DIGITAL SIGNAL PROCESSING LAB

LABORATORY MANUAL

B.TECH. Semester -V

Subject Code: KEC-553

Session: 2023-24, Odd Semester

DRONACHARYA GROUP OF INSTITUTIONS

DEPARTMENT OF ECE
#27 KNOWLEDGE PARK 3
GREATER NOIDA

AFFILIATED TO Dr. A P J ABDUL KALAM TECHNICAL UNIVERSITY

LUCKNOW
 List of Experiments mapped with COs

S. No. Name of the Experiments COs

1. To verify linear convolution on DSP Processors. CO1

2. To verify the circular convolution on DSP Processors. CO1

3. To verify N-point DFT & IDFT on DSP Processors. CO3

4. To verify N-point FFT algorithm on DSP Processors. CO2

To design FIR filter (LP/HP/BP/BR) using windowing technique
5. a) Using Rectangular window CO4
b) Using Triangular window
To design FIR filter (LP/HP/BP/BR) using windowing technique
a) Using Hamming window
6. CO4
b) Using Hanning window
c) Using Blackmann window
7. To design FIR filter (LP/HP/BP/BR) using Kaiser window. CO4
To find the frequency response of analog Butterworth prototype filters
8. CO4
(LP/HP/BP/BR).
To find the frequency response of analog Chebyshev prototype filters
9. (LP/HP/BP/BR). CO4

To Implement IIR Butterworth filter (LP/HP/BP/BR) using

10. CO4
transformation techniques.
To Implement IIR Chebyshev filter (LP/HP/BP/BR) using
11. CO4
transformation techniques.
12. Design of FIR filters using frequency sampling method. CO2
Experiment-1 LINEAR CONVOLUTION
AIM: To verify Linear Convolution.

EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 4.0 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe, 5V DC supply

THEORY:
Convolution is a formal mathematical operation, just as multiplication, addition, and
integration. Addition takes two numbers and produces a third number, while convolution
takes two signals and produces a third signal. Convolution is used in the mathematics of
many fields, such as probability and statistics. In linear systems, convolution is used to
describe the relationship between three signals of interest: the input signal, the impulse
response, and the output signal.

In this equation, x1(k), x2(n-k) and y(n) represent the input to and output from the system at
time n. Here we could see that one of the input is shifted in time by a value every time it is
multiplied with the other input signal. Linear Convolution is quite often used as a method of
implementing filters of various types.

1.1 PROGRAM:
#include<stdio.h>
int x[15],h[15],y[15];
main()
{
inti,j,m,n;
printf("\n Enter value for m");
scanf("%d",&m);
printf("\n Enter value for n");
scanf("%d",&n);
printf("Enter values for input x(n)\n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf("Enter values for input h(n)\n");
for(i=0;i<n;i++)
scanf("%d",&h[i]);
//Padding of Zeros//
for(i=m;i<=m+n-1;i++)
x[i]=0;
for(i=n;i<=m+n-1;i++)
h[i]=0;
//Convolution Operation//
for(i=0;i<m+n-1;i++)
{
y[i]=0;
for(j=0;j<=i;j++)
{
y[i]=y[i]+(x[j]*h[i-j]);
}
//Displaying the Output//
}
for(i=0;i<m+n-1;i++)
printf("\n The value of Output y[%d]=%d",i,y[i]);
}

Result:
enter value for m: 4
enter value for n: 4
Enter values for input x(n):
1
2
3
4
Enter Values for input h(n):
1
2
3
4
The value of Output y[0]=1
The value of Output y[1]=4
The value of Output y[2]=10
The value of Output y[3]=20
The value of Output y[4]=25
The value of Output y[5]=24
The value of Output y[6]=16
Graph:

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 Digital Signal Processing Lab Manual (KEC 553)

1.2 % MATLAB program for linear convolution

%linear convolution program

clc;
clear all;
close all;
disp('linear convolution program');

x=input('enter i/p x(n):');

m=length(x);
h=input('enter i/p h(n):');
n=length(h);

x=[x,zeros(1,n)];
subplot(2,2,1), stem(x);
title('i/p sequence x(n)is:');
xlabel('--- >n');
ylabel('--- >x(n)');grid;

h=[h,zeros(1,m)];
subplot(2,2,2), stem(h);
title('i/p sequence h(n)is:');
xlabel('--- >n');
ylabel('--- >h(n)');grid;
disp('convolution of x(n) & h(n) is y(n):');
y=zeros(1,m+n-1);
for i=1:m+n-1
y(i)=0;
for j=1:m+n-1
if(j<i+1)
y(i)=y(i)+x(j)*h(i-j+1);
end
end
end
y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is :');
xlabel('--- >n');
ylabel('--- >y(n)');grid;

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 Digital Signal Processing Lab Manual (KEC 553)

Graph:

Result : Hence, linear convoluion of two sequences has been verified using CCS and
MATLAB.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-2 CIRCULAR CONVOLUTION

AIM: To verify Circular Convolution.

EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply

THEORY
Circular convolution is another way of finding the convolution sum of two input
signals. It resembles the linear convolution, except that the sample values of one of the input
signals is folded and right shifted before the convolution sum is found. Also note that circular
convolution could also be found by taking the DFT of the two input signals and finding the
product of the two frequency domain signals. The Inverse DFT of the product would give the
output of the signal in the time domain which is the circular convolution output. The two
input signals could have been of varying sample lengths. But we take the DFT of higher
point, which ever signals levels to. For eg. If one of the signal is of length 256 and the other
spans 51 samples, then we could only take 256 point DFT. So the output of IDFT would be
containing 256 samples instead of 306 samples, which follows N1+N2 – 1 where N1 & N2
are the lengths 256 and 51 respectively of the two inputs. Thus the output which should have
been 306 samples long is fitted into 256 samples. The
256 points end up being a distorted version of the correct signal. This process is called
circular convolution.

