Updated - DSP Lab Manual

DSP Lab Manual TCL 56
DEPARTMENT OF TELECOMMUNICATION
ENGINEERING
DIGITAL SIGNAL PROCESSING LAB

LAB MANUAL
Vth Sem
TCL 56
In-charge Faculty
Mr. VENU K N
Mrs. Akshatha Kori
Dept. of Telecommunication Engineering MSRIT

Prerequisite Courses with codes: Signals & Systems (TC45)
Course Objectives
1. Understand the basic concepts of discrete-time signal processing like
sampling theorem.
2. Identify the concepts behind convolution, DFT & IDFT, auto correlation and
cross correlation.
3. Developing the solution for a difference equation and implement the same
by writing the program.
4. Identify the concepts behind FIR filters and design using different windows
like rectangular, hamming, hanning and Kaiser and implement the same
using matlab
5. Identify the concepts behind IIR filters and design the same using different
windows like rectangular, hamming, hanning and Kaiser and implement the
same using matlab
Course Outcomes
1. Analyze DSP concepts like Sampling theorem, convolution, DFT & IDFT,
correlation and solution for a difference equation on Mat-lab.
2. Computation of convolution and correlation.
3. Design of FIR filter by using different window methods.
4. Design IIR filter by using different window methods.
5. Design of FIR and IIR filters and implementation through DSP Processor.
Course
List of Experiments
Outcomes
Verification of sampling theorem. CO1

1
Impulse response of a given system CO1

2
Linear convolution of two given sequences. CO1

3
Circular convolution of two given sequences CO2

4

Autocorrelation of a given sequence and

CO2
5
verification of its properties.
Cross correlation of given sequences and
CO2
6
verification of its properties.
Linear convolution of two sequences using DFT
and IDFT.
CO3
7
Circular convolution of two given sequences using
DFT and IDFT
Solving a given difference equation.
Computation of N point DFT of a given sequence CO3
8
and to plot magnitude and phase spectrum.
Design and implementation of FIR filter to meet

CO4
9 given specifications
Design and implementation of IIR filter to meet

CO4
10
given specifications.
Linear convolution of two given sequences using
radix-2 algorithm.
CO5
11
radix-2 algorithm
Computation of N- Point DFT of a given sequence.

Realization of an FIR filter (any type) to meet
CO5
12
given specifications. The input can be a signal
from function generator / speech signal
Computation of DFT and FFT of a given sequence CO5

13
Realization of an FIR and IIR filter to meet given

CO5
14
specifications.

COURSE DESIGN, DELIVERY AND ASSESMENT
Course Code and Title : Digital Signal Processing Lab, Course Credits : 0:0:1
TCL506
CIE : 50 Marks SEE : 50 Marks
Total No of Lab Hours : 24, No. of sessions: 12
Prepared by : Umesharaddy Date :31/08/2020
Reviewd by : Dr. B K Sujatha Date :31/08/2020
Prerequisites
Prerequisite Courses with codes : Signals & Systems(TC405)
Course Objectives
1. Understand the basic concepts of discrete-time signal processing like
sampling theorem.
2. Identify the concepts behind convolution, DFT & IDFT, auto correlation and
cross correlation.
3. Developing the solution for a difference equation and implement the same
by writing the program.
4. Identify the concepts behind FIR filters and design using different windows
like rectangular, hamming, hanning and Kaiser and implement the same
using matlab
5. Identify the concepts behind IIR filters and design the same using different
windows like rectangular, hamming, hanning and Kaiser and implement the
same using matlab
List of experiments:
Duratio
Sl No.
Topics n
1 Verification of sampling theorem. 2 hr
2 Impulse response of a given system 2 hr
3 Linear convolution of two given sequences. 2 hr
4 Circular convolution of two given sequences 2 hr
Autocorrelation of a given sequence and verification

5 2 hr
of its properties.

Cross correlation of given sequences and verification

6 2 hr
of its properties.
Linear convolution of two sequences using DFT and
IDFT.
7 2 hr
DFT and IDFT
Solving a given difference equation.
8 Computation of N point DFT of a given sequence and 2 hr
to plot magnitude and phase spectrum.

Design and implementation of FIR filter to meet given
9 2 hr
Specifications
Design and implementation of IIR filter to meet given

10 2hr
specifications.
Linear convolution of two given sequences using

radixz2 algorithm.
11 2hr
radix-2 algorithm
Computation of N- Point DFT of a given sequence

Realization of an FIR filter (any type) to meet given
12 2hr
specifications. The input can be a signal from
function generator / speech signal
13 Computation of DFT and FFT of a given sequence 2hr
Realization of an FIR and IIR filter to meet given

14 2hr
specifications.
REFERENCE BOOKS:
1. Oppenheim & Schaffer, Discrete Time Signal Processing, PHI, 2003.

2. Proakis & Monalakis, Digital signal processing - Principles Algorithms &
Applications, Pearson education, 4th Edition, New Delhi, 2007.
3. S. K. Mitra, Digital Signal Processing, Tata Mc-Graw Hill, 2nd Edition, 2004.

4. Dr.D. Ganesh Rao, Vineeta P Gejji, Digital Signal Processing-, 2E,

Sanguine Technical Publications.
Course Outcomes
1. Analyze DSP concepts like Sampling theorem, convolution, DFT & IDFT,
correlation and solution for a difference equation on Mat-lab.
2. Computation of convolution and correlation.
3. Design of FIR filter by using different window methods.
4. Design IIR filter by using different window methods.
5. Design of FIR and IIR filters and implementation through DSP Processor.
SUBJECT WILL MAP TO PEO1, PEO2 AND PEO3.
Course Delivery:
The Course will be delivered through black board teaching, exercises, self-study and
demonstrations.
Course Assessment and Evaluation:
What To When/ Where Max Evidence Contributing

whom (Frequency in marks collected to Course
the course) Outcomes
Twice
Internal
(Average of two 20 Blue books C01-C05
assessment
tests)
Every lab
Direct Assessment Methods
session Lab Records,

Record &
CIE (Average of the 20 Observation C01-C05
Observation
all-experiment Books
marks)
Students Twice
(Summation of
Viva viva conducted 10 Bluebooks C01-C05
during two
tests)
End of course
Semester
(Answering 1
SEE End 50 Answer scripts C01-C05
Lab
examination
Experiment)
CO1 – CO5
Indire
sment
Asses
Meth
End of the Feedback

ods
Student’s feedback
ct
Students -
course forms

Feedback
End of course - C01-C05
Course survey forms
Mapping Course Outcomes with Program Outcomes:
Course Programme Outcomes (PO) Program Specific

Outcomes Outcomes (PSO)
1 2 3 4 5 6 7 8 9 10 11 12 1 2 3
1. 3 3 - 1 2 - - - 2 - - 2 1 1
2. 3 3 - 1 2 - - - 2 - - 2 1 1
3. 3 2 - 2 3 - - - 2 - - 2 1 1
4. 3 3 - 1 2 - - - 2 1 1 2 2 1 2
5. 3 3 - 1 2 - - - 1 - 1 2 2 1 1
Overall 3 3 - 2 3 - - - 2 1 1 1 2 1 2
Justification for CO-PO mapping:
1) CO1 maps to PO1,2,4,5, 9
a. PO1: DFT - IDFT and their properties are derived requiring

mathematics and engineering concepts.
b. PO2: DFT & FFT(both DIT-FFT & DIF-FFT) include complex problems
that need analysis. So, the ability to formulate and analyze such
problems is dealt in the course.
c. PO5: Modern tool such as matlab simulation tool was used.
d. PO9: CO1 gives an ability to work effectively as an individual and

members or leader in diverse and multi-disciplinary teams

2) CO2 maps to PO1,2,4 ,5, 9

a. PO1: The knowledge of Symmetry & Anti Symmetry conditions for
linear phase of FIR filters require a good engineering background as
well as a foundation in mathematics.
b. PO2: Analysis of linear phase in FIR filters allow a student to simplify
a complex engineering problems.
c. PO3: With the design of FIR & IIR filters, the specified needs with
societal considerations are met.
d. PO4: The ability to analyze, design complex engineering problems are
dealt here.
e. PO5: Modern tool such as matlab simulation tool was used.
f. PO9: CO2 gives an ability to work effectively as an individual and
3) CO3 maps to PO1,2,4,5, 9
a. PO1: The realization of FIR & IIR filters require a good engineering
and science (modeling) background as well as a foundation in
mathematics.
b. PO2: Use of partial differential equations to solve IIR filter
expressions allow a deeper understanding of complex problems.
c. PO3: With the structures of FIR & IIR filters, the specified needs with
societal, environmental considerations are met.
d. PO4: In realizing both FIR & IIR filters, students will be able to
analyze complex engineering problems.
e. PO5: Modern tool such as matlab simulation tool was used.
f. PO9: CO3 gives an ability to work effectively as an individual and
4) CO4 maps to PO1,2,4,5, 9,10,11,12

a. PO1: Data representation & arithmetic in Harvard architecture for
signal processing require a good engineering background as well as
a foundation in mathematics.
b. PO2: Pipelining-Harvard and data representations allow a deeper
understanding of complex engineering problems.

