(x);
grid on;
title('Input Data Sequence x(n)');
subplot(2,2,3);
stem(mag);
grid on;
title('Magnitude Plot');
subplot(2,2,2);
stem(phase);
grid on;
title('Phase plot');
z=ifft(y,N);
mag1=real(z);

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subplot(2,2,4);
stem(mag1);
grid on;
title('Signal Sequence constituted from Spectrum');

Result:
Enter the sequence x(n) [0.5 0.5 0.5 0.5 0 0.5 0.5 0.5]
3.5000 0.5000 -0.5000 0.5000 -0.5000 0.5000 -0.5000 0.5000
Output waveform
Input Data Phase
0 Sequence x(n) 4 plot
.0 3
.0 2
.0 1
.0 00
0 2 4 6 8 2 4 6 8
Magnitude Signal Sequence constituted from
4 0
Spectrum
Plot
3 .0

2 .0

1 .0

00 .0
2 4 6 8 0 2 4 6 8

DFT of a given sequence

To find the N-point DFT of a sequence using a function file dft
clc:
xn=input(‘enter the sequence x(n)’);
N=input(‘enter the number of points of computation’);
y=dft(xn,N);
disp(y);
amp=abs(y);

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ph=angle(y);
subplot(2,1,1)
stem(amp);
xlabel (‘MAGNITUDE’);
ylabel (‘K’);
title (‘magnitude plot’);
stem (ph*180/pi);
xlabel(‘phase’);
ylabel(‘K’);
title(‘phase plot’);

%function to find dft

function[xk] =dft(xn,N)
n=[0:1:N-1];
k=[0:1:N-1];
WN=exp(-j*2*pi/N);
nk=n`*k;
WNnk=WN.^nk;
xk=xn*WNnk;
Result:
Enter the sequence x(n)=[1 1 1 1 0 0 0 0]
Xk=4.000 1.0000-2.4142i -0.0000-0.0000i 0.0000-0.0000i 1.0000+0.4142i
0.0000-0.0000i

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IDFT of a given sequence

To find the N-point IDFT of a sequence using a function file idft

clc:
xk=input(‘enter the sequence x(k)’);
N=input(‘enter the number of points of computation’);
y=idft(xk,N);
disp(y);
amp=abs(y);
ph=angle(y);
subplot(2,1,1)
stem(amp);
xlabel(‘MAGNITUDE’);
ylabel(‘K’);
title(‘magnitude plot’);
stem(ph*180/pi);
xlabel(‘phase’);
ylabel(‘K’);
title(‘phase plot’);

%function to find idft

function[xn] =idft(xk,N)
n=[0:1:N-1];
k=[0:1:N-1];
WN=exp(-j*2*pi/N);
nk=n`*k;

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WNnk=WN.^(-nk);
xn=(xk*WNnk)/N;
Result:
Enter the sequence Xk=[4, 1-2.4142i ,0,1-0.4142i,0, 1.0000+0.4142i ,
0,1+2.4142i]
x(n)=[1 1 1 1 0 0 0 0]

Inputs -
Run1: Give 4 point sequence and find 4 point DFT
Run2: Give 8 point sequence and find 8 point DFT
Run 3: Give a Sinusoidal Signal with a frequency 250 Hz with fs = 8 KHz
and sample it for 64 points and perform 64 point DFT on it.
Run 4: Give a Sinusoidal Signal with a frequency 250 Hz and 500 Hz with fs
= 8 KHz and sample it for 64 point and perform 64 point DFT on it.
Open ended Experiments :

1. Find the 4 point DFT of x[n]=cosnπ/4

2. Find the 3 point IDFT of X[K]=[1-j2,-1,1+j2]

VIVA Questions:
1. Explain DFT
2. What are the methods of finding DFT
3. Explain differences between DFT and FFT
4. Explain how functions can be declared in MATLB.
5. Explain MATLAB function fft(x)
6. How can user create a MATLAB function

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EXPERIMENT NO 3
A. Circular Convolution of Two Given Sequences

Aim: To obtain circular convolution of two finite duration sequences.

Theory:
● As seen in the last experiment, the output y[n] of a LTI (linear time invariant)
system can be obtained by convolving the input x[n] with the system’s
impulse response h[n].
● The above linear convolution is generally applied to periodic sequences.
Whereas the Circular Convolution is used to study the interaction of two
signals that are periodic.
● The linear convolution sum is
+ +
y[n] = x[n]  h[n] =  x[k]h[n − k] =  x[n − k]h[k]
k=− k=− .
● To compute this equation, the linear convolution involves the
following operations:

o Folding- fold h[k] to get h[-k] ( for y[0])

o Multiplication – vk[n] = x[k] × h[n-k] (both sequences are multiplied
sample by sample)
o Addition- Sum all the samples of the sequence vk[n] to obtain y[n]
o Shifting – the sequence h[-k] to get h[n-k] for the next n.
+∞
y [ n ]= x [ n ](N )h [n ]= ∑ x [ k ]h[ 〈n− k 〉N ]
● The circular convolution sum is k= − ∞

where the index 〈n− k 〉N implies circular shifting operation and

〈− k 〉N
implies folding the sequence circularly.

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Steps for circular convolution are the same as the usual convolution, except all
index calculations are done "mod N" = "on the wheel".
o Plot f [m] and h [−m] as shown in Fig. 4.1. (use f(m)
instead of x(k))
o Multiply the two sequences
o Add to get y[m]
o "Spin" h[−m] n times Anti Clockwise (counter-
clockwise) to get h[n-m].
x[n] and h[n] can be both finite or infinite duration sequences. If infinite sequences,
they should be periodic, and the N is chosen to be at least equal to the period. If
they are finite sequences N is chosen as >= to max(xlength, hlength). Whereas in
linear convolution N>= xlength+hlength-1.
● Say x[n]={-2,3,1,1} and N = 5, then the x[-1] and x[-2] samples are plotted
at x[N-1] = x[-4] and x[N-2] =x[-3] places. Now x[n] is entered as x[n] = {1,
1, 0,-2,3}.

Fig. 4.1 Plotting of f(m) and h(-m) for circular convolution

Algorithm:
1. Input the two sequences as x and h.
2. Circularly convolve both to get output y.
3. Plot the sequences.

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MATLAB Implementation:
MATLAB recognizes index 1 to be positive maximum. Index 0 is not
recognized. Hence in the below program wherever y, x and h sequences are
accessed, the index is added with +1. The modulo index calculation for circular
convolution is carried out using the function - MOD Modulus (signed remainder
after division). MOD( x, y) is x - y.*floor(x./y) if y ~= 0. By convention, MOD(x,0)
is x. The input x and y must be real arrays of the same size, or real scalars. MOD( x,
y) has the same sign as y while REM( x, y) has the same sign as x. MOD( x, y) and
REM( x, y) are equal if x and y have the same sign, but differ by y if x and y have
different signs.