3.1 Program:
#include <stdio.h>
int n1,n2,i,j,n,c,x[10]=0,h[10]=0,y[10]=0,a[10]=0;
main()
{
printf("Enter the length of the sequence n1: ");
scanf("%d",&n1);
printf("Enter the length of the sequence n2: ");
scanf("%d",&n2);
printf("Enter the first sequence: ");
for(i=0;i<n1;i++)
{
scanf("%d",&x[i]);
}
printf("Enter the second sequence: ");
for(i=0;i<n2;i++)

{
scanf("%d",&h[i]);
}
if(n1>n2)
n=n1;
else
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 Digital Signal Processing Lab Manual (KEC 553)
n=n2;
a[0]=h[0];
for(i=1;i<n;i++)
{
a[i]=h[n-i];
}
for(j=0;j<n;j++)
{ y[j]=
0;
c=x[0];
for(i=0;i<n;i++)
{
y[j]=(y[j]+(x[i]*a[i]));
x[i]=x[i+1];
}
x[n-1]=c;
printf("result is %d /n",y[j]);
}
}
Result:
Enter the length of the sequence n1: 4
Enter the length of the sequence n2: 3
Enter the first sequence: 1
2
3
4
Enter the second sequence: 1
2
3
The Circular Convolution is
18 16 10 16
Graph:

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 Digital Signal Processing Lab Manual (KEC 553)

3.2 %Circular Convolution Program

clc;
clear all;
close all;
disp('circular convolution program');

x=input('enter i/p x(n):');

m=length(x);
h=input('enter i/p sequence h(n)');
n=length(h);

subplot(2,2,1), stem(x);
title('i/p sequencce x(n)is:');
xlabel('--- >n');
ylabel('--- >x(n)');grid;

subplot(2,2,2), stem(h);
title('i/p sequencce h(n)is:');
xlabel('--- >n');
ylabel('--- >h(n)');grid;

disp('circular convolution of x(n) & h(n) is y(n):');

if(m-n~=0)
if(m>n)
h=[h,zeros(1,m-n)];
n=m;
end
x=[x,zeros(1,n-m)];
m=n;
end

y=zeros(1,n);
y(1)=0;
a(1)=h(1);

for j=2:n
a(j)=h(n-j+2);
end

%ciruclar conv
for i=1:n
y(1)=y(1)+x(i)*a(i);
end

for k=2:n
y(k)=0;
% circular shift
for j=2:n
x2(j)=a(j-1);
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 Digital Signal Processing Lab Manual (KEC 553)
end
x2(1)=a(n);

for i=1:n
if(i<n+1)
a(i)=x2(i);
y(k)=y(k)+x(i)*a(i);
end
end
end
y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is:');
xlabel('--- >n');
ylabel('--- >y(n)');grid;

Graph :

Result : Hence Circular convoluion of two sequences has been verified using CCS and
MATLAB

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-3. DISCRETE FOURIER TRANSFORM

AIM: To verify N-point DFT & IDFT of discrete time sequence.

EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply
THEORY:

DFT of a discrete time signal transforms the signal in time domain into frequency domain
signal.

DFT EQUTION:

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

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 Digital Signal Processing Lab Manual (KEC 553)

Note:
1 It is periodic. (i.e. it goes round and round the circle !!)
2 That the vectors are symmetric
3 The vectors are equally spaced around the circle
The inverse discrete Fourier transform (IDFT) is given by:

3.1 DFT
Program

#include<stdio.h>
#include<math.h>
int x[32],N,k,n,i;
float
pi=3.1416,sumre=0,sumim=0,out_real[8]={0.0},out_imag[8]={0.0},m[20]={0.0},p[20]={0.0};
main()
{
printf ("enter lenght\n");
scanf ("%d",&N);
printf("enter the first sequence\n");
for(i=0;i<N;i++)
scanf ("%d",&x[i]);
for(k=0;k<N;k++)
{
sumre=0;
sumim=0;
for(n=0;n<N;n++)
{
sumre=sumre+x[n]*cos(2*pi*k*n/N);
sumim=sumim-x[n]*sin(2*pi*k*n/N);
}
out_real[k]=sumre;
out_imag[k]=sumim;
printf("x[%d]=\t%f\t+j*\t%f\n",k,out_real[k],out_imag[k]);
m[k]=sqrt(out_real[k]*out_real[k]+out_imag[k]*out_imag[k]);
printf("magnitude =%f\n",m[k]);
p[k]=atan(out_imag[k]/out_real[k]);
printf("phase value =%f\n",p[k]);
}
}

Console Window

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 Digital Signal Processing Lab Manual (KEC 553)
enter length 4
enter the first sequence
1
1
0
0
x[0]= 2.000000 +j* 0.000000
magnitude =2.000000
phase value =0.000000
x[1]= 0.999996 +j* -1.000000
magnitude =1.414211
phase value =-0.785400
x[2]= 0.000000 +j* 0.000007
magnitude =0.000007
phase value =1.570793
x[3]= 1.000011 +j* 1.000000
magnitude =1.414221
phase value =0.785393
Output Graph