c. PO4: Various processing like Extended parallelism and static super

scalar processing gives the ability for a student to analyze & design
complex engineering problems.
d. PO5: Modern tool such as matlab simulation tool was used.
e. PO9: CO4 gives an ability to work effectively as an individual and
f. PO10: The concepts of implementing applications on DSP processor
and communicating with peers requires effective reporting and
design documents
g. PO11: Knowledge of fixed point & floating processors & their
structure is a basic requirement for any career in communication
and electrical sciences
h. PO12: DSP projects are implemented on DSP processor hence proper
project planning and managing skills are required
5) CO5 maps to PO1,2,,4,5, 9,10,11,12

a. PO1: The architecture of 67x require a good engineering background
as well as a foundation in mathematics
b. PO2: The various addressing modes allow a student to simplify
complex engineering problems.
c. PO4: In implementing both FIR & IIR filters, students will be able to
analyze complex engineering problems.
d. PO5: Modern tool such as matlab simulation tool was used.
e. PO9: CO4 gives an ability to work effectively as an individual and
f. PO10: The concepts of implementing applications on DSP processor
and communicating with peers requires effective reporting and
design documents
g. PO11: Knowledge of fixed point & floating processors & their
structure is a basic requirement for any career in communication
and electrical sciences
h. PO12: DSP projects are implemented on DSP processor hence proper
project planning and managing skills are required

AIM: PROGRAM FOR GENERATING VARIOUS FORMS OF SIGNALS
PROGRAM:
%UNIT IMPULSE SIGNAL
y1=[1;zeros(999,1)];
subplot(4,3,1)
plot(y1,'r');
%axis([-1 500 -1.5 1.5 ]);------------->not required
xlabel('time(sec)')
ylabel('amplitude')
title('unit impulse signal');
%UNIT STEP SIGNAL

y2=ones(100,1);
subplot (4,3,2);
plot(y2,'b');
xlabel('time(sec)')
ylabel('amplitude')
title('unit step signal');
%RAMP SIGNAL
t=(0:0.001:1);
y3=t;
subplot(4,3,3)
plot(y3,'b');
xlabel('time(sec)')
ylabel('amplitude')
title('ramp signal');
%SINE WAVE SIGNAL

y3=-pi:0.001:pi;
subplot(4,3,4:6);
plot(y3,sin(y3));
xlabel('time(sec)')
ylabel('amplitude')
title('sine wave signal');

%EXPONENTIAL SIGNAL
t=(0:0.001:1);
y5=t.^2;
subplot(4,3,7:9)
plot(y5,'r');
xlabel('time(sec)')
ylabel('amplitude')
title('exponential signal');
%SQUARE WAVE SIGNAL

t=(0:0.001:1);
y6=square(4*t,50);
subplot(4,3,10:12)
plot(y6,'r');
xlabel('time(sec)')
ylabel('amplitude')
title('square signal');

OUTPUT:
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO PROVE SAMPLING THEOREM FOR A

SINUSOIDAL INPUTS
THEORITICAL CALCULATION:
1. COSINE
Input Signal = cos (2π 150t)
Frequency = 150 Hz
INPUT WAVEFORM
EXPECTED WAVEFORM
2. SINE
Input Signal = sin (2π 150t)
Frequency = 150 Hz
INPUT WAVEFORM

PROGRAM:
1. COSINE WAVEFORM
%UNDER SAMPLED SIGNAL
tfinal=0.05;
t=0:0.00005:0.05;
fa=input('enter analog frequency(in Hz):');
%Generate analog signal for the given frequency
xt=cos(2*pi*fa*t);
%condition considered for under sampling is fs is 1.3 times fmax
instead of 2 times of fmax
fs1=1.3*fa;
n1=0:1/fs1:tfinal;
%generate undersampled signal
xn1=cos(2*pi*fa*n1);
%desired frequency is fs is 1.3 times fmax
subplot(3,1,1);
plot(t,xt,'b',n1,xn1,'r*-');
title ('undersampled signal');
%OVER SAMPLED SIGNAL

%condition considered for over sampling that fs is 15 times fmax
fs1=15*fa;
n2=0:1/fs1:tfinal;
%generate oversampled signal
%desired frequency is fs is 5 times fmax
subplot(3,1,2);
title ('oversampled signal');
%NYQUIST RATE SAMPLED SIGNAL

%condition for nyquist rate sampling is that fs is 2 times of fmax
fs1=2*fa;
n3=0:1/fs1:tfinal;
subplot(3,1,3);
title ('nyquist rate sampled signal');

2. SINE WAVEFORM
%UNDER SAMPLED SIGNAL
tfinal=0.05;
t=0:0.00005:0.05;
fa=input('enter analog frequency(in Hz):');
%Generate analog signal for the given frequency
xt=sin(2*pi*fa*t);
%condition considered for under sampling is fs is 1.3 times fmax
fs1=1.3*fa;
n1=0:1/fs1:tfinal;
%generate undersampled signal
xn1=sin(2*pi*fa*n1);
%desired frequency is fs is 1.3 times fmax
subplot(3,1,1);
title ('undersampled signal');
%OVER SAMPLED SIGNAL

%condition considered for over sampling that fs is 15 times fmax
fs1=15*fa;
n2=0:1/fs1:tfinal;
subplot(3,1,2);
title ('oversampled signal');
%NYQUIST RATE SAMPLED SIGNAL

%condition for nyquist rate sampling is that fs is 2 times of fmax
fs1=2*fa;
n3=0:1/fs1:tfinal;
subplot(3,1,3);
title ('nyquist rate sampled signal');

OUTPUT:
COSINE WAVE
WORKSPACE
COMMAND WINDOW

GRAPH

SINE WAVE
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: LINEAR CONVOLUTION OF TWO SEQUENCES USING

PROPERTIES OF DFT AND IDFT
Let x1(n)={1,1,2} ={1,1,2,0} &
x2(n)={1,2}={1,2,0,0}
𝑁−1
DFT of x(n) = X(k) = ∑𝑛=0 𝑥(𝑛)𝑊𝑁𝑘𝑛 ;0≤ 𝑘 ≤N-1
X(k) = 𝑥 (0)𝑊4𝑘0 + 𝑥 (1)𝑊4𝑘1 + 𝑥(2)𝑊4𝑘2 + 𝑥 (3)𝑊4𝑘3 ;0≤ 𝑘 ≤3
For sequence one,
X1(k)= 1 + 1𝑊4𝑘1 + 2𝑊4𝑘2 + 0𝑊4𝑘3 ; 0≤ 𝑘 ≤3
For sequence two,
X2(k)= 1 + 2𝑊4𝑘1 + 0𝑊4𝑘2 + 0𝑊4𝑘3 ; 0≤ 𝑘 ≤3
TWIDDLE FACTOR
2𝜋
𝑊4𝑘𝑛 = ⅇ −𝑗 𝑁
𝑊40 = 1; 𝑊41 = −𝑗; 𝑊42 = −1; 𝑊43 = 𝑗
OUTPUT
K X1(k) X2(k)
0 1+1(1)+2(1)=4 1+2(1)=3
1 1+1(-j)+2(-1)=-1-j 1+2(-j)=1-2j
2 1+1(-1)+2(1)=2 1+2(-1)=-1
3 1+1(j)+2(-1)= -1+ j 1+2(j)=1+2j
X1(k)={4,-1- j,2,-1+ j}
X2(k)={3,1- 2j,-1,1+2j}