MATLAB Program:
Program to find circular convolution of two sequences by padding
zeroes
clc;
clear all;
close all;
g=input('Enter the first sequence');
h=input('Enter the second sequence');
N1=length(g);
N2=length(h);
N=max(N1,N2);
N3=N1-N2;
if(N3>=0)
h=[h,zeros(1,N3)];
else

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g=[g,zeros(1,-N3)];
end
for n=0:N-1
y(n+1)=0;
for k=0:N-1
i=mod((n-k),N); %calculation of x index
if i<0
i=i+N;
end %end of ‘if’
y(n+1)=y(n+1)+h(k+1)*g(i+1);
end %end of inner ‘for loop’
end %end of inner ‘for loop’
disp('The resultant signal is');y
figure(1);
subplot(3,1,1);
stem(g);
title('The first sequence is');
grid on;
subplot(3,1,2);
stem(h);
title('The second sequence is');
grid on;
subplot(3,1,3);
stem(y);
title('The circularly convolved sequence is');
grid on;

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Result:
Enter the first sequence [1 1 1 1]
Enter the second sequence [1 2 3 4]
The resultant signal is
y =10 10 10 10

Output Waveform

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2 Circular convolution using Matrix method

clc;
clear all;
close all;
X=input('ENTER THE FIRST SEQUENCE');
H=input('ENTER THE SECOND SEQUENCE IN MATRIX FORMAT');
Y=X*H;
disp(‘circular convolution’);
disp(Y);
stem(Y);
xlabel(‘n');
ylabel('Y(n)');
grid on;
Result:
Enter the first sequence [1 1 1 1]
Enter the second sequence in matrix format[1 4 3 2;2 1 4 3;3 2 1 4;4 3 2 1]
Circular convolution is: 10 10 10 10

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Open ended Experiments:

Find the circular convolution of x[n] = [1 3 5 6] and y[n] = [1 2 ]

VIVA Questions :
1. Write A Program to find the circular convolution of two sequences using
Matrix Method
2. Explain how circular convolution can be obtained.
3. Give the different methods for performing circular convolution.

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EXPERIMENT NO 4

A. Auto correlation of a Given Sequence and Verification of its

Properties.

Aim: To obtain autocorrelation of the given sequence and verify its

properties.

Theory:
● Correlation is mathematical technique which indicates whether 2 signals are
related and in a precise quantitative way how much they are related. A measure
of similarity between a pair of energy signals x[n] and y[n] is given by the cross
∞
r xy [l ]= ∑ x [ n ] y [n− l]; l= 0,± 1,± 2, . ..
correlation sequence rxy [l] defined by n= − ∞ .

● The parameter ‘l’ called ‘lag’ indicates the time shift between the pair.
● Autocorrelation sequence of x[n] is given by
∞
r xx [l ]= ∑ x [ n ]x [ n− l ]; l= 0,± 1,± 2, .. .
n= − ∞

● Some of the properties of autocorrelation are enumerated below

o The autocorrelation sequence is an even function i.e., r xx [l ]= rxx [− l]
o At zero lag, i.e., at l=0, the sample value of the autocorrelation sequence has its
maximum value (equal to the total energy of the signal εx) i.e.,
∞
r xx [l ]≤ r xx [0 ]= ε x= ∑ x 2[ n ]
n= − ∞ .

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This is verified in Fig. 5.1, where the autocorrelation of the rectangular pulse
(square) has a maximum value at l=0. All other samples are of lower value. Also
the maximum value = 11 = energy of the pulse [12+12+12..].
o A time shift of a signal does not change its autocorrelation sequence. For
example, let y[n]=x[n-k]; then ryy[l] = rxx[l] i.e., the autocorrelation of x[n] and
y[n] are the same regardless of the value of the time shift k. This can be verified
with a sine and cosine sequences of same amplitude and frequency will have
identical autocorrelation functions.
For power signals the autocorrelation sequence is given by
k
1
r xx [l ]= lim
2k+
∑ x[ n] x[n− l];l= 0,± 1,± 2,...
1 n= − k
k →∞ and for periodic signals with period N it is
N− 1
1
r xx [l ]=
N
∑ x [ n ] x [n− l ]; l= 0,± 1,± 2,. ..
n= 0 and this rxx[l] is also

o Periodic with N. This is verified in Fig. 5.3 where we use the periodicity
property of the autocorrelation sequence to determine the period of the periodic
signal y[n] which is x[n] (=cos(0.25*pi*n)) corrupted by an additive uniformly
distributed random noise of amplitude in the range [-0.5 0.5]

Algorithm:
1. Input the sequence as x.
2. Use the ‘xcorr’ function to get auto correlated output r.
3. Plot the sequences.

MATLAB Implementation:
MATLAB has the inbuilt function XCORR(A), when A is a vector, is the
auto-correlation sequence. If A is of length M vector (M>1), then the xcorr function

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returns the length 2*M-1 auto-correlation sequence. The zeroth lag of the
output correlation is in the middle of the sequence at element M.
XCORR(...,MAXLAG) computes the (auto/cross) correlation over the range
of lags: -MAXLAG to MAXLAG, i.e., 2*MAXLAG+1 lags. If missing, default is
MAXLAG = M-1.
[C,LAGS] = XCORR(...) returns a vector of lag indices (LAGS).

MATLAB Programs
%Computation of Autocorrelation of a rectangular Sequence
x = input ('Type in the reference sequence = ');
% Compute the correlation sequence
n = length(x)-1;
r = conv(x, fliplr(x));
disp('autocorrelation sequence r=');
disp(r);
%plot the sequences
subplot (2,1,1)
stem (x);
title ('square sequence');
subplot
(2,1,2)
k = -n: n;
stem (k, r);
title('autocorrelation output');
xlabel('Lag index');
ylabel('Amplitude');

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Result:
Autocorrelation sequence r = 1.0000 2.0000 3.0000 4.0000 5.0000
6.0000 7.0000 8.0000 9.0000 10.0000 11.0000 10.0000 9.0000
8.0000 7.0000 6.0000 5.0000 4.0000 3.0000 2.0000 1.0000

% Computation of Auto -correlation sequence using xcorr function

x = input('Type in the reference sequence = ');
% Compute the correlation sequence
[r,lag]=xcorr(x,x);
disp('auto correlation output is =');
disp(r);
stem(lag,r);
xlabel('Lag index');
ylabel('Amplitude');
Output Waveform