Magnitude

Phase

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 Digital Signal Processing Lab Manual (KEC 553)

3.2 IDFT

Program

#include<stdio.h>
#include<math.h>
int x[32],N,k,n,i;
float
pi=3.1416,sumre=0,sumim=0,out_real[8]={0.0},out_imag[8]={0.0},m[20]={0.0},p[20]={0.0};
main()
{
printf ("enter lenght\n");
scanf ("%d",&N);
printf("enter the first sequence\n");
for(i=0;i<N;i++)
scanf ("%d",&x[i]);
for(k=0;k<N;k++)
{
sumre=0;
sumim=0;
for(n=0;n<N;n++)
{
sumre=sumre+x[n]*cos(2*pi*k*n/N);
sumim=sumim-x[n]*sin(2*pi*k*n/N);
}
out_real[k]=sumre/N;
out_imag[k]=sumim/N;
printf("x[%d]=\t%f\t+j*\t%f\n",k,out_real[k],out_imag[k]);
m[k]=sqrt(out_real[k]*out_real[k]+out_imag[k]*out_imag[k]);
printf("magnitude =%f\n",m[k]);
p[k]=atan(out_imag[k]/out_real[k]);
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 Digital Signal Processing Lab Manual (KEC 553)
printf("phase value =%f\n",p[k]);
}
}

Console Window
enter length 4
enter the first sequence
1
0
1
0
x[0]= 0.500000 +j* 0.000000
magnitude =0.500000
phase value =0.000000
x[1]= 0.000000 +j* 0.000002
magnitude =0.000002
phase value =1.570793
x[2]= 0.500000 +j* -0.000004
magnitude =0.500000
phase value =-0.000007
x[3]= 0.000000 +j* 0.000005
magnitude =0.000005
phase value =1.570785
Output Graph

Magnitude

Phase

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 Digital Signal Processing Lab Manual (KEC 553)

RESULTS: DFT And IDFT Of The Discret Time Sequence has been verified through CCS.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-4. N-POINT FFT ALGORITHM

AIM: To verify N-point FFT algorithm discrete Time sequence.
EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply
THEORY:
The Fast Fourier Transform is useful to map the time-domain sequence into
a continuous function of a frequency variable. The FFT of a sequence
{x(n)} of length N is given by a complex-valued sequence X(k).
M nk
 j 2
X (k)   x(n) e n
;0  k  N 1
k 0

The above equation is the mathematical representation of the DFT. As the number of
computations involved in transforming a N point time domain signal into its corresponding
frequency domain signal was found to be N2 complex multiplications, an alternative
algorithm involving lesser number of computations is opted.
When the sequence x(n) is divided into 2 sequences and the DFT performed separately, the
resulting number of computations would be N2/2
(i.e.)
N2 N2
21 21
x(k)   x(2n)
n0
WN
2nk
  x(2n 1) WN
n0
(2n1)k
(6)

Consider x(2n) be the even sample sequences and x(2n+1) be the odd sample sequence
derived form x(n).
N2
21

 x(2n)
n0
WN2nk would result in (7)

N2
21

(N/2)2multiplication‟s  x(2n  1) WN(2n1)k (8)

n0

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 Digital Signal Processing Lab Manual (KEC 553)
an other (N/2)2 multiplication's finally resulting in (N/2)2 + (N/2)2
N2 N2 N2
=   Computations
4 4 2

Further solving Eg. (2)

N2 N
21 21 k
x(k)   x(2n) WN2nk   x(2n 1) WN(2nk ) W (9)
N
n0 n0

N N
21 k 21

  x(2n) WN WN  x(2n 1) WN (10)

2nk (2nk )

n0 n0

Dividing the sequence x(2n) into further 2 odd and even sequences would reduce the
computations.
WN  is the twiddle factor
 j 2

e n

  j 2 nk
W Nnk
 e n 

 N  N
K  K 
W  2  WN W  2
(11)
N N

 j 2  j 2 n
k
e n
e n 2

 j 2
k
 WN e k n

 W Nk (cos  j sin  )

 K  N 
 WN  W Nk (1)

2

 K  N 
W N

2
 WNk (12)

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 Digital Signal Processing Lab Manual (KEC 553)
Employing this equation, we deduce
N2 N
21 21
x(k)   x(2n)
n0
W N2nk   x(2n  1) W N(2nk ) (13)
n0

N
N
N 21 K
x(k  )   x(2n) WN W  x(2n 1) 21 WN(2nk ) (14)
2nk

2 n0 N

The time burden created by this large number of computations limits the usefulness of
DFT in many applications. Tremendous efforts devoted to develop more efficient ways of
computing DFT resulted in the above explained Fast Fourier Transform algorithm. This
mathematical shortcut reduces the number of calculations the DFT requires drastically.
The above mentioned radix-2 decimation in time FFT is employed for domain
transformation.
Dividing the DFT into smaller DFTs is the basis of the FFT. A radix-2 FFT divides the
DFT into two smaller DFTs, each of which is divided into smaller DFTs and so on,
resulting in a combination of two-point DFTs. The Decimation -In-Time (DIT) FFT
divides the input (time) sequence into two groups, one of even samples and the other of
odd samples. N/2 point DFT are performed on the these sub-sequences and their outputs
are combined to form the N point DFT.

FIG. 3A.1
The above shown mathematical representation forms the basis of N point FFT and is
called the Butterfly Structure.