Linear convolution in time domain is given by y(n)=x(n)*h(n)
Linear convolution in K-domain is given by Y(k)= X(k) H(k)
Therefore for the desired result Y(k)= X1(k) X2(k)
Y(k) = {(4)(3) (-1- j)( 1- 2j) (2)(-1) (-1+ j)( 1+2j) }
Therefore , Y(k) = {12,-3+1i,-2,-3-1i}

1 𝑁−1
IDFT of X(k) = x(n) = ∑𝑘=0 𝑋(𝑘 )𝑊𝑁−𝑘𝑛 ; 0≤ 𝑛 ≤N-1
𝑁
1 3
y(n)= ∑𝑘=0 𝑋(𝑘 )𝑊4−𝑘𝑛 ; 0≤ 𝑛 ≤N-1
4
1
y(n) = { 𝑌 (0)𝑊4−0𝑛 + 𝑌(1)𝑊4−1𝑛 + 𝑌(2)𝑊4−2𝑛 + 𝑌(3)𝑊4−3𝑛 } ; 0≤ 𝑛 ≤N-1
4
TWIDDLE FACTOR
2𝜋
𝑗
𝑊4−𝑘𝑛 =ⅇ 𝑁
𝑊40 = 1; 𝑊4−1 = 𝑗; 𝑊4−2 = −1; 𝑊4−3 = −𝑗
OUTPUT
n y(n)
1
0 {12+(-3+1i)(1)+(-2)(1)+( -3-1i)(1)}=1
4
1
1 {12+(-3+1i)(j)+(-2)(-1)+( -3-1i)(-j)}=3
4
1
2 {12+(-3+1i)(-1)+(-2)(1)+( -3-1i)(-1)}=4
4
1
3 {12+(-3+1i)(-j)+(-2)(-1)+( -3-1i)(j)}=1
4
y(n)={1,3,4,4}
The required sequence is given by y(n)={1,3,4,4}

PROGRAM:
%LINEAR CONVOLUTION OF TWO SEQUENCES USING DFT
AND IDFT
x1=input('Enter first sequence = ');

x2=input('Enter second sequence = ');
N=length(x1)+length(x2)-1;
X1=fft(x1,N);
disp('DFT of first sequence = ');
disp(X1);
X2=fft(x2,N);
disp('DFT of second sequence = ');
disp(X2);
Y=X1.*X2;
%TO OBTAIN DESIRED RESULT EACH ELEMENT OF FIRST VECTOR IS TO
MULTIPLIED WITH EACH TERM IN SECOND VECTOR SEPARATELY.
disp('DFT of the Linear convolution of the two sequences');

disp(Y);
y=ifft(Y,N);
disp('Linear convolution using DFT and IDFT');
disp(y);
n=0:1:N-1;
subplot(1,1,1);
stem(n,y);
xlabel('Time index n');
ylabel('Amplitude');
title('Linear convolution using DFT and IDFT');

OUTPUT:
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO FIND LINEAR CONVOLUTION WHEN THE

GIVEN SEQUENCE IS SINGLY SIDED i.e. RIGHT SIDED
Let the two single sided sequences be x1(n) is {2, 3, 4} & x2(n) is {1, 2, 1}
i.e. x1(n) is {2,3,4} &
x2(n) is {1,2,1}
* 2 3 4
1 2 3 4
2 4 6 8
1 2 3 4
y(n) = x(n)*h(n)
Therefore the required result after single sided linear convolution is

y(n) = { 2,7,12,11,4 }

PROGRAM:
%SINGLE SIDED LINEAR CONVOLUTION
X1=input('Enter the first sequence =');
X2=input('Enter the second sequence =');
y=conv(X1,X2);
disp('Linear convolution of X1 and X2 is y =');
disp(y);
subplot(2,1,1);
stem(y);
title('Convolution output');
subplot(2,2,3);
stem(X1);
title('Plot of X1');
subplot(2,2,4);
stem(X2);

OUTPUT:
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO FIND LINEAR CONVOLUTION WHEN THE

GIVEN SEQUENCE IS DOUBLY SIDED i.e. LEFT AND RIGHT
SIDED
* 1 2 3 2 1 3 4
2 2 4 6 4 2 6 8
-3 -3 -6 -9 -6 -3 -9 -12
4 4 8 12 8 4 12 16
-1 -1 -2 -3 -2 -1 -3 -4
0 0 0 0 0 0 0 0
1 1 2 3 2 1 3 4
Therefore the required result after double sided linear convolution is
y(n)={2,1,4,2,6,9,3,2,15,-3,3,4}

PROGRAM:
%DOUBLE SIDED LINEAR CONVOLUTION
X1 =input('Enter the first sequence =');
x1=input('Enter position of zero in the first sequnce = ');
%enter position considering first term as zero not 1.
disp('sequence 1');
disp(X1);
n1 =(length(X1)-(length(X1)+x1)):(length(X1)-(x1+1));
%the above statement helps in determining the range of values
that is taken up by n1
X2 =input('Enter the second sequence =');
x2=input('Enter position of zero in the second sequnce =
');
n2 = (length(X2)-(length(X2)+x2)):(length(X2)-(x2+1));
%the above statement helps in determining the range of values
that is taken up by n2
disp('sequence 2');
disp(X2);
ybegin = n1(1)+n2(1);
yend = n1(length(X1))+n2(length(X2));
ny=[ybegin:yend];
y=conv(X1,X2);
disp('Linear convolution of the two sequence');
disp(y);
subplot(3,2,1:2);
stem(ny,y);
xlabel('Amplitude');
ylabel('Time index n');
subplot(3,2,3:4);
stem(n1,X1);
xlabel('Amplitude');
ylabel('Time index n');
title('Plot of input X1');
subplot(3,2,5:6);
stem(n2,X2);
xlabel('amplitude');
ylabel('time index n');
title('Plot of input X2');

OUTPUT:
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO DETERMINE OUTPUT RESPONSE OF A

DIFFERENCE EQUATION FOR THE FOLLOWING INPUTS
1. COSINE INPUT
2. IMUPLSE INPUT
3. UNIT STEP INPUT
 A difference equation with constant coefficient describes a LTI system.
 Difference equation containing past samples of output leads to recursive
system, whose impulse response is of infinite duration(IIR).for such systems
the impulse response is computed for large value of ‘n’ (say 100).
 The function ‘filter’ evaluates the convolution of an infinite sequence (IIR)
and x[n], which isn’t possible with ‘conv’ function as it requires both the
sequence to be finite.
 The difference equation having only y[n] and present and past samples of
input, represents a system whose impulse response is finite in nature hence
both functions ‘filter’ and ‘conv’ can be used.
Let the difference equation be given by:
y(n)-1.5y(n-1)+y(n-2)=2x(n-2
Therefore,
Y(z)
H(z) =
X(z)
Y(z) 2z−2
H(z) = =
X(z) 1−1.5z−1 +1z−2
Y(z)=X(z)H(z)

Number of samples at output = 100
1. COSINE INPUT
x(n) = cos(0.05Πn) ; 0 ≤ n ≤N-1
1−z−1 cos(0.05Π)
X(z) = ROC ∈ ∀ |𝑧|>1
1−2z−1 cos(0.05Π)+z−2
1−z−1cos(0.05Π) 2z−2
Y(z) = x 1−1.5z−1+1z−2
1−2z−1 cos(0.05Π)+z−2
2. IMPULSE INPUT
x(n) = 𝛿 (n) ; 0 ≤ n ≤N-1
X(z) = 1 ROC ∈ ∀ z
2z−2
Y(z) = 1 x
1−1.5z−1 +1z−2
3. UNIT STEP INPUT
x(n) = u(n) ; 0 ≤ n ≤N-1

1
X(z) = ROC ∈ ∀ |𝑧|>1
1−z−1
1 2z−2
Y(z) = x
−1 1−1.5z−1 +1z−2
1−z

PROGRAM:
%DIFFERENCE EQUATION-COSINE INPUT
N=input('The length of the impulse response required: ');
b=input('The co-efficients of numerator: ');
a=input('The co-efficients of denominator: ');
n=0:N-1;
x=cos(0.05*pi*n);
h=filter(b,a,x);
subplot(211);
stem(n,x);
title('cosine input');
subplot(212);
stem(n,h);
title('Cosine input response');
%DIFFERENCE EQUATION-IMPULSE INPUT
N=input('The length of the impulse response required: ');

n=0:N-1;
x=[1,zeros(1,N-1)];
h=filter(b,a,x);
subplot(211);
stem(n,x);
title('Impulse input');
subplot(212);
stem(n,h);
title('Impulse input response');
%DIFFERENCE EQUATION-UNIT STEP INPUT