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Inference: Following Autocorr properties are verified

1) Max peak at zero lag (l=0)
2) The autocorr seq is an even function

Exercise: Find the autocorrelation of x[n] = cos (nPi/2) for n=0, 1, 2, 3

VIVA Questions
1. Explain the differences between convolution and correlation
2. List the methods for performing correlation
3. Which are the Mat lab functions used to perform correlation
4. Explain the following functions
a. rand(1,N)-0.5
b. fliplr(Y)
c. xcorr(Y)

B. Cross Correlation of a Given Sequence and Verification of its Properties.

Aim: To obtain cross correlation of the given sequence and verify its
properties.
Theory:
Cross Correlation has been introduced in the last experiment. Comparing the
equations for the linear convolution and cross correlation we find that
∞ ∞
r xy [l ]= ∑ x [ n ] y [n− l]= ∑ x [n ] y [− ( l− n )]= x [l ]∗ y [− l ]
n= − ∞ n= − ∞ . i.e., convolving the
reference signal with a folded version of sequence to be shifted (y[n]) results in
cross correlation output. (Use ‘fliplr’ function for folding the sequence for
correlation).

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● The properties of cross correlation are 1) the cross correlation sequence

sample values are upper bounded by the inequality

r xx [l ]≤ √
r xx [ 0 ]r yy [0 ]= √
εx εy

2) The cross correlation of two sequences x[n] and y[n]=x[n-k] shows a peak
at the value of k. Hence cross correlation is employed to compute the exact
value of the delay k between the 2 signals. Used in radar and sonar
applications, where the received signal reflected from the target is the
delayed version of the transmitted signal (measure delay to determine the
distance of the target).
3) The ordering of the subscripts xy specifies that x[n] is the reference
sequence that remains fixed in time, whereas the sequence y[n] is shifted
r yx [l ]= r xy [− l]
w.r.t x[n]. If y[n] is the reference sequence then . Hence ryx[l]
is obtained by time reversing the sequence rxy[l].

Algorithm:
1. Input the sequence as x and y.
2. Use the ‘xcorr’ function to get cross correlated output r.
3. Plot the sequences.
MATLAB Implementation:
MATLAB has the inbuilt function XCORR: Say C = XCORR(A,B), where A and
B are length M vectors (M>1), returns the length 2*M-1 cross-correlation sequence
C. If A and B are of different length, the shortest one is zero-padded. Using
convolution to implement correlation, the instruction is FLIPLR Flip matrix in
left/right direction. FLIPLR(X) returns X with row preserved and columns flipped
in the left/right direction. X = 1 2 3 becomes 3 2 1.

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MATLAB Programs
% Computation of Cross-correlation Sequence using folded sequence and
convolution

x = input('Type in the reference sequence = ');

y = input('Type in the second sequence = ');
% Compute the correlation sequence
n1 = length(x)-1;
n2 = length(y)-1;
r = conv(x,fliplr(y));
disp(‘Cross correlation output is =’);
disp(r);
k = (-n1):n2;%time vector for plotting
stem(k,r);
xlabel('Lag index');
ylabel('Amplitude');

Result:
Type in the reference sequence = [1 3 -2 1 2 -1 4 4 2]
Type in the second sequence = [2 -1 4 1 -2 3]
Cross correlation output is =
3 7 -11 14 13 -15 28 6 -2 21 12 12 6 4

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Output Waveforms

% Computation of Cross-correlation sequence using xcorr function

x = input('Type in the reference sequence = ');
y = input('Type in the second sequence = ');
% Compute the correlation sequence
[r,lag]=xcorr(x,y);
disp('Cross correlation output is =');
disp(r);
stem(lag,r);
xlabel('Lag index');
ylabel('Amplitude');
Result:
Type in the reference sequence = [1 3 -2 1 2 -1 4 4 2]
Type in the second sequence = [2 -1 4 1 -2 3]
Verification: with xcorr(x,y), the output is

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{-0.0000 , -0.0000 , 0.0000, 3.0000, 7.0000, -11.0000, 14.0000, 13.0000, -

15.0000, 28.0000, 6.0000, -2.0000, 21.0000, 12.0000, 12.0000, 6.0000 ,
4.0000}

%Verification of the properties

x1=[1,2,3,-1,0,0];
x2=[0,0,1,2,3,-1]; %=xr[n-2],i.e., k=2
>> xcorr(x1,x2)
ans = {-1,1,5,15,5,1-1,0,0}

Inference:
Strong peak of 15 at lag = -2 implies the delay between xi and x2 is 2.
Also peak =15=energy of xi (1+22+32+ (-1)2) implies that both xi and x2 are
strongly correlated.
%consider the below sequences
xr=[1,2,3,-1];
x2=[3,-1,1,2];
xcorr(xr,x2) ans ={2,5,7,2,2,10,-3}
xcorr(x2,xr) ans ={-3,10,2,2,7,5,2}

Inference:
Strong peak of 10 at lag = 2 implies the delay between xr and x2 is 2, but since
10<15, it implies that xr and x2 are uncorrelated slightly (may be due to noise, etc).
r yx [l ]= r xy [− l]
is verified.

Open ended Experiments: Find the cross correlation of x[n]=[1 2 0 1] and

y[n]=[2 2 1 1]

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EXPERIMENT NO 5
Solving a Given Difference Equation

Aim: To obtain the impulse response/step response of a system described by

the given difference equation

Theory:
● A difference equation with constant coefficients describes a LTI system. For
example the difference equation y[n] + 0.8y[n-2] + 0.6y[n-3] = x[n] +
0.7x[n-1] + 0.5x[n-2] describes a LTI system of order 3. The coefficients
0.8, 0.7, etc. are all constant i.e., they are not functions of time (n). The
difference equation y[n]+0.3ny[n-1]=x[n] describes a time varying system
as the coefficient 0.3n is not constant.
● The difference equation can be solved to obtain y[n], the output for a given
input x[n] by rearranging as y[n] = x[n] + 0.7x [n-1]+0.5x[n-2]- 0.8y[n-2]-
0.6y[n-3] and solving.
● The output depends on the input x[n]
o With x[n]= δ[n], an impulse, the computed output y[n] is the impulse
response.
o If x[n]=u[n], a step response is obtained.
o If x[n] = cos(wn) is a sinusoidal sequence, a steady state response is
obtained (wherein y[n] is of the same frequency as x[n], with only an
amplitude gain and phase shift-refer Fig.7.3).
o Similarly for any arbitrary sequence of x[n], the corresponding output
response y[n] is computed.