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 Digital Signal Processing Lab Manual (KEC 553)

STAGE – I STAGE - II

STAGE – III
8 POINT DIT
4.1 PROGRAM:
%fast fourier transform
% Matlab code for N point Fast Fourier Transform
close all; clear all;
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 Digital Signal Processing Lab Manual (KEC 553)
x=input('enter the input sequence');
N=length(x);
%Code for FFT
disp('Fast Fourier Transformed Signal');
xk=fft(x,N); disp(xk);
%code for IFFT
disp('Inverse Fast Fourier Transformed Signal');
ixk=ifft(xk,N); disp(ixk);
%code to plot all sequences
%input sequence
n=0:N-1; subplot(2,2,1);
stem(n,x); xlabel('Time Index'); ylabel('Amplitude'); title('input sequence');
%FFT response sequence
k=0:N-1; subplot(2,2,2); stem(k,abs(xk)); xlabel('Frequency Index'); ylabel('Magnitude');
title('FFT response sequence');
%IFFT response sequence
n=0:N-1; ixk=ixk/N; subplot(2,2,3); stem(n,abs(ixk)); xlabel('Time Index'); ylabel('Magnitude');
title('IFFT response sequence');
Graph:

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 Digital Signal Processing Lab Manual (KEC 553)
4.2 C Program to Implement FFT
#include <math.h>
#define PTS 128 //# of points for FFT
#define PI 3.14159265358979
typedef struct {float real,imag;} COMPLEX;
void FFT(COMPLEX *Y, int n); //FFT prototype
float iobuffer[PTS]; //as input and output buffer
float x1[PTS],x[PTS]; //intermediate buffer
short i; //general purpose index variable
short buffercount = 0; //number of new samples in iobuffer
short flag = 0; //set to 1 by ISR when iobuffer full
float y[128];
COMPLEX w[PTS]; //twiddle constants stored in w
COMPLEX samples[PTS]; //primary working buffer
main()
{
float j,sum=0.0 ;
int n,k,i,a;
for (i = 0 ; i<PTS ; i++) // set up twiddle constants in w
{
w[i].real = cos(2*PI*i/(PTS*2.0)); /*Re component of twiddle constants*/
w[i].imag =-sin(2*PI*i/(PTS*2.0)); /*Im component of twiddle constants*/
}
/****************Input Signal X(n) *************************/
for(i=0,j=0;i<PTS;i++)
{
x[i] = sin(2*PI*5*i/PTS); // Signal x(Fs)=sin(2*pi*f*i/Fs);
samples[i].real=0.0;
samples[i].imag=0.0;
}
/********************Auto Correlation of X(n)=R(t) ***********/
for(n=0;n<PTS;n++)
{
sum=0;

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 Digital Signal Processing Lab Manual (KEC 553)
for(k=0;k<PTS-n;k++)
{
sum=sum+(x[k]*x[n+k]); // Auto Correlation R(t)
}
iobuffer[n] = sum;
}
/********************** FFT of R(t) ***********************/
for (i = 0 ; i < PTS ; i++) //swap buffers
{
samples[i].real=iobuffer[i]; //buffer with new data
}
for (i = 0 ; i < PTS ; i++)
samples[i].imag = 0.0; //imag components = 0
FFT(samples,PTS); //call function FFT.c
/******************** PSD ********************/
for (i = 0 ; i < PTS ; i++) //compute magnitude
{
x1[i] = sqrt(samples[i].real*samples[i].real
+ samples[i].imag*samples[i].imag);
}
} //end of main
FFT.c:
#define PTS 128 //# of points for FFT
typedef struct {float real,imag;} COMPLEX;
extern COMPLEX w[PTS]; //twiddle constants stored in w
void FFT(COMPLEX *Y, int N) //input sample array, # of points
{
COMPLEX temp1,temp2; //temporary storage variables
int i,j,k; //loop counter variables
int upper_leg, lower_leg; //indexof upper/lower butterfly leg
int leg_diff; //difference between upper/lower leg
int num_stages = 0; //number of FFT stages (iterations)
int index, step; //index/step through twiddle constant
i = 1; //log(base2) of N points= # of stages

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 Digital Signal Processing Lab Manual (KEC 553)
do
{
num_stages +=1;
i = i*2;
}while (i!=N);
leg_diff = N/2; //difference between upper&lower legs
step = (PTS*2)/N; //step between values in twiddle.h// 512
for (i = 0;i < num_stages; i++) //for N-point FFT
{
index = 0;
for (j = 0; j < leg_diff; j++)
{
for (upper_leg = j; upper_leg < N; upper_leg += (2*leg_diff))
{
lower_leg = upper_leg+leg_diff;
temp1.real = (Y[upper_leg]).real + (Y[lower_leg]).real;
temp1.imag = (Y[upper_leg]).imag + (Y[lower_leg]).imag;
temp2.real = (Y[upper_leg]).real - (Y[lower_leg]).real;
temp2.imag = (Y[upper_leg]).imag - (Y[lower_leg]).imag;
(Y[lower_leg]).real = temp2.real*(w[index]).real
-temp2.imag*(w[index]).imag;
(Y[lower_leg]).imag = temp2.real*(w[index]).imag
+temp2.imag*(w[index]).real;
(Y[upper_leg]).real = temp1.real;
(Y[upper_leg]).imag = temp1.imag;
}
index += step;
}
leg_diff = leg_diff/2;
step *= 2;
}
j = 0;
for (i = 1; i < (N-1); i++) //bit reversal for resequencing data
{

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 Digital Signal Processing Lab Manual (KEC 553)
k = N/2;
while (k <= j)
{
j = j - k;
k = k/2;
}
j = j + k;
if (i<j)
{
temp1.real = (Y[j]).real;
temp1.imag = (Y[j]).imag;
(Y[j]).real = (Y[i]).real;
(Y[j]).imag = (Y[i]).imag;
(Y[i]).real = temp1.real;
(Y[i]).imag = temp1.imag;
}
}
return;
}

RESULT: FFT of a Discrete Time Sequence has been verified through MATLAB.