N=input('The length of the unit response required: ');

n=0:N-1;
x=ones(1,N);
h=filter(b,a,x);
subplot(211);
stem(n,x);
title('Unit input');
subplot(212);
stem(n,h);
title('Unit input response');
OUTPUT:
COSINE INPUT
WORKSPACE

COMMAND WINDOW
GRAPH

IMPULSE INPUT
WORKSPACE
COMMAND WINDOW

GRAPH

UNIT STEP INPUT

WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO DETERMINE DFT OF A GIVEN SEQUENCE AND

SHOWCASE ITS MAGNITUDE AND PHASE SPECTRUM
 Discrete Fourier Transform (DFT) is used for performing frequency analysis
of discrete time signals.
 DFT gives a discrete frequency domain representation whereas the other
transformations are continuous in frequency domain.
 N point DFT of discrete time signals x (n) is given by
𝑁−1
DFT of x (n) = X (k) = ∑𝑛=0 𝑥 (𝑛)𝑊𝑁𝑘𝑛 ; 0≤ 𝑘 ≤N-1
2𝜋
, where 𝑊𝑁𝑘 is called as Twiddle factor given by 𝑊4𝑘𝑛 = ⅇ −𝑗 𝑁
Determine DFT of {1, 1, 0, and 0} and showcase its magnitude and phase
spectrum.
Let x (n) = {1, 1, 0, 0}
X (k) = 𝑥 (0)𝑊4𝑘0 + 𝑥 (1)𝑊4𝑘1 + 𝑥 (2)𝑊4𝑘2 + 𝑥(3)𝑊4𝑘3 ; 0≤ 𝑘 ≤3
For sequence one,
X (k) = 1 + 1𝑊4𝑘1 + 0𝑊4𝑘2 + 0𝑊4𝑘3 ; 0≤ 𝑘 ≤3
TWIDDLE-FACTOR
2𝜋𝑛
𝑊40 = 1; 𝑊41 = −𝑗; 𝑊42 = −1; 𝑊43 = 𝑗
OUTPUT
K X(k)
0 1+1(1)=2
1 1+1(-j)=-1-j
2 1+1(-1)=0

3 1+1(j)= 1+ j
Therefore the required result after linear convolution is y(n)= {2,1-j,0,1+j}
Magnitude = √ ((real part) 2+ (complex part) 2)
X(k) |X(k)|
2 2
1-j √2
0 0
1+j √2
−1
Phase = tan (complex part/ real part)
X(k) phase
2 0
1-j −𝜋/2
0 0
1+j 𝜋/2

PROGRAM:
%DFT OF A GIVEN SEQUENCE
x1=input('Enter the sequence:');

N=input('Enter the length:');
X=fft(x1,N);
%DFT of the given sequence found using function 'fft'
disp('DFT of the given sequence is ');
disp(X);
n=0:1:N-1;
subplot(2,1,1);
p=abs(X);
disp('amplitude of each term in DFT of the given sequence is ');
disp(p);
%magnitude of each term in DFT
stem(n,p);
xlabel('length');
ylabel('magnitude of X(k)');
title('magnitude spectrum');
q=angle(sym(X));
%if q=angle(X);is used instead of q=angle(sym(X));the phase
angle retrned
%is term of absolute value not in terms of 'PI'
%phase angle of each term in DFT

disp('phase angle of each term in DFT of the sequence is');
disp(q);
subplot(2,1,2);
stem(n,q);
xlabel('length');
ylabel('phase of X(k)');
title('angle spectrum');

OUTPUT:
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM FOR CORSS –CORRELATION AND AUTO

CORRELATION
According to Cross-correlation
−∞
rxy(l)= ∑𝑛=∞ 𝑥(𝑛)y(n − l) ; l = ±1, ±2, ±3, ±4, ±5, ±6, ±7 ….
OR
−∞
rxy(l)= ∑𝑛=∞ 𝑥(𝑛 + 𝑙)y(n) ; l = ±1, ±2, ±3, ±4, ±5, ±6, ±7 ….
Similarly,
−∞
ryx(l)= ∑𝑛=∞ 𝑦(𝑛)x(n − l) ; l = ±1, ±2, ±3, ±4, ±5, ±6, ±7 ….
OR
−∞
ryx(l)= ∑𝑛=∞ 𝑦(𝑛 + 𝑙)x(n) ; l = ±1, ±2, ±3, ±4, ±5, ±6, ±7 ….
Therefore,
rxy(l)= ryx(-l)
We know that,
−∞
X(n)*H(n)= ∑𝑚=∞ 𝑥 (𝑛)y(n − m)
−∞ −∞
Therefore, rxy(l)= ∑𝑛=∞ 𝑥(𝑛)y(n − l) = ∑𝑛=∞ 𝑥 (𝑛)y(−(l − n)) ;
l = ±1, ±2, ±3, ±4, ±5, ±6, ±7 ….
ryx(l)= x(l)*y(-l)
Cross-correlation calculates correlation between two different sequences of a

random process and Autocorrelation is handled as a special case of cross-
correlation.

1.CROSS-CORRELATION
x(l)= [2 -1 3 7 1 2 -3]
y(-l)= [5 -2 1 4 -2 2 -1 1]
ryx(l)= x(l)*y(-l)
2 -1 3 7 1 2 -3
5 -2 1 4 -2 2 -1 1
2 -1 3 7 1 2 -3
-2 1 -3 -7 -1 -2 3 x
4 -2 6 14 2 4 -6 x x
4 -2 6 14 2 4 -6 x x x
8 -4 12 28 4 8 -12 x x x x
2 -1 3 7 1 2 -3 x x x x x
4 -2 6 14 2 4 -6 x x x x x x
10 -5 15 35 5 10 -15 x x x x x x x
10 -9 19 36 -14 33 0 7 13 -18 16 -7 5 -3
ANSWER:

2.AUTOCORRELATION
x(l)= [1 2 3 4]
x(-l)= [4 3 2 1]
ryx(l)= x(l)*x(-l)
1 2 3 4
4 3 2 1
1 2 3 4
2 4 6 8 x
3 6 9 12 x x
4 8 12 16 x x x
4 11 20 30 20 11 4
ANSWER:

PROGRAM:
%GENERAL PROGRAM FOR CROSS CORRELATION
X1 = input('Enter the first sequence = ');
x1=input('Enter position of zero in the first sequnce = ');
%enter position assuming that the leftmost element is beginings as zero and
%the position of zero in the sequence is with respect to this leftmost
element
n1 =(length(X1)-(length(X1)+x1)):(length(X1)-(x1+1));
%(x1+1)is subtracted instead of x1 as position numbering begins from with a
%zero
X2 = input('Enter the second sequence = ');

x2=input('Enter position of zero in the second sequnce = ');
n2 = (length(X2)-(length(X2)+x2)):(length(X2)-(x2+1));
%(x2+1)is subtracted instead of x1 as position numbering begins from with a
%zero
ybegin = n1(1)-n2(length(X2));
%n1(1)is the first element of vector n1
%n2(length(X2) is the last element of vector n2
%notice values of (n1(1)),(n2(length(X2)-1)),ybegin),yend in the workspace
if length(X2)==length(X1)
yend =ybegin+(length(X2)+length(X1))-2;
%this involves subtraction with 2 as yend is found with respect to
%ybegin(which is by itself a position)i.e
%[((length(X2)+length(X1))-1)-1]
else
yend =ybegin+(length(X2)+length(X1))-2+abs((length(X1)-length(X2)));
%although yend=ybegin+(length(X2)+length(X1))-(length(X1)-
length(X2))equals to
%one extra term than required ,its used to accomodate one extra zero
%seen at right most end of plot
end
ny=[ybegin:yend];
if length(X1)>length(X2)
X2=[zeros(1,abs((length(X1)-length(X2)))) X2];
end
%concatenation of 2 matrices
[r,lag]=xcorr(X1,X2);
disp('The cross correlated O/P is as following:');

disp(r);
stem(ny,r);
xlabel('Position');
ylabel('Cross correlated O/P');
title('Cross correlation');