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● The difference equation containing past samples of

output, i.e., y[n-1], y[n-2], etc. leads to a recursive
system, whose impulse response is of infinite duration
(IIR). For such systems the impulse response is
computed for a large value of n, say n=100 (to
approximate n=∞). The MATLAB function filter is used
to compute the impulse response/ step response/
response to any given x[n]. Note: The filter function
evaluates the convolution of an infinite sequence (IIR)
and x[n], which is not possible with conv function
(remember conv requires both the sequences to be
finite).
● The difference equation having only y[n] and present and past samples of
input (x[n], x[n-k]), represents a system whose impulse response is of finite
duration (FIR). The response of FIR systems can be obtained by both the
‘conv’ and ‘filter’ functions. The filter function results in a response whose
length is equal to that of the input x[n], whereas the output sequence from
conv function is of a longer length (xlength + hlength-1).
Algorithm:
1. Input the two sequences as a and b representing the coefficients of y and x.
2. If IIR response, then input the length of the response required (say 100,
which can be made constant).
3. Compute the output response using the ‘filter’ command.
4. Plot the input sequence & impulse response (and also step response, etc if
required).

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MATLAB Implementation:
MATLAB has an inbuilt function ‘filter’ to solve difference equations
numerically, given the input and difference equation coefficients (b, a).
y=filter (b,a,x)
where x is the input sequence, y is the output sequence which is of same length as
x.
Given a difference equation a0y[n]+a1y[n-1]+a2y[n-2]=b0x[n]+b2x[n-2], the
coefficients are written in a vector as b=[b0 0 b2] and a=[a0 a1 a2]. Note the zero in
b (x[n-1] term is missing).
Also remember a0, the coefficient of y[n] should always be 1.
For impulse response x[n] = {1, 0, 0, 0, ----} the number of zeros = the length of
the IIR response required (say 100 implemented by function zeros(1,99)).
For step response x[n]={1,1,1,1,----} the number of one’s = the length of the IIR
response required-1 (say 100 implemented by function ones(1,100).
Similarly for any given x[n] sequence, the response y[n] can be calculated.

Given Problem
1) Difference equation y(n) - y(n-1) + 0.9y(n-2) = x(n);
Calculate impulse response h(n) and also step response at n=0,…..,100
2) Plot the steady state response y[n] to x[n]=cos(0.05πn)u(n), given y[n]-0.8y[n-
1]=x[n]
MATLAB Program:
%To find Impulse Response
N=input ('Length of response required=');
b=[1]; %x[n] coefficient
a=[1,-1,0.9]; %y coefficients

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%impulse input

h=[1,zeros(1,N-1)];
%time vector for plotting
n=0:1:N-1;
%impulse response
y=filter(b,a,h);

%plot the waveforms

subplot (2,1,1);
stem(n,h);
title('impulse input');
xlabel('n');
ylabel('impulse)');
subplot(2,1,2);
stem(n,y);
title('impulse response');
xlabel('n');
ylabel('y(n)');

Result:
Length=100

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Output Waveform

%To find step response

N=input('Length of response required=');

b=[1]; %x[n] coefficient
a=[1,-1,0.9]; %y coefficients
h=[ones(1,N)]; %step input
n=0:1:N-1; %time vector for plotting
y=filter(b,a,h); %step response

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%plot the waveforms

subplot(2,1,1);
stem(n,h);
title('step input');
xlabel('n');
ylabel('u(n)');
subplot(2,1,2);
stem(n,y);
title('step response');
xlabel('n');
ylabel('y(n)');
Result:
Length=100
Output Waveform

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%To find steady state response

%y[n]-0.8y[n-1]=x[n]
N=input('Length of response required=');
b=[1]; %x[n] coefficient
a=[1,-0.8]; %y coefficients
n=0:1:N-1; %time vector
h=cos(0.05*pi*n); % sinusoidal input
y=filter(b,a,h); %steady state response
subplot(2,1,1);
stem(n,h);
title('steady state input');
xlabel('n');
ylabel('u(n)');
subplot(2,1,2);
stem(n,y);
title('steady state response');
xlabel('n');
ylabel('y(n)');
Result:
Length=100

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Output Waveform

Open ended Experiments :

Given y[n] = y[n-1]+y[n-2]+x[n-1] find
a. Impulse response
b. Step response
c. Sinusoidal response

VIVA Questions
1.Explain the Matlab function filter
2.How to find various responses using filter function

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EXPERIMENT NO 6
Design and Implementation of IIR filter to meet given specifications.

Aim: To design and implement an IIR filter for given specifications.

Theory:
There are two methods of stating the specifications as illustrated in previous
program. In the first program, the given specifications are directly converted to
digital form and the designed filter is also implemented. In the last two programs
the butterworth and chebyshev filters are designed using bilinear transformation
(for theory verification).
Method I: Given the order N, cutoff frequency fc, sampling frequency fs and the
IIR filter type (butterworth, cheby1, cheby2).
● Step 1: Compute the digital cut-off frequency Wc (in the range -π <Wc< π, with
π corresponding to fs/2) for fc and fs in Hz. For example let fc=400Hz,
fs=8000Hz
Wc = 2*π* fc / fs = 2* π * 400/8000 = 0.1* π radians
For MATLAB the Normalized cut-off frequency is in the range 0 and 1, where
1 corresponds to fs/2 (i.e.,fmax)). Hence to use the MATLAB commands
wc = fc / (fs/2) = 400/(8000/2) = 0.1
Note: if the cut off frequency is in radians then the normalized frequency is
computed as wc = Wc / π
● Step 2: Compute the Impulse Response [b,a] coefficients of the required IIR
filter and the response type (lowpass, bandpass, etc) using the appropriate
butter, cheby1, cheby2 command. For example given a butterworth filter,
order N=2, and a high pass response, the coefficients [b,a] of the filter are

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computed using the MATLAB inbuilt command ‘butter’ as [b,a] =butter(N, wc ,

'high');
Method 2:
Given the pass band (Wp in radians) and Stop band edge (Ws in radians)
frequencies, Pass band ripple Rp and stopband attenuation As.
● Step 1: Since the frequencies are in radians divide by π to obtain normalized
frequencies to get wp=Wp/pi and ws=Ws/pi
If the frequencies are in Hz (note: in this case the sampling frequency should
be given), then obtain normalized frequencies as wp=fp/(fs/2),
ws=fstop/(fs/2), where fp, fstop and fs are the passband, stop band and
sampling frequencies in Hz
● Step 2: Compute the order and cut off frequency as
[N, wc] = BUTTORD(wp, ws, Rp, Rs)
● Step 3: Compute the Impulse Response [b,a] coefficients of the required IIR
filter and the response type as [b,a] =butter(N, wc , 'high');
●
IMPLEMENTATION OF THE IIR FILTER
1. Once the coefficients of the IIR filter [b,a] are obtained, the next step is to
simulate an input sequence x[n], say input of 100, 200 & 400 Hz (with
sampling frequency of fs), each of 20/30 points. Choose the frequencies such
that they are >, < and = to fc.
2. Filter the input sequence x[n] with Impulse Response, to obtain the output
of the filter y[n] using the ‘filter’ command.
3. Infer the working of the filter (low pass/ high pass, etc).