Department of ECE, Dronacharya Group of Institutions, Greater Noida. 25 | P a g e

 Digital Signal Processing Lab Manual (KEC 553)

Experiment-5. FIR FILTER DESIGN

AIM: To design FIR Filter (LP/HP/BP/BR) Using Windowing technique

a. Rectangular window
b. Triangular window
c. Kaiser window

EQUIPMENTS:

Operating System – Windows XP

Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply

THEORY:
In this method, the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

In general, unit sample response hd(n) obtained from the above relation is infinite in duration, so it
must be truncated at some point say n= M-1 to yield an FIR filter of length M (i.e. 0 to M-1). This
truncation of hd(n) to length M-1 is same as multiplying hd(n) by the rectangular window defined as

Thus the unit sample response of the FIR filter becomes

Now, the multiplication of the window function w(n) with hd(n) is equivalent to convolution of Hd(w)
with W(w), where W(w) is the frequency domain representation of the window function

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 Digital Signal Processing Lab Manual (KEC 553)

Thus the convolution of Hd(w) with W(w) yields the frequency response of the truncated FIR filter

The frequency response can also be obtained using the following relation

But direct truncation of hd(n) to M terms to obtain h(n) leads to the Gibbs phenomenon effect which
manifests itself as a fixed percentage overshoot and ripple before and after an approximated
discontinuity in the frequency response due to the non-uniform convergence of the fourier series at a
discontinuity. In order to reduce the ripples, instead of multiplying hd(n) with a rectangular window
w(n), hd(n) is multiplied with a window function that contains a taper and decays toward zero
gradually, instead of abruptly as it occurs in a rectangular window. As multiplication of sequences
hd(n) and w(n) in time domain is equivalent to convolution of Hd(w) and W(w) in the frequency
domain, it has the effect of smoothing Hd(w).
The several effects of windowing the Fourier coefficients of the filter on the result of the frequency
response of the filter are as follows:
(i) A major effect is that discontinuities in H(w) become transition bands between values on either
side of the discontinuity.
(ii) The width of the transition bands depends on the width of the main lobe of the frequency response
of the window function, w(n) i.e. W(w).
(iii) Since the filter frequency response is obtained via a convolution relation , it is clear that the
resulting filters are never optimal in any sense.
(iv) As M (the length of the window function) increases, the mainlobe width of W(w) is reduced
which reduces the width of the transition band, but this also introduces more ripple in the frequency
response.
(v) The window function eliminates the ringing effects at the bandedge and does result in lower
sidelobes at the expense of an increase in the width of the transition band of the filter.
Some of the windows commonly used are as follows:

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 Digital Signal Processing Lab Manual (KEC 553)

The Bartlett window reduces the overshoot in the designed filter but spreads the transition region
considerably. The Hanning, Hamming and Blackman windows use progressively more complicated
cosine functions to provide a smooth truncation of the ideal impulse response and a frequency
response that looks better. The best window results probably come from using the Kaiser window,
which has a parameter ß that allows adjustment of the compromise between the overshoot reduction
and transition region width spreading.
The major advantages of using window method is their relative simplicity as compared to other
methods and ease of use. The fact that well defined equations are often available for calculating the
window coefficients has made this method successful.

There are following problems in filter design using window method:

(i) This method is applicable only if Hd(w) is absolutely integrable . When Hd(w) is complicated or
cannot easily be put into a closed form mathematical expression, evaluation of hd(n) becomes
difficult.
(ii) The use of windows offers very little design flexibility e.g. in low pass filter design, the passband
edge frequency generally cannot be specified exactly since the window smears the discontinuity in
frequency. Thus the ideal LPF with cut-off frequency fc, is smeared by the window to give a
frequency response with passband response with passband cutoff frequency f1 and stopband cut-off
frequency f2.
(iii) Window method is basically useful for design of prototype filters like lowpass,highpass,bandpass
etc. This makes its use in speech and image processing applications very limited.

5.1 FIR Filters Using Triangular Window

Program:
clc;
clear all;
rp = input('Enter the passband ripple = ');
rs = 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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 Digital Signal Processing Lab Manual (KEC 553)

wp = 2*fp/f;
ws = 2*fs/f;
num = -20*log10(sqrt(rp*rs))-13;
dem = 14.6*(fs-fp)/f;
n = ceil(num/dem);
n1 = n+1;
if (rem(n,2)==50)
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);
title('Magnitude Response of LPF');
ylabel('Gain in dB---- >');
xlabel('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);
title('Magnitude Response of HPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band pass filter
wn = [wp ws];
b = fir1(n,wn,y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('Magnitude Response of BPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band stop filter
b = fir1(n,wn,'stop',y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);

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 Digital Signal Processing Lab Manual (KEC 553)

title('Magnitude Response of BSF');