OUTPUT:
CROSS-CORRELATION
WORKSPACE
COMMAND WINDOW

GRAPH

AUTOCORRELATION
WORKSPACE
COMMAND WINDOW
GRAPH


AIM: PROGRAM TO FIND CIRCULAR CONVOLUTION WHEN THE

GIVEN SEQUENCES ARE
1. EQUAL IN LENGTH
2. UNEQUAL IN LENGTH
1. EQUAL IN LENGTH
Let x1(n)= {1,1,2,1} and x2(n)= {1,2,3,4}
N x1(n) x2(n-m)4 SUM

1 1,1,2,1 1,4,3,2 1+4+6+2=13
2 1,1,2,1 2,1,4,3 2+1+8+3=14
3 1,1,2,1 3,2,1,4 3+2+2+4=11
4 1,1,2,1 4,3,2,1 4+3+4+1=12
ANSWER : (13, 14, 11, 12)
Let x1(n)= {1,1,2,1,3} and x2(n)= {1,2,3,4}
N x1(n) x2(n-m)5 SUM

1 1,1,2,1,3 1,0,4,3,2 1+0+8+3+6=18
2 1,1,2,1,3 2,1,0,4,3 2+1+0+4+9=16
3 1,1,2,1,3 3,2,1,0,4 3+2+2+0+12=19
4 1,1,2,1,3 4,3,2,1,0 4+3+4+1+0=12
5 1,1,2,1,3 0,4,3,2,1 0+4+6+2+3=15
ANSWER : (18,16,19,12,15)

PROGRAM:
%GENERAL CIRCULAR CONVOLUTION PROGRAM
X1=input('Enter the first sequence:');
X2=input('Enter the second sequence:');
if length(X1)>length(X2)
b=length(X1);
else
b=length(X2);
end
%b is the length of the largest sequence of the two
y = cconv(X1,X2,b);
%b is required in this as a parameter otherwise the result

consists of
%length(a)+length(B)-1 terms
disp('Linear convolution of X1 and X2 is y =');
disp(y);
subplot(2,2,1:2);
stem(y);
subplot(2,2,3);
stem(X1);
subplot(2,2,4);
stem(X2);

OUTPUT:
EQUAL LENGTH
WORKSPACE
COMMAND WINDOW

GRAPH

UNEQUAL LENGTH
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO DETERMINE CIRCULAR CONVOLUTION USING

DFT AND IDFT GIVEN THE SEQUENCES ARE
1. EQUAL IN LENGTH
1. EQUAL
Let x1(n)={1,1,2,1} and x2(n)={1,2,3,4}

𝑁−1
DFT of x(n) = X(k) = ∑𝑛=0 𝑥(𝑛)𝑊𝑁𝑘𝑛 ; 0≤ 𝑘 ≤N-1
X (k) = 𝑥 (0)𝑊4𝑘0 + 𝑥 (1)𝑊4𝑘1 + 𝑥(2)𝑊4𝑘2 + 𝑥 (3)𝑊4𝑘3 ;0≤ 𝑘 ≤3
For sequence one,
X1(k) = 1 + 1𝑊4𝑘1 + 2𝑊4𝑘2 + 1𝑊4𝑘3 ; 0≤ 𝑘 ≤3
For sequence two,
X2(k) = 1 + 2𝑊4𝑘1 + 3𝑊4𝑘2 + 4𝑊4𝑘3 ; 0≤ 𝑘 ≤3
Therefore, Y (k) = X1(k) X2(k)
Y (k) =(1 + 12) + (3 + 11)𝑊4𝑘1 + (7 + 4)𝑊4𝑘2 + 12𝑊4𝑘3 ; 0≤ 𝑘 ≤3
Y (k) =13+14𝑊4𝑘1 + 11𝑊4𝑘2 + 12𝑊4𝑘3 ; 0≤ 𝑘 ≤3
ANSWER : (13,14,11,12)
TWIDDLE FACTOR
2𝜋
𝑊40 = 1; 𝑊41 = −𝑗; 𝑊42 = −1; 𝑊43 = 𝑗

2. UNEQUAL
Let x1(n)={1,1,0,0} and x2(n)={1,2,0}

𝑁−1
DFT of x(n) = X(k) = ∑𝑛=0 𝑥(𝑛)𝑊𝑁𝑘𝑛 ; 0≤ 𝑘 ≤N-1
X (k) = 𝑥 (0)𝑊4𝑘0 + 𝑥 (1)𝑊4𝑘1 + 𝑥(2)𝑊4𝑘2 + 𝑥 (3)𝑊4𝑘3 ;0≤ 𝑘 ≤3
For sequence one,
X1(k) = 1 + 1𝑊4𝑘1 + 0𝑊4𝑘2 + 0𝑊4𝑘3 ; 0≤ 𝑘 ≤3
For sequence two,
X2(k) = 1 + 2𝑊4𝑘1 + 0𝑊4𝑘2 + 0𝑊4𝑘3 ; 0≤ 𝑘 ≤3
Therefore, Y (k) = X1(k) X2(k)
Y (k) = 1 + 3𝑊4𝑘1 + 2𝑊4𝑘2 + 0𝑊4𝑘3 ; 0≤ 𝑘 ≤3
ANSWER : (1,3,2,0)
TWIDDLE FACTOR
2𝜋
𝑊40 = 1; 𝑊41 = −𝑗; 𝑊42 = −1; 𝑊43 = 𝑗

PROGRAM:
%CIRCULAR CONVOLUTION OF TWO SEQUENCES USING DFT AND IDFT
METHOD.
x1=input('Enter first sequence:');

x2=input('Enter second sequence:');
N=max(length(x1),length(x2));
X1=fft(x1,N);
X2=fft(x2,N);
Y=X1.*X2;
y=ifft(Y,N);
disp('Linear convolution using DFT and IDFT');
disp(y);
n=0:1:N-1;
subplot(1,1,1);
stem(n,y);
xlabel('time');
ylabel('amplitude');
title('circular convolution using DFT and IDFT');

OUTPUT:
EQUAL
WORKSPACE
COMMAND WINDOW

GRAPH

UNEQUAL
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM FOR DESIGNING FILTERS USING BILINEAR

TRANSFORMATION
1. BUTTERWORTH LOWPASS FILTER

2. CHEBYSHEV LOWPASS FILTER

1. BUTTERWORTH LOWPASS FILTER

Pass band edge frequency (fp) = 200 Hz
Pass band attenuation (Kp) = - 1 dB
Stop band edge frequency (fs) = 300 Hz
Stop band attenuation (Ks) = - 4 dB
Sampling frequency = 2000 Hz
ANALOG TO DIGITAL SPECIFICATION:
ω=ΩxT ; where T (i.e. Sampling time) =1/2000 sec

2𝜋×200
ωp=Ωpx T = 2πfpT = = 0.2 π rad
2000
2𝜋×300
ωs=Ωsx T = 2πfsT = = 0.3 π rad
2000
PREWARP THE BAND EDGE DIGITAL FREQUENCY USING T = 1 sec

2 𝜔
Ω’ = × tan
𝑇 2
2 0.2𝜋
Ω p’ = × tan = 0.6498 rad / sec
1 2
2 0.3𝜋
Ω s' = × tan = 1.0191 rad / sec
1 2
DESIGN OF ANALOG FILTER

-Kp
10 10 -1
[log10 { -K }] 100.1 -1
s [log10 { }]
10 10 -1 100.4 -1
N = Ωp = 0.6498 = 1.96 ≈ 2
[2 log( )] [2 log( )]
Ωs 1.0191

Ωp 0.6498
Ωc = 1 = 1 = 0.9109 rad /sec
-Kp 2𝑁 0.1 4
(10 -1)
(10 10 -1)
Ωs 1.0191
Ωc = 1 = 1 = 0.9190 rad /sec
-Ks 2𝑁 0.4 4
(10 -1)
(10 10 -1)

Therefore the idealized normalized equation of the filter is given by
1
𝐻2 (𝑠) =
[𝑠 2 + √2𝑠 + 1]
For required analog low pass filter
Ha(s) = H d (s) s->s/0.9190

1
𝐻𝑎 (𝑠) = 2
𝑠 𝑠
[{ } + √2 { } + 1]
0.919 0.919
0.844561
Ha(s) =
s2 +1.2996 s1 +0.844561
APPLYING BLT TO GET H(z)

2 1−𝑧 −1
H(z) = H a (s) 𝑠 → × [ ]
𝑇 1+𝑧 −1
0.844561
H(z) = 2 1
2 1−𝑧−1 2 1−𝑧−1
{1×[ ]} +1.2996 { ×[ ]} +0.844561
1+𝑧−1 1 1+𝑧−1
0.1135 𝑧 2+0.2269 𝑧 1+0.1135