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MATLAB IMPLEMENTATION
BUTTORD Butterworth filter order selection.
[N, Wn] = BUTTORD (Wp, Ws, Rp, Rs) returns the order N of the lowest order
digital Butterworth filter that loses no more than Rp dB in the passband and has at
least Rs dB of attenuation in the stopband. Wp and Ws are the passband and
stopband edge frequencies, normalized from 0 to 1 (where 1 corresponds to pi
radians/sample). For example,
Lowpass: Wp = .1, Ws = .2 Highpass: Wp = .2, Ws = .1
Bandpass: Wp = [.2 .7], Ws = [.1 .8] Bandstop: Wp = [.1 .8], Ws = [.2 .7]
BUTTORD also returns Wn, the Butterworth natural frequency (or, the "3 dB
frequency") to use with BUTTER to achieve the specifications.
[N, Wn] = BUTTORD(Wp, Ws, Rp, Rs, 's') does the computation for an analog
filter, in which case Wp and Ws are in radians/second.
When Rp is chosen as 3 dB, the Wn in BUTTER is equal to Wp in BUTTORD.

BUTTER Butterworth digital and analog filter design.

[B,A] = BUTTER(N,Wn) designs an Nth order lowpass digital Butterworth filter
and returns the filter coefficients in length N+1 vectors B (numerator) and A
(denominator). The coefficients are listed in descending powers of z. The cutoff
frequency Wn must be 0.0 <Wn< 1.0, with 1.0 corresponding to half the sample
rate. If Wn is a two-element vector, Wn = [W1 W2], BUTTER returns an order 2N
bandpass filter with passband W1< W < W2. [B,A] = BUTTER(N,Wn,'high')
designs a highpass filter. [B, A] = BUTTER (N,Wn,'stop') is a bandstop filter if Wn
= [W1 W2].BUTTER (N,Wn,'s'), BUTTER (N,Wn,'high','s') and BUTTER
(N,Wn,'stop','s') design analog Butterworth filters. In this case, Wn is in
[rad/s] and it can be greater than 1.0.

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MATLAB PROGRAMS
Butterworth Filters
1.Butterworth LPF
clc;
clear all;
close all;
lp=input('Enter the passband attenuation:');
ls=input('Enter the stopband attenuation:');
wp=input('Enter the passband frequency:');
ws=input('Enter the stopband frequency:');
fs=input('Enter the sampling frequency:');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,lp,ls);disp(n);disp(wn)
[b,a]=butter(n,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure(1);
subplot(2,1,1);
title('Butterworth Low Pass Filter');
plot(om/pi,m);
grid on;
ylabel('Gain in db-->');
xlabel('(a)Normalised frequency-->');

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subplot(2,1,2);
plot(om/pi,an);
xlabel('(b)Normalised frequency-->');
ylabel('Phase in radians-->');
grid on;
Result:
Enter the passband attenuation: 0.5
Enter the stopband attenuation: 50
Enter the passband frequency: 1200
Enter the stopband frequency: 2400
Enter the sampling frequency: 10000
Output Wave form

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2 Butterworth HPF
clc;
clear all;
close all;
format long
lp=input('Enter the passband attenuation:');
ls=input('Enter the stopband attenuation:');
wp=input('Enter the passband frequency:');
ws=input('Enter the stopband frequency:');
fs=input('Enter the sampling frequency:');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,lp,ls); disp(n);disp(wn)
[b,a]=butter(n,wn,'high');
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure(1);
subplot(2,1,1);
title('Butterworth Low Pass Filter');
plot(om/pi,m);
grid on;
ylabel('Gain in db-->');
xlabel('(a)Normalised frequency-->');
subplot(2,1,2);

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plot(om/pi,an);
xlabel('(b)Normalised frequency-->');
ylabel('Phase in radians-->');
grid on
Result:
Enter the passband attenuation: 0.5
Enter the stopband attenuation: 50
Enter the passband frequency: 1200
Enter the stopband frequency: 2400
Enter the sampling frequency: 10000

Output Wave form

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CHEBYSHEV FILTERS
1 Chebyshev LPF
clc;
close all;
clear all;
format long
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband frequency');
ws=input('enter the stopband frequency');
fs=input('enter the sampling frequency');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);plot(om/pi,m);
ylabel('gain in db-->');
xlabel('(a) normalised frequency-->');
subplot(2,1,2);plot(om/pi,an);
xlabel('(b) normalised frequency-->');
ylabel('phase in radians-->');

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Result:
enter the passband ripple 0.2
enter the stopband ripple 45
enter the passband frequency 1300
enter the stopband frequency 1500
enter the sampling frequency 10000

Output Waveform

2 Chebyshev HPF
clc;
close all;
clear all;
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband frequency');
ws=input('enter the stopband frequency');

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

w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn,'high');
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));

an=angle(h);
subplot(2,1,1);plot(om/pi,m);
ylabel('gain in db-->');
xlabel('(a) normalised frequency-->');
subplot(2,1,2);plot(om/pi,an);
xlabel('(b) normalised frequency-->');
ylabel('phase in radians-->');

Result:
enter the passband ripple 0.3
enter the stopband ripple 60
enter the passband frequency 1500
enter the stopband frequency 2000
enter the sampling frequency 9000

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Output Waveform

VIVA Questions:
1. Define an IIR filter.
2. Compare Impulse invariance and bilinear transformations.
3. How do you convert analog filter prototype to a digital filter?
4. What are the functions used in MATLAB for designing a digital
Butterworth and Chebyshev low pass filter using BLT?
5. Draw typical responses of Chebyshev filter for order odd & even.
6. Compare FIR & IIR filters.

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EXPERIMENT NO 7
Design and Implementation of FIR Filter to Meet Given Specifications

Aim: To design and implement a FIR filter for given specifications.