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

Output:
Enter the passband ripple = 0.04
Enter the stopband ripple = 0.02
Enter the passband frequency = 1500
Enter the stopband frequency = 2000
Enter the sampling frequency = 8000
Waveform:

5.2 FIR Filters Using Rectangular Window

Program:
clc;
clear all;
rp = input('Enter the passband ripple = ');
rs = 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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 Digital Signal Processing Lab Manual (KEC 553)

wp = 2*fp/f;
ws = 2*fs/f;
num = -20*log10(sqrt(rp*rs))-13;
dem = 14.6*(fs-fp)/f;
n = ceil(num/dem);
n1 = n+1;
if (rem(n,2)==50)
n1 = n;
n = n-1;
end
y = boxcar(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);
title('Magnitude Response of LPF');
ylabel('Gain in dB---- >');
xlabel('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);
title('Magnitude Response of HPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band pass filter
wn = [wp ws];
b = fir1(n,wn,y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('Magnitude Response of BPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band stop filter
b = fir1(n,wn,'stop',y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);

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 Digital Signal Processing Lab Manual (KEC 553)

title('Magnitude Response of BSF');

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

Output:
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

Waveform:

6.3 FIR Filters Using Kaiser Window

Program:
clc;
clear all;
rp = input('Enter the passband ripple = ');
rs = 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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 Digital Signal Processing Lab Manual (KEC 553)

beta = input('Enter the beta value = ');

wp = 2*fp/f;
ws = 2*fs/f;
num = -20*log10(sqrt(rp*rs))-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 = 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);
title('Magnitude Response of LPF');
ylabel('Gain in dB---- >');
xlabel('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);
title('Magnitude Response of HPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band pass filter
wn = [wp ws];
b = fir1(n,wn,y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('Magnitude Response of BPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band stop filter
b = fir1(n,wn,'stop',y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,4);

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 Digital Signal Processing Lab Manual (KEC 553)

plot(o/pi,m);
title('Magnitude Response of BSF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
Output:
Enter the passband ripple = 0.02
Enter the stopband ripple = 0.01
Enter the passband frequency = 1000
Enter the stopband frequency = 1500
Enter the sampling frequency = 10000
Enter the beta value = 5.8 Waveform:

RESULT: FIR Filter (LP/HP/BP/BR) Using Windowing technique has been designed
i) Rectangular window ii) Triangular window iii) Kaiser window and their magnitude
responses are visualized.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-6. FIR FILTER DESIGN

AIM: To design FIR Filter (LP/HP/BP/BR) Using Windowing technique
a) Hamming window
b) Hanning window
c) Blackmann window
EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply

THEORY:

A Finite Impulse Response (FIR) filter is a discrete linear time-invariant system whose output
is based on the weighted summation of a finite number of past inputs. An FIR transversal filter
structure can be obtained directly from the equation for discrete-time convolution.
N 1

y(n)   x(k)h(n  k) 0  n  N  1
k0
In this equation, x(k) and y(n) represent the input to and output from the filter at time n. h(n-k) is the
transversal filter coefficients at time n. These coefficients are generated by using FDS (Filter Design
Software or Digital filter design package).

FIR – filter is a finite impulse response filter. Order of the filter should be specified. Infinite response
is truncated to get finite impulse response. placing a window of finite length does this. Types of
windows available are Rectangular, Barlett, Hamming, Hanning, Blackmann window etc. This FIR
filter is an all zero filter.
In this method, the desired frequency response specification Hd(w), corresponding unit sample
response hd(n) is determined using the following relation

In general, unit sample response hd(n) obtained from the above relation is infinite in duration, so it
must be truncated at some point say n= M-1 to yield an FIR filter of length M (i.e. 0 to M-1). This
truncation of hd(n) to length M-1 is same as multiplying hd(n) by the rectangular window defined as

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 Digital Signal Processing Lab Manual (KEC 553)

Thus the unit sample response of the FIR filter becomes

Now, the multiplication of the window function w(n) with hd(n) is equivalent to convolution of Hd(w)
with W(w), where W(w) is the frequency domain representation of the window function

Thus the convolution of Hd(w) with W(w) yields the frequency response of the truncated FIR filter

The frequency response can also be obtained using the following relation

But direct truncation of hd(n) to M terms to obtain h(n) leads to the Gibbs phenomenon effect which
manifests itself as a fixed percentage overshoot and ripple before and after an approximated
discontinuity in the frequency response due to the non-uniform convergence of the fourier series at a
discontinuity. In order to reduce the ripples, instead of multiplying hd(n) with a rectangular window
w(n), hd(n) is multiplied with a window function that contains a taper and decays toward zero
gradually, instead of abruptly as it occurs in a rectangular window. As multiplication of sequences
hd(n) and w(n) in time domain is equivalent to convolution of Hd(w) and W(w) in the frequency
domain, it has the effect of smoothing Hd(w).
The several effects of windowing the Fourier coefficients of the filter on the result of the frequency
response of the filter are as follows:
(i) A major effect is that discontinuities in H(w) become transition bands between values on either
side of the discontinuity.
(ii) The width of the transition bands depends on the width of the main lobe of the frequency response
of the window function, w(n) i.e. W(w).