H(z) =
𝑧 2 −0.8478𝑧+0.3016
2. CHEBYSHEV LOWPASS FILTER
Pass band edge frequency (fp) = 100 Hz

Pass band attenuation (Kp) = - 2 dB
Stop band edge frequency (fs) = 500 Hz
Stop band attenuation (Ks) = - 20 dB

Sampling frequency = 4000 Hz
ANALOG TO DIGITAL SPECIFICATION:
ω =ΩxT ; where T (i.e. Sampling time) =1/4000 sec

2𝜋×100
ωp = Ωp x T = 2πfpT = = 0.05 π rad
4000
2𝜋×500
ωs = Ωs x T = 2πfsT = = 0.25 π rad
4000
PREWARP THE BAND EDGE DIGITAL FREQUENCY USING T = 1 sec

2 𝜔
Ω‘ = × tan
𝑇 2
2 0.05𝜋
Ωp‘= × tan = 0.1574 rad / sec
1 2
2 0.25𝜋
Ωs‘= × tan = 0.8284 rad / sec
1 2
DESIGN OF ANALOG FILTER
2
1
Kp = 20𝑙𝑜𝑔 = -2 ⇒∈ = √1010 − 1 = 0.7648
√1+∈2
1
𝛿p = 1− = 0.20567
√1+∈2
1
Ks = 20𝑙𝑜𝑔 𝛿𝑠 = -2 ⇒ 𝛿𝑠 =
10
= 0.1
Ωp′ 0.1574
K = = = 0.19
Ωs′ 0.8284
(1−𝛿𝑝)2 −1
D=√ = 0.0768
𝛿𝑠 2 −1

1
𝑐𝑜𝑠ℎ−1 ( )
𝑑
N≥ 1 ⇒ 𝑁 ≥ 1.39
𝑐𝑜𝑠ℎ−1 (𝑘)
N = 2
1 1+√1+ɛ2 1 1 1+√1+ɛ2 −1
A = { } − {
𝑁 }𝑁 = 0.568
2 ɛ 2 ɛ
1 1+√1+ɛ2 1 1 1+√1+ɛ2 −1
B = { }𝑁 + { }𝑁 = 1.15
2 ɛ 2 ɛ
π
𝜎𝑘 = −a sin((2k − 1) )
2N
π
𝛺𝑘 = −a cos ((2k − 1) ) ; Where k=1,2,3 …. 2N
2N
K 𝜎𝑘 𝛺𝑘
1 -0.401 0.813
2 -0.401 -0.813
Therefore S1= - 0.401 + j 0.813 & S2= - 0.413 - j 0.813
V2(s) = s2 + 0.802 s + 0.82177
b0 = 0.82177
0.82177
KN = = 0.653
√1+𝜖 2
0.653
H2(s) =
s2 +0.802s+0.82177
Ha(s)=H2(s)s->s/(0.1574)
0.653
Ha(s)= s s
( )2 +0.644( )+0.7074
0.1574 0.1574

0.01618
Ha(s)=
s2 +0.126s+0.0204
APPLYING BLT TO GET H(z)

2 1−𝑧 −1
H(z) = H a (s) 𝑠 → × [ ]
𝑇 1+𝑧 −1
0.01618
H(z) = 2 1
2 1−𝑧−1 2 1−𝑧−1
{1×[ ]} +0.126 { ×[ ]} +0.0204
1+𝑧−1 1 1+𝑧−1
0.00378 𝑧 2+0.0076 𝑧 1 +0.00378

H(z) =
𝑧 2 −1.863𝑧+0.8820

PROGRAM:
%DIGITAL BUTTERWORTH LOWPASS FILTER USING
BILINEAR TRANSFORM.
% Digital Butterworth Lowpass Filter using Bilinear
Transformation
% inputs : 200, 1, 300, 4, 2000
fp = input('Enter pass band edge frequency in Hz = ');
Ap = input('Enter pass band ripple in dB = ');
fs = input('Enter stop band edge frequency in Hz = ');
As = input('Enter stop band ripple in dB = ');
Fs = input('Enter the sampling frequency in Hz = ');
W1 = 2*pi*fp/Fs;
W2 = 2*pi*fs/Fs;
% Prewarping
Wp = 2*tan(W1/2);
Ws = 2*tan(W2/2);
[n, Wc] = buttord(Wp, Ws, Ap, As, 's');

disp('The order of the filter is N = ');
disp(n);
disp('The cutoff frequency is = ');
disp(Wc);
disp('The Analog filter numerator and denominator coefficients

are ');
[num, den] = butter(n, Wc, 's');
disp(num);
disp(den);
disp('The Digital filter numerator and denominator coefficients
are ');
[b, a] = bilinear(num, den, 1);
freqz(b,a, 512, Fs); % 512 point complex frequency response
vector
disp(b);
disp(a);

%DIGITAL CHEBYSHEV LOWPASS FILTER USING BILINEAR

TRANSFORM.
% Digital Chebyshev-I Lowpass Filter using Bilinear
Transformation
Ap = input('Enter pass band ripple in dB = ');
As = input('Enter stop band ripple in dB = ');
Fs = input('Enter the sampling frequency in Hz = ');
W1 = 2*pi*fp/Fs;
W2 = 2*pi*fs/Fs;
% Prewarping
Wp = 2*tan(W1/2);
Ws = 2*tan(W2/2);
[n, Wc] = cheb1ord(Wp, Ws, Ap, As, 's');

disp(n);
disp(Wc);
disp('The Analog filter numerator and denominator coefficients

are ');
[num, den] = cheby1(n, Ap, Wc, 's');
disp(num);
disp(den);
disp('The Digital filter numerator and denominator coefficients
are ');
[b, a] = bilinear(num, den, 1);
freqz(b,a, 512, Fs); % 512 point complex frequency response
vector
disp(b);
disp(a);

OUTPUT:
BLT -BUTTERWORTH LOWPASS
WORKSPACE
COMMAND WINDOW

GRAPH

BLT-CHEBYSHEV LOWPASS
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM TO DESIGN AN ANALOG CHEBYSHEV FILTER
1. LOWPASS FILTER
2. HIGH PASS FILTER
3. BANDPASS FILTER
For a Chebyshev filter the magnitude response is given by
Then,

Where,
Order of filter is given by
Where,
1. LOWPASS FILTER
Pass band edge frequency (fp) = 2Hz i.e. Ωp = 4 𝛱 rad/sec

Pass band attenuation (Kp) = -3dB
Stop band edge frequency (fs) = 5Hz i.e. Ωs =10 𝛱 rad/sec
Stop band attenuation (Ks) = -10dB

N ≥ 1.12
N = 2
1 1+√1+ɛ2 1 1 1+√1+ɛ2 −1
A = { }𝑁 − { }𝑁 = 0.4560239
2 ɛ 2 ɛ
1 1+√1+ɛ2 1 1 1+√1+ɛ2 −1
B = { }𝑁 + { }𝑁 = 1.09907
2 ɛ 2 ɛ

π
𝜎𝑘 = −a sin((2k − 1) )
2N
π
𝛺𝑘 = −a cos ((2k − 1) ) ; Where k=1,2,3 …. 2N
2N
K 𝜎𝑘 𝛺𝑘
1 -0.322 0.777
2 -0.322 -0.777
Therefore S1= - 0.322 + j 0.777 & S2= - 0.322 - j 0.777
0.5008
H2(s)=
s2 +0.644s+0.7074
0.7074
KN =
√1+𝜖 2
Ha(s)=H2(s)s->s/(4π)
0.5008
Ha(s)= s s
(4π)2+0.644(4π)+0.7074
79.083
Ha(s)=
s2 +8.093s+111.708
2. HIGHPASS FILTER
No theory required
3. BAND PASS FILTER