Theory:
There are two types of systems – Digital filters (perform signal filtering in time
domain) and spectrum analyzers (provide signal representation in the frequency
domain). The design of a digital filter is carried out in 3 steps- specifications,
approximations and implementation.

DESIGNING AN FIR FILTER (using window method):

Method 1: Given the order N, cutoff frequency fc, sampling frequency fs and the
window.
● Step 1: Compute the digital cut-off frequency Wc (in the range -π <Wc< π,
with π corresponding to fs/2) for fc and fs in Hz. For example let fc=400Hz,
fs=8000Hz
Wc = 2*π* fc / fs = 2* π * 400/8000 = 0.1* π radians
For MATLAB the Normalized cut-off frequency is in the range 0 and 1,
where 1 corresponds to fs/2 (i.e.,fmax)). Hence to use the MATLAB
commands
wc = fc / (fs/2) = 400/(8000/2) = 0.1
Note: if the cut off frequency is in radians then the normalized frequency is
computed as wc = Wc / π

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● Step 2: Compute the Impulse Response h(n) of the required FIR filter using
the given Window type and the response type (low pass, band pass, etc). For
example given a rectangular window, order N=20, and a high pass response,
the coefficients (i.e., h[n] samples) of the filter are computed using the
MATLAB inbuilt command ‘fir1’ as
h =fir1(N, wc , 'high', boxcar(N+1));
Note: In theory we would have calculated h[n]=hd[n]×w[n], where hd[n] is
the desired impulse response (low pass/ high pass, etc given by the sinc
function) and w[n] is the window coefficients. We can also plot the window
shape as stem(boxcar(N)).
Plot the frequency response of the designed filter h(n) using the freqz
function and observe the type of response (lowpass / highpass /bandpass).

Method 2:
Given the pass band (wp in radians) and Stop band edge (ws in radians)
frequencies, Pass band ripple Rp and stopband attenuation As.
● Step 1: Select the window depending on the stopband attenuation required.
Generally if As>40 dB, choose Hamming window. (Refer table )
● Step 2: Compute the order N based on the edge frequencies as
Transition bandwidth = tb=ws-wp;
N=ceil (6.6*pi/tb);
● Step 3: Compute the digital cut-off frequency Wc as
Wc=(wp+ws)/2
Now compute the normalized frequency in the range 0 to 1 for MATLAB
as
wc=Wc/pi;

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Note: In step 2 if frequencies are in Hz, then obtain radian frequencies (for
computation of tb and N) aswp=2*pi*fp/fs, ws=2*pi*fstop/fs, where fp,
fstop and fs are the pass band, stop band and sampling frequencies in Hz
● Step 4: Compute the Impulse Response h(n) of the required FIR filter using
N, selected window, type of response(low/high,etc) using ‘fir1’ as in step 2
of method 1.

IMPLEMENTATION OF THE FIR FILTER

1. Once the coefficients of the FIR filter h[n] are obtained, the next step is to
simulate an input sequence x[n], say input of 100, 200 & 400 Hz (with
sampling frequency of fs), each of 20/30 points. Choose the frequencies such
that they are >, < and = to fc.
2. Convolve input sequence x[n] with Impulse Response, i.e., x (n)*h (n) to
obtain the output of the filter y[n]. We can also use the ‘filter’ command.
3. Infer the working of the filter (low pass/ high pass, etc).

MATLAB IMPLEMENTATION
FIR1 Function
B = FIR1(N,Wn) designs an N'th order low pass FIR digital filter and returns the
filter coefficients in length N+1 vector B. The cut-off frequency Wn must be
between 0 <Wn< 1.0, with 1.0 corresponding to half the sample rate. The filter B
is real and has linear phase, i.e., even symmetric coefficients obeying B(k) =
B(N+2-k), k = 1,2,...,N+1.
If Wn is a two-element vector, Wn = [W1 W2], FIR1 returns an order N band pass
filter with pass band W1< W < W2. B = FIR1(N,Wn,'high') designs a high pass
filter. B = FIR1(N,Wn,'stop') is a band stop filter if Wn = [W1 W2]. If Wn is a

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multi-element vector, Wn = [W1 W2 W3 W4 W5 ... WN], FIR1 returns an order

N multiband filter with bands
0 < W < W1, W1 < W < W2, ..., WN < W < 1.
FREQZ Digital filter frequency response. [H,W] = FREQZ(B,A,N) returns the N-
point complex frequency response vector H and the N-point frequency vector W in
radians/sample of the filter whose numerator and denominator coefficients are in
vectors B and A. The frequency response is evaluated at N points equally spaced
around the upper half of the unit circle. If N isn't specified, it defaults to 512.
For FIR filter enter A=1 and B = h[n] coefficients. Appropriately choose N as 128,
256, etc

Matlab Program:
1 RECTANGULAR WINDOW
clc;
clear all;
close all;
lp=input('Enter the pass band ripple:');
ls=input('Enter the stop band ripple:');
fp=input('Enter the pass band frequency:');
fs=input('Enter the stop band frequency:');
f=input('Enter the sampling frequency:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);

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n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;
end
y=boxcar(n1);% or y=rect(n1);
% Low Pass Filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;
% High Pass Filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

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Result
Enter the pass band ripple: 0.05
Enter the stop band ripple: 0.04
Enter the pass band frequency: 1500
Enter the stop band frequency: 2000
Enter the sampling frequency: 9000

Output waveform

2 HAMMING WINDOW
clc;
clear all;
close all;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');

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wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;
end
y=hamming(n1);

% Low Pass Filter

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;
% High Pass Filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);

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plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

Result:
Enter the passband ripple: 0.05
Enter the stopband ripple: 0.001
Enter the passband frequency: 1200
Enter the stopband frequency: 1700
Enter the sampling frequency: 9000
Output waveform

3 BARTLET
clc;
clear all;
close all;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');
fp=input('Enter the passband frequency:');

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fs=input('Enter the stopband frequency:');

f=input('Enter the sampling frequency:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;
end
y=bartlett(n1);

% Low Pass Filter

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;

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% High Pass Filter

b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);

m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

Result:
Enter the passband ripple: 0.05
Enter the stopband ripple: 0.04
Enter the passband frequency: 1500
Enter the stopband frequency: 2000
Enter the sampling frequency: 9000
Output Waveform

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4 HANNING
clc;
clear all;
close all;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;
end
y=hanning(n1);
% Low Pass Filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);

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ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;

% High Pass Filter

b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

Result
Enter the passband ripple: 0.03
Enter the stopband ripple: 0.01
Enter the passband frequency: 1400
Enter the stopband frequency: 2000
Enter the sampling frequency: 8000