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 Digital Signal Processing Lab Manual (KEC 553)
(iii) Since the filter frequency response is obtained via a convolution relation , it is clear that the
resulting filters are never optimal in any sense.
(iv) As M (the length of the window function) increases, the mainlobe width of W(w) is reduced
which reduces the width of the transition band, but this also introduces more ripple in the frequency
response.
(v) The window function eliminates the ringing effects at the bandedge and does result in lower
sidelobes at the expense of an increase in the width of the transition band of the filter.
Some of the windows commonly used are as follows:

The Bartlett window reduces the overshoot in the designed filter but spreads the transition region
considerably. The Hanning, Hamming and Blackman windows use progressively more complicated
cosine functions to provide a smooth truncation of the ideal impulse response and a frequency
response that looks better. The best window results probably come from using the Kaiser window,
which has a parameter ß that allows adjustment of the compromise between the overshoot reduction
and transition region width spreading.
The major advantages of using window method is their relative simplicity as compared to other
methods and ease of use. The fact that well defined equations are often available for calculating the
window coefficients has made this method successful.

There are following problems in filter design using window method:

(i) This method is applicable only if Hd(w) is absolutely integrable . When Hd(w) is complicated or
cannot easily be put into a closed form mathematical expression, evaluation of hd(n) becomes
difficult.
(ii) The use of windows offers very little design flexibility e.g. in low pass filter design, the passband
edge frequency generally cannot be specified exactly since the window smears the discontinuity in
frequency. Thus the ideal LPF with cut-off frequency fc, is smeared by the window to give a
frequency response with passband response with passband cutoff frequency f1 and stopband cut-off
frequency f2.

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 Digital Signal Processing Lab Manual (KEC 553)
(iii) Window method is basically useful for design of prototype filters like lowpass,highpass,bandpass
etc. This makes its use in speech and image processing applications very limited.

6.1 FIR Filters Using Hamming Window

Program:

clc;
clear all;
rp = input('Enter the passband ripple = ');
rs = 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(rp*rs))-13;
dem = 14.6*(fs-fp)/f;
n = ceil(num/dem);
n1 = n+1;
if (rem(n,2)==50)
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);
title('Magnitude Response of LPF');
ylabel('Gain in dB---- >');
xlabel('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);
title('Magnitude Response of HPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band pass filter
wn = [wp ws];
b = fir1(n,wn,y);
[h,o] = freqz(b,1,256);

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 Digital Signal Processing Lab Manual (KEC 553)

m = 20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('Magnitude Response of BPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band stop filter
b = fir1(n,wn,'stop',y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);
title('Magnitude Response of BSF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;

Output:

Enter the passband ripple = 0.02

Enter the stopband ripple = 0.01
Enter the passband frequency = 1200
Enter the stopband frequency = 1700
Enter the sampling frequency = 9000

Waveform:

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 Digital Signal Processing Lab Manual (KEC 553)

6.2 FIR Filters Using Hanning Window

Program:
clc;
clear all;
rp = input('Enter the passband ripple = ');
rs = 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(rp*rs))-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

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 Digital Signal Processing Lab Manual (KEC 553)

b = fir1(n,wp,y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
title('Magnitude Response of LPF');
ylabel('Gain in dB---- >');
xlabel('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);
title('Magnitude Response of HPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band pass filter
wn = [wp ws];
b = fir1(n,wn,y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('Magnitude Response of BPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band stop filter
b = fir1(n,wn,'stop',y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);
title('Magnitude Response of BSF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;

Output:

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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 Digital Signal Processing Lab Manual (KEC 553)

Waveform:

6.3 FIR Filters Using Blackman Window

Program:
clc;
clear all;
rp = input('Enter the passband ripple = ');
rs = 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(rp*rs))-13;
dem = 14.6*(fs-fp)/f;
n = ceil(num/dem);
n1 = n+1;
if (rem(n,2)==50)
n1 = n;
n = n-1;

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 Digital Signal Processing Lab Manual (KEC 553)

end
y = blackman(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);
title('Magnitude Response of LPF');
ylabel('Gain in dB---- >');
xlabel('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);
title('Magnitude Response of HPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band pass filter
wn = [wp ws];
b = fir1(n,wn,y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('Magnitude Response of BPF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
% band stop filter
b = fir1(n,wn,'stop',y);
[h,o] = freqz(b,1,256);
m = 20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);
title('Magnitude Response of BSF');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;

Output:

Enter the passband ripple = 0.03

Enter the stopband ripple = 0.01

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 Digital Signal Processing Lab Manual (KEC 553)

Enter the passband frequency = 2000

Enter the stopband frequency = 2500
Enter the sampling frequency = 7000

Waveform:

RESULT: FIR Filter (LP/HP/BP/BR) Using Windowing technique has been designed
i) Blackmann window ii) Hamming window iii) Hanning window and their magnitude
responses are visualized.

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 Digital Signal Processing Lab Manual (KEC 553)

7. FREQUENCY RESPONSE OF ANALOG BUTTERWORTH PROTOTYPE

LP/HP/BP/BR FILTERS

AIM: To find the frequency response of analog LP/HP/BP/BR filters using matlab.
EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply

THEORY

BUTTERWORTH FILTER

The Butterworth filter has a maximally flat response that is, no passband ripple and roll-off of
minus 20db per pole. It is “flat maximally magnitude” filters at the frequency of jω= 0, as the
first 2N - 1 derivatives of the transfer function when jω = 0 are equal to zero [5]. The phase
response of the Butterworth filter becomes more nonlinear with increasing N. This filter is
completely defined mathematically by two parameters; they are cut off frequency and
number of poles. The magnitude squared response of low pass Butterworth filter is given by,

Filter Selectivity,

Attenuation,

The frequency response of the Butterworth filter is maximally flat in the passband and rolls
off towards zero in the stopband . When observed on a logarithmic bode plot the response
slopes off linearly towards negative infinity. A first-order filter's response rolls off at −6 dB
per octave (−20 dB per decade). A second-order filter‟s response rolls off at −12 dB per
octave and a third-order at −18 dB. Butterworth filters have a monotonically varying
magnitude function with ω, unlike other filter types that have non-monotonic ripple in the
passband and the stopband.

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 Digital Signal Processing Lab Manual (KEC 553)
Compared with a Chebyshev Type I filter or an Elliptic filter, the Butterworth filter has a
slower roll-off and therefore will require a higher order to implement a particular stopband
specification. Butterworth filters have a more linear phase response in the pass-band than
Chebyshev Type I and Elliptic filters .

The Butterworth filter rolls off more slowly around the cut off frequency than the Chebyshev
Type I and Elliptic filters without ripple. All of these filters are in fifth order. Fig.2:Phase
response of the various types of IIR filters

7.1 Butterworth Analog Low Pass Filter

Program

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n,wn] = buttord(w1,w2,rp,rs,'s');
[z,p,k] = butter(n,wn);
[b,a] = zp2tf(z,p,k);
[b,a] = butter(n,wn,'s');
w = 0:0.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Amplitude Response');
ylabel('Gain in dB---- >');
xlabel('Normalised frequency---- >');
grid on;
subplot(2,1,2);
plot(om/pi,an);
title('Phase Response');
xlabel('Normalised frequency---- >');
ylabel('Phase in radians---- >');
grid on;

Output:
Enter the passband ripple = 0.15
Enter the stopband ripple = 60
Enter the passband frequency = 1500
Enter the stopband frequency = 3000
Enter the sampling frequency = 7000

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 Digital Signal Processing Lab Manual (KEC 553)

7.2 Butterworth Analog High Pass Filter

Program:

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n,wn] = buttord(w1,w2,rp,rs,'s');
[b,a] = butter(n,wn,'high','s');
w = 0:0.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
ylabel('Gain in dB---- >');

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 Digital Signal Processing Lab Manual (KEC 553)
xlabel('Normalised frequency---- >');
title('Amplitude Response');
grid on;
subplot(2,1,2);
plot(om/pi,an);
xlabel('Normalised frequency---- >');
ylabel('Phase in radians---- >');
title('Phase Response');
grid on;

Output:

Enter the passband ripple = 0.2

Enter the stopband ripple = 40
Enter the passband frequency = 2000
Enter the stopband frequency = 3500
Enter the sampling frequency = 8000

Waveform:

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 Digital Signal Processing Lab Manual (KEC 553)

7.3 Butterworth Analog Band Pass Filter

Program:

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n] = buttord(w1,w2,rp,rs);
wn = [w1 w2];
[b,a] = butter(n,wn,'bandpass','s');
w = 0:.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
title('Magnitude Response');
grid on;
subplot(2,1,2);
plot(om/pi,an);
xlabel('Normalised Frequency---- >');
ylabel('Phase in Radians---- >');
title('Phase Response');
grid on;

Output:

Enter the passband ripple = 0.36

Enter the stopband ripple = 36
Enter the passband frequency = 1500
Enter the stopband frequency = 2000
Enter the sampling frequency = 6000

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 Digital Signal Processing Lab Manual (KEC 553)

Waveform:

7.4 Butterworth Analog Band Stop Filter

Program:

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n] = buttord(w1,w2,rp,rs,'s');
wn = [w1 w2];
[b,a] = butter(n,wn,'stop','s');
w = 0:0.01:pi;

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 Digital Signal Processing Lab Manual (KEC 553)
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Response');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
subplot(2,1,2);
plot(om/pi,an);
title('Phase Response');
xlabel('Normalised Frequency---- >');
ylabel('Phase in radians---- >');
grid on;

Output:

Enter the passband ripple = 0.28

Enter the stopband ripple = 28
Enter the passband frequency = 1000
Enter the stopband frequency = 1400
Enter the sampling frequency = 5000

Waveform:

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 Digital Signal Processing Lab Manual (KEC 553)

RESULTS: Hence Butterworth analog (LP/HP/BP/BR) proto type filters are implemented
and their magnitude responses are visualized.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-8. FREQUENCY RESPONSE OF ANALOG CHEBYSHEV

LP/HP/BP/BR FILTERS
AIM: To find the frequency response of analog LP/HP filters using matlab.
EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply
THEORY
CHEBYSHEV TYPE -I FILTER The absolute difference between the ideal and actual
frequency response over the entire passband is minimized by Chebyshev Type I filter by
incorporating equal ripple in the passband. Stopband response is maximally flat. The
transition from passband to stopband is more rapid than for the Butterworth filter. The
magnitude squared Chebyshev type I response is:

Where

The magnitude squared response peaks occur in the pass band when

The ripple is often given in dB:

9.1 Chebyshev Type 1 Analog Low Pass Filter

Program:
clc;
clear all;
rp = input('Enter the passband ripple = ');
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 Digital Signal Processing Lab Manual (KEC 553)
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,'s');
[b,a] = cheby1(n,rp,wn,'s');
w = 0:0.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Response');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
subplot(2,1,2);
plot(om/pi,an);
title('Phase Response');
xlabel('Normalised Frequency---- >');
ylabel('Phase in radians---- >');
grid on;

Output:
Enter the passband ripple = 0