No Theory required

PROGRAM:
%DESIGN OF ANALOG CHEBYSHEV LOW PASS FILTER
rp = input('Enter pass band ripple in dB = ');
rs = input('Enter stop band ripple in dB = ');
Wp = 2*pi*fp;
disp(Wp);
Ws = 2*pi*fs;
disp(Ws);
[n, Wp] = cheb1ord(Wp, Ws, rp, rs, 's');
disp(n);
 Wp is the Chebyshev natural frequency to use with
CHEBY1 to achieve the required specifications.
disp(Wp);
disp('The numerator and denominator coefficients of the analog
filter are ');
% for low pass
[b, a] = cheby1(n, rp, Wp,'low','s');
disp('Numerator co-efficients :');
disp(b);
disp('Denominator co-efficients :');
disp(a);
[h, w] = freqs(b, a);
m = 20*log10(abs(h));
an = (angle(h)*180/pi);
subplot(211);
plot(w, m);
title('Frequency response of analog Chebyshev filter');
xlabel('Frequency in rad/s');
ylabel('Magnitude in dB');
grid on;
subplot(212);
plot(w, an);
title('Phase response of analog Chebyshev filter');
xlabel('Frequency in rad/sec');
ylabel('Phase(degrees)');
grid on;

OUTPUT:
CHEBYSHEV LOWPASS
WORKSPACE
COMMAND WINDOW

GRAPH

AIM: PROGRAM FOR DESIGNING AN ANALOG BUTTERWORTH

FILTER
1. LOWPASS FILTER
2. HIGH PASS FILTER
3. BANDPASS FILTER
For a butterworth filter
Magnitude frequency response is given by
Order of filter is given by
Cutoff frequency is given by
, using pass band parameters

OR
, using stop band parameters

OR
the mean of the above two can considered depending on requirement

ANALOG TO ANALOG CONVERSION

1. LOWPASS FILTER
Pass band edge frequency (fp) = 2Hz i.e. Ωp = 4 Π rad/sec
Stop band edge frequency (fs) = 5Hz i.e. Ωs =10 Π rad/sec
10π
Ωs = = 2.5 rad/sec (backward transformation for low pass filter)
4𝜋
New normalized Ω p = 1
New normalized Ω s = 2.5

-Kp
10 10 -1
[log10 { -K }] 100.3 -1
s [log10 { }]
10 10 -1 101 -1
N = Ωp = 1 = 1.2015 ≈ 2
[2 log( )] [2 log( )]
Ωs 2.5
Ωp 1
Ωc = 1 = 1 = 1 rad /sec
-Kp 2𝑁 0.3 4
(10 -1)
(10 10 -1)

Ωs 2.5
Ωc = 1 = 1 = 1.4434 rad /sec
-Ks 2𝑁 1 4
(10 -1)
(10 10 -1)

1
𝐻2 (𝑠) =
[𝑠 2 + √2𝑠 + 1]
Required normalized Low pass transform is given by
H d (s) = H2(s) s->s/1.4434
(In order to get a curve which satisfy cutoff frequency closely we use Ωc based on cutoff parameter
instead of one based on pass band based cutoff)
Ha(s) = H d (s) s->s/4π = H2(s) s->s / (4π x 1.4434) = H2(s) s->s/18.138
1
𝐻𝑎 (𝑠) = 2
𝑠 𝑠
[{ } + √2 { } + 1]
18.138 18.138
328.9868
Ha(s) =
s2 +25.6510s1 +328.9868

2. HIGHPASS FILTER
Pass band edge frequency (fp) = 5Hz i.e. Ωp = 10 𝛱 rad/sec
Stop band edge frequency (fs) = 2Hz i.e. Ωs =4 𝛱 rad/sec
Normalized filter
10π
Ωs = = 2.5 rad/sec (backward transformation for high pass filter)
4𝜋

New normalized Ω p = 1
New normalized Ω s = 2.5

-Kp
10 10 -1
[log10 { -K }] 100.3 -1
s [log10 { }]
10 10 -1 101 -1
N = Ωp = 1 = 1.2015 ≈ 2
[2 log( )] [2 log( )]
Ωs 2.5
Ωp 1
Ωc = 1 = 1 = 1 rad/sec
-Kp 2𝑁 0.3 4
(10 -1)
(10 10 -1)
Ωs 2.5
Ωc = 1 = 1 = 1.4434 rad/sec
-Ks 2𝑁 1 4
(10 -1)
(10 10 -1)
1
𝐻2 (𝑠) =
[𝑠 2 + √2𝑠 + 1]
H d (s) = H2(s) s->s/1.4434
For required analog high pass filter
Ha(s) = H2(s) s->10π / 1.4434s i.e. H2(s) s->21.765 /s

1
𝐻𝑎 (𝑠) =
21.765 2 21.765
[{ } +{ } + 1]
𝑠 𝑠
s2
Ha(s) =
s2 +30.781s1 +473.73
3. BANDPASS FILTER
Lower pass band edge frequency (f l) = 50Hz i.e. Ω l = 100π rad/sec
Higher pass band edge frequency (f u) = 20kHz i.e. Ω u = 40000π rad/sec
Pass band attenuation (Kp) = -3.0103 dB
Lower stop band edge frequency (f1) = 20Hz i.e. Ω1 =40π rad/sec
Lower stop band edge frequency (f2) = 45kHz i.e. Ω2 =90000π rad/sec
Ωc = 4 𝛱/(100.3-1)^0.25 = 12.581 rad/sec
OR

Ωc = 10 𝛱/(101-1)^0.25 = 18.138 rad/sec
−Ω1 2 +Ω𝑙 Ω𝑢
A = = 2.51
Ω1 (Ω𝑙 −Ω𝑢 )
Ω2 2 −Ω𝑙 Ω𝑢
B = = 2.25
Ω2 (Ω𝑢 −Ω𝑙 )
Ωs = Min{|A|,|B|} = 2.25
-Kp
10 10 -1
[log10 { -K }] 100.30103 -1
s [log10 { }]
10 10 -1 102 -1
N = Ωp = 1 = 2.83 ≈ 3
[2 log( Ωs )] [2 log(2.25)]
Ωp 1
Ωc = 1 = 1 = 0.99979 rad/sec
-Kp 2𝑁 0.30103 6
(10 -1)
(10 10 -1)

Ωs 2.25
Ωc = 1 = 1 = 1.0461 rad/sec
-Ks 2𝑁 2 6
(10 -1)
(10 10 -1)


1
𝐻3 (𝑠) =
[𝑠 3 + 2𝑠 2 + 2𝑠 + 1]
H d (s) = H2(s) s->s/1.046
Ha(s) = H3 (s) 𝑠2 +Ω𝑙 Ω𝑢 = H3 (s) 𝑠2 +3.949 ×107

𝑠→ 𝑠→
𝑠(Ω𝑙−Ω𝑢 ) 𝑠(1.2538×105 )
Therefore the required band pass filter is given by
1.9695 x 1015 s3
Ha(s) = 6
s +2.51 x 105 s5 +3.154 x 1010 s4 +1.989 x 1015 s3
+1.2453 x 1018 s2 +3.9073 x 1020 s+6.1529 x 1022

PROGRAM:
%ANALOG BUTTERWORTH LOW PASS FILTER
Wp = 2*pi*fp;
disp('pass band edge frequency in rad/sec:');
disp(Wp);
Ws = 2*pi*fs;
disp('stop band edge frequency in rad/sec:');
disp(Ws);
[n, Wn] = buttord(Wp, Ws, rp, rs,'s');
disp(n);
disp(Wn);
filter are ');
[b, a] = butter(n,Wn,'low','s');
disp(b);
disp(a);
m = 20*log10(abs(h));
subplot(211);
plot(w, m);
title('Frequency response of analog butterworth filter');
grid on;
subplot(212);
plot(w, an);
title('Phase response of analog Butterworth filter');
grid on;

%ANALOG BUTTERWORTH HIGH PASS FILTER

Wp = 2*pi*fp;
disp('pass band edge frequency in rad/sec:');
disp(Wp);
Ws = 2*pi*fs;
disp('stop band edge frequency in rad/sec:');
disp(Ws);
[n, Wn] = buttord(Wp, Ws, rp, rs,'s');
disp(n);
disp(Wn);
filter are ');
[b, a] = butter(n,Wn,'high','s');
disp(b);
disp(a);
m = 20*log10(abs(h));
subplot(211);
plot(w, m);
grid on;
subplot(212);
plot(w, an);
grid on;

%ANALOG BUTTERWORTH BANDPASS FILTER
fp1 = input('Enter lower pass band edge frequency in Hz = ');

fp2 = input('Enter higher pass band edge frequency in Hz = ');
fs1 = input('Enter stop band edge frequency in Hz = ');
fs2 = input('Enter stop band edge frequency in Hz = ');
Wp = 2*pi.*[fp1 fp2];
% similarly Wp = 2*pi*[fp1 fp2];
disp('pass band frequncies are:');
disp(Wp);
disp('stop band frequncies are:');
Ws = 2*pi.*[fs1 fs2];
% similarly Ws = 2*pi*[fs1 fs2];
disp(Ws);
[n, Wn] = buttord(Wp, Ws, rp, rs, 's');
disp(n);
disp(Wn);
disp('The numerator & denominator coefficients of the filter is');
[b, a] = butter(n,Wn,'s');
disp(b);
disp(a);
m = 20*log10(abs(h));
subplot(211); plot(w, m);
grid on;
subplot(212); plot(w, an);
grid on;

OUTPUT:
BUTTERWORTH LOWPASS
WORKSPACE
COMMAND WINDOW

GRAPH

BUTTERWORTH HIGHPASS
WORKSPACE
COMMAND WINDOW

GRAPH

BUTTERWORTH BANDPASS
WORKSPACE

COMMAND WINDOW

GRAPH

AIM: PROGRAM FOR DESIGNING AN FINITE IMPULSE RESPONSE

FILTER DEPENDING ON THE SPECIFICATION
THEORY:
The window function that satisfies the pass band or stop band specifications is
selected (refer table). The number of coefficients is determined using the appropriate
relationship between the filter length and transition bandwidth.
Filter coefficients are given by
h(n) = hd(n) w(n)
where hd(n) is the ideal desired impulse response of the filter which is obtained by
inverse Fourier transform of the desired frequency response H D(ω) and w(n) is the
window function.
The desired filter can be a low pass, high pass, band pass or band reject filter.
Routine fir1 in its various forms is used to design filters based on various windows.
if you need help use the matlab help or happy googling

PROGRAM:
%PLOT THE MAGNITUDE RESPONSE AND CHECK THE
SPECIFICATION OF THE OBTAINED FILTER
1. Pass band edge frequency = 21 KHz

Stop band edge frequency = 4 KHz
Pass band attenuation = 2dB
Stopband attenuation = 40dB
Sampling frequency = 20Khz
clc
clear
closeall
% Pass band and Stop band edge frequencies
fp =2000;
fs=4000;
Fs=20000; % sampling frequency
fn=Fs/2; %normalizing to half the sampling frequency
fc=(fs+fp)/2/fn; %cut off frequency (normalized to fn)
%normalized transition width
tw=(fs-fp)/Fs;
N=ceil(4/tw);
% as stop band attenuation is = 40db
wn=harming(N);% window coefficients
%calculate truncated ideal impulse response coeffs
hd =fir1(N-1,fc,boxcar(N));
% to obtain Windowed coefficients
hn=fir1(N-1,fc,wn);
% Frequency response
[H,f] = freqz(hn,1,512,Fs);
mag = 20*log10(abs(H));
plot(f,mag)
grid on
xlabel(‘Frequency(Hz)’);
ylabel(‘Magnitude response (dB)’);
title('Frequency response of BPF’);

2. Passband = 150 - 250 Hz

Transition width = 50 Hz
Passbandripple = 0.1 dB
Stopband attenuation = 60dB
Sampling frequency = 1 KHz
clc
clear
closeall
%Passband and transition width
fp1=150;
fp2=250;
tw=50;
Fs=1000; % sampling frequency
fn=Fs/2; % normalised to half the sampling frequency
fc1=(fp1-(tw/2))/fn; % cutoff frequency(normalised to fn)
fc2=(fp2+(tw/2))/fn;
tw=tw/Fs; %Normailsed transition width
% stopband attenuation is 60db so we choose blackman window
N=ceil(6/tw);
wn=blackman(N);
%calculate truncated ideal impulse response coeffs
hn=fir1(N-1,[fcl,fc2),boxcar(N));
% obtain windowed coefficients
hn=firl(N-1,[fcl,fc2],wn);
[H,f]=freqz(hn,1,512,Fs);
mag =20*log10(abs(H));
plot(f,mag)
grid on;
xlabel(‘Frequency(Hz)’);
ylabel('Magnitude response (dB)’);
title(‘Frequency response of BPF’);

%FIR FILTER USING HAMMING WINDOW(KNOWN WINDOW

LENGTH)
N=input(‘Enter the window length = ’);

fc=input(‘Enter the cut-off frequency = ’);
fs=input('Enter the sampling frequency = ');
wc=2πfc/fs;
wh=(hamming(N));
b=fir1(N-1,wc,wh);
[h,omega]=freqz(b,1,256);
mag=20*log10(abs(h));
plot(omega/pi,mag);
grid on;
xlabel(‘Frequency in radians in terms of pi’);
ylabel(‘Gain in dB’);
title(‘Response of FIR Filter using Hamming Window’);
disp(‘Hamming window co-efficients’),
disp(wh);
disp(‘FIR Filter output’);
disp(b);
%FIR FILTER USING HAMMING WINDOW(UNKNOWN WINDOW

LENGTH)
wp=input(‘Passband frequency =’);

ws=input(‘Stopband frequency =’);
wt= ws-wp:
N=(8/wt)+1
N=round(N);
wh=(hamming(N));
b=fir1(N-1,wp,wh);
disp(‘impulse response co-efficients =’);
disp(b);
freqz(b);
title(‘Response of FIR filter using Hamming window’);

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DSP Lab Manual TCL 56

DIGITAL SIGNAL PROCESSING LAB

Dept. of Telecommunication Engineering MSRIT

Prerequisite Courses with codes: Signals & Systems (TC45)

Verification of sampling theorem. CO1

Impulse response of a given system CO1

Linear convolution of two given sequences. CO1

Circular convolution of two given sequences CO2

Dept. of Telecommunication Engineering MSRIT

Autocorrelation of a given sequence and

Design and implementation of FIR filter to meet

Design and implementation of IIR filter to meet

Computation of N- Point DFT of a given sequence.

Computation of DFT and FFT of a given sequence CO5

Realization of an FIR and IIR filter to meet given

Dept. of Telecommunication Engineering MSRIT

COURSE DESIGN, DELIVERY AND ASSESMENT

Prerequisite Courses with codes : Signals & Systems(TC405)

1 Verification of sampling theorem. 2 hr

2 Impulse response of a given system 2 hr

3 Linear convolution of two given sequences. 2 hr

4 Circular convolution of two given sequences 2 hr

Autocorrelation of a given sequence and verification

Dept. of Telecommunication Engineering MSRIT

Cross correlation of given sequences and verification

to plot magnitude and phase spectrum.

Design and implementation of IIR filter to meet given

Linear convolution of two given sequences using

Computation of N- Point DFT of a given sequence

Realization of an FIR and IIR filter to meet given

1. Oppenheim & Schaffer, Discrete Time Signal Processing, PHI, 2003.

Dept. of Telecommunication Engineering MSRIT

4. Dr.D. Ganesh Rao, Vineeta P Gejji, Digital Signal Processing-, 2E,

SUBJECT WILL MAP TO PEO1, PEO2 AND PEO3.

Course Assessment and Evaluation:

What To When/ Where Max Evidence Contributing

session Lab Records,

End of the Feedback

Dept. of Telecommunication Engineering MSRIT

Mapping Course Outcomes with Program Outcomes:

Course Programme Outcomes (PO) Program Specific

Justification for CO-PO mapping:

1) CO1 maps to PO1,2,4,5, 9

a. PO1: DFT - IDFT and their properties are derived requiring

d. PO9: CO1 gives an ability to work effectively as an individual and

Dept. of Telecommunication Engineering MSRIT

2) CO2 maps to PO1,2,4 ,5, 9

3) CO3 maps to PO1,2,4,5, 9

4) CO4 maps to PO1,2,4,5, 9,10,11,12

Dept. of Telecommunication Engineering MSRIT

c. PO4: Various processing like Extended parallelism and static super

5) CO5 maps to PO1,2,,4,5, 9,10,11,12

Dept. of Telecommunication Engineering MSRIT

AIM: PROGRAM FOR GENERATING VARIOUS FORMS OF SIGNALS

%UNIT STEP SIGNAL

%SINE WAVE SIGNAL

Dept. of Telecommunication Engineering MSRIT

%SQUARE WAVE SIGNAL

Dept. of Telecommunication Engineering MSRIT

Dept. of Telecommunication Engineering MSRIT

Dept. of Telecommunication Engineering MSRIT

AIM: PROGRAM TO PROVE SAMPLING THEOREM FOR A

Dept. of Telecommunication Engineering MSRIT