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Output Waveform

6 KAISER
clc;
clearall;
closeall;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');
beta=input('Enter the beta value:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;

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end
y=kaiser(n1,beta);

% Low Pass Filter

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
gridon;

% High Pass Filter

b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
gridon;

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Result:
Enter the passband ripple: 0.05
Enter the stopband ripple: 0.04
Enter the passband frequency: 1500
Enter the stopband frequency: 2000
Enter the sampling frequency: 9000
Enter the beta value: 5.8
Output Waveform

VIVA Questions:
1. What is a Filter? What are its specifications?
2. Mention the types of filters.
3. Compare analog and digital filters.
4. Define a FIR filter.
5. What are the important properties of FIR filter?
6. What are the window techniques used?
7. Why the Kaiser window is important?
8. How do you calculate the length of Kaiser Window?
9. What are the functions used in designing a FIR filter in Matlab

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PART II

EXPERIMENTS USING DSP PROCESSOR

CCS INTRODUCTION

Procedure
1. Double click on CCS icon
2. Create a new project: project 🡪 New
Type the project name (x.pjt) or (USN) & store in my projects folder
To open an existing project: project 🡪 open
3. Files to be added to the project: project 🡪 Add Files to project
a. c: \ CCStudio \ c6000 \ cgtools \ lib \ rts6700.lib
<Select type of file as: library>
b. c: \ CCStudio \ tutorial \ dsk6713 \ hello1 \ hello.cmd
< Select type of file as: Linker Command file>
4. File 🡪 New 🡪 source file 🡪 same as xx.c and compile
( Select type of file as: C/C++ file before saving) & add this C file to your
project
5. Project 🡪 Build. (obtain Build successful)
6. To execute ; File 🡪 load program (select pjtname.out file)
7. To Run : Debug 🡪 Run
8. Observe the output on the stdout window or waveforms using graph or CRO
9. To view waveforms View 🡪 Graph
changes in graph property dialog box to be made
a. Display type 🡪 Dual (to observe 2 waveforms)

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b. Title 🡪 User defined

c. Start address – upper display 🡪 x (user defined variable array names used in
C program. Note: x, y should be made global to be used by graph)
d. Start address – lower display 🡪 y
e. Acquisition Buffer size 🡪 depends on the array size
f. Display Buffer size 🡪 verify given output data size and change
g. DSP data type – 32-bit Floating pt
h. Auto scale 🡪 ON (if off 🡪 choose max yvalue=1)

EXPERIMENT NO 8
a. Linear convolution for the given sequences

Aim: To perform linear convolution for the given sequences

+∞ +∞
y [ n ]= x [ n ]∗ h [n ]= ∑ x [ k ]h[ n− k ]= ∑ x [ n− k ]h [k ]
The linear convolution sum is k= − ∞ k= − ∞

C Program

# include<stdio.h>
# include<math.h>
float y[10];
main()
{
//the two sequences can be of different lengths, no need to zero pad for this
program
float h[4] = { 2,2,2,2};
float x[4] ={1,2,3,4};
intxlen=4;
inthlen=4;

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int N=xlen+hlen-1;
intk,n;
for(n=0;n<N;n++) //outer loop for y[n] array
{ y[n]=0;
for(k=0;k<hlen;k++) //inner loop for computing each y[n] point
{ if (((n-k)>=0) & ((n-k)<xlen))
y[n]=y[n]+h[k]*x[n-k]; //compute output
} //end of inner for loop

printf("%f\t", y[n]);
} //end of outer for loop
} //end of main

Exercise: Find the linear convolution of x[n]=[2,1,2,1] and h[n]=[1,2,3,4]

b. Circular convolution of two given sequences

Aim: To perform circular convolution of two given sequences

+∞
y [ n ]= x [ n ](N )h [n ]= ∑ h[ k ]x [〈n− k 〉N ]
The circular convolution sum is k= − ∞

C Program
# include<stdio.h>
# include<math.h>
float y[10]; //output sequence
main()

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{ //input the two sequences, if of uneven lengths zero pad the smaller one such
that both are of same size
float x[5]={1,2,3,4,5};
float h[5]={2,1,3,4,5};
int N=5; // N=max of xlen and hlen//
intk,n,i;
for(n=0;n<N;n++) //outer loop for y[n] array
{ y[n]=0;
for(k=0;k<N;k++) //inner loop for computing each y[n] point
{ i=(n-k)%N; //compute the index modulo N
if(i<0) //if index is <0, say x[-1], then convert to x[N-1]
i=i+N;
y[n]=y[n]+h[k]*x[i]; //compute output
} //end of inner for loop
printf("%f\t",y[n]);
} //end of outer for loop
} //end of main

Open ended Experiments :

Find the circular convolution of x[n]=[2 3 1 1] and h[n]=[1 3 5 3]

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EXPERIMENT NO 9
Computation of N- Point DFT of a given sequence

Aim: To compute the N (=4/8/16) point DFT of the given sequence

Theory:
The N point DFT of discrete time signal x[n] is given by the equation
N-1 − j2π kn
X ( k )= ∑ x [n ]e N
; k= 0,1,2, .. .. N-1
n= 0
Where N is chosen such that N≥ L , where L=length of x[n]. To implement using
− j2π kn

C program we use the expression

e N
= cos (2πNkn )− j sin (2πNkn )and allot memory
space for real and imaginary parts of the DFT X(k)

C Program
#include <stdio.h>
#include <math.h>
main()
{float y[16]; //for 8 point DFT to store real & imaginary
float x[4]={1,3,2,5}; //input only real sequence
float w;
int n,k,k1,N=8,xlen=4;
for(k=0;k<2*N;k=k+2)
{y[k]=0; y[k+1]=0; //initialize real &image parts
k1=k/2; //actual k index
for(n=0;n<xlen;n++)
{w=-2*3.14*k1*n/N; //careful about minus sign
y[k]=y[k]+x[n]*cos(w);

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y[k+1]=y[k+1]+x[n]*sin(w);
}
printf("%f+j%f \n",y[k],y[k+1]);
}
}//end of main

MATLAB verification
>> x=[1 3 2 5];
>>fft(x,8) 11.0000 -0.4142 - 7.6569i -1.0000 + 2.0000i 2.4142 -
3.6569i -5.0000 2.4142 + 3.6569i -1.0000 - 2.0000i -0.4142 + 7.6569i

Open ended Experiments: Find the 8 point DFT of x[n]=[0.5,0,0.5,0]

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EXPERIMENT NO 10
Impulse response of the given system

Aim: To find the Impulse response of the given first order / second order
system
Theory:
A linear constant coefficient difference equation representing a second order
system is given by y[ n ]+ a 1 y[n− 1]+ a2 y[ n− 2 ]= b0 x[ n]+ b1 x[n− 1]+ b2 x[n− 2];
For a first order system the coefficients a2 and b2 are zero.
Since the difference equation contains past samples of output, i.e., y[n-1], y[n-
2], its impulse response is of infinite duration (IIR). For such systems the
impulse response is computed for a large value of N, say N=100 (to
approximate n=∞).
Impulse response implies x[n]= δ[n], an impulse, and the initial conditions x[-1],
x[-2], y[-1] and y[-2] are zero.
(referexpt 2 &7of part A - h=impz(b,a,N); %impulse response verification in
MATLAB)
To implement in C the following algorithm is used.

Algorithm
1. Input the coefficients b0 b1 b2, a1 a2 (Say the coefficients from a 2nd order
Butterworth filter. Note coefficient of y[n] is a0=1 always)
2. Generate the impulse input of size n (1st element=1, rest all=0)
3. Compute the output as
y[ n ]= − a1 y[ n− 1]− a 2 y[n− 2]+ b 0 x[n]+ b1 x[ n− 1 ]+ b2 x[ n− 2 ];

4. Repeat step 3 for n=0 through N (say 50,100, etc)

5.

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C Program
#include <stdio.h>
float x[60],y[60];
main()
{float a1,a2,b0,b1,b2;
inti,j,N=20;
a1= -1.1430; a2= 0.4128;
b0=0.0675;b1=0.1349;b2=0.0675;
//generate impulse input
x[0]=1;
for(i=1;i<N;i++)
x[i]=0;
//generate output
for(j=0;j<N;j++)
{y[j]=b0*x[j];
if(j>0)
y[j]=y[j]+b1*x[j-1]-a1*y[j-1];
if ((j-1)>0)
y[j]=y[j]+b2*x[j-2]-a2*y[j-2];
printf("%f \t",y[j]);
}
}//end of main
Note: For a first order system just enter a2=0 and b2=0.
Open ended Experiments:
Find the impulse response of the given difference equation
c. y[n]-1/9y[n-2]=x[n-1]
d. y[n]-3/4y[n-1]+1/8y[n-2]=x[n]+1/2x[n-1]

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EXPERIMENT NO 11
Noise removal programs:
Aim: To Add noise above 3kHz and then remove using adaptive filters

Theory:

In the previous experiments involving IIR & FIR filters, the filter coefficients
were determined before the start of the experiment and they remained constant.
Whereas Adaptive filters are filters whose transfer function coefficients or taps
change over time in response to an external error signal. Typical applications of
these filters are Equalization, Linear Prediction, Noise Cancellation, Time-Delay
Estimation, Non-stationary Channel Estimation, etc. Different types of adaptive
filter algorithms are the Kalman adaptive filter algorithm, LMS adaptive filter
algorithm and RLS adaptive filter algorithm

Fig.7.1 Adaptive Structure for noise cancellation

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Fig.7.2 Adaptive Filter (In program the coefficients (weights) ‘a’ replaced by
w[T], X0-> delay [0]

In the below program, two signals - a desired sinusoidal signal (can be the output
of the PC speaker/ signal generator of square/ sine wave of frequency not
correlated to 3kHz and not above the fsample/2 of the codec) into the Left
channel and
Noise signal of 3KHz into the Right channel of the Line In are given (generally
using two signal generators with common ground is sufficient).

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C Program

#include "NCcfg.h"
#include "dsk6713.h"
#include "dsk6713_aic23.h"
#define beta 1E-13 //rate of convergence
#define N 30 //adaptive FIR filter length-vary this parameter &
observe

float delay[N];
float w[N];

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 */ \
0x0011, /* 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 */
\

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0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \

0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */
\
0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \
};

/* main() - Main code routine, initializes BSL and generates tone*/

void main()
{ DSK6713_AIC23_CodecHandle hCodec;
int l_input, r_input,l_output, r_output,T;
/* Initialize the board support library, must be called first */
DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0, &config); /* Start the codec */
DSK6713_AIC23_setFreq(hCodec, 1);

for (T = 0; T < 30; T++) //Initialize the adaptive FIR coeffs=0

{ w[T] = 0; //init buffer for weights
delay[T] = 0; //init buffer for delay samples
}

while(1)
{ /* Read a sample to the left channel */
while (!DSK6713_AIC23_read(hCodec,&l_input));
/* Read a sample to the right channel */
while (!DSK6713_AIC23_read(hCodec, &r_input));

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l_output=(short int)adaptive_filter(l_input,r_input);
r_output=l_output;
while (!DSK6713_AIC23_write(hCodec, l_output)); /* Send o/p to the left
channel */
while (!DSK6713_AIC23_write(hCodec, r_output)); /* Send o/p to the right
channel*/
}
DSK6713_AIC23_closeCodec(hCodec); /* Close the codec */
}

signed int adaptive_filter(int l_input,int r_input) //ISR

{
short i,output;
float yn=0, E=0, dplusn=0,desired,noise;

desired = l_input;
noise = r_input;
dplusn = (short)(desired + noise); //desired+noise
delay[0] = noise; //noise as input to adapt FIR

for (i = 0; i < N; i++) //to calculate out of adapt FIR

yn += (w[i] * delay[i]); //output of adaptive filter
E = (desired + noise) - yn; //"error" signal=(d+n)-yn
for (i = N-1; i >= 0; i--) //to update weights and delays
{

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w[i] = w[i] + beta*E*delay[i]; //update weights

delay[i] = delay[i-1]; //update delay samples
}

//output=((short)E); //error signal as overall output

output=((short)dplusn);//output (desired+noise)
//overall output result
return(output);
}

Result:
Observe the waveform that appears on the CRO screen. Verify that the 3 KHz
noise signal is being cancelled gradually.

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Open Ended Experiment

In this open end experiments students are expected to write programs to create
wave forms such as square wave, triangular wave and saw tooth wave. Students
may write separate programs for each waveform.

Mini Project
Students are expected to create a GUI based signal generator using MATLAB. The
signal generator will have three buttons to select three signals ie., sine wave, square
wave and triangular wave. Upon the selection the wave form in time domain and
frequency domain needs to be displayed graphically. There should be a control to
change the amplitude and frequency of the signals.

Questions:
1. Why do we need SP processors?
2. What is the difference between Fixed and floating point processors?
3. Explain the architectural features of DSP processor.
4. How do perform signed arithmetic on DSP processor?
5. What are the advantages and disadvantages of DSP processor?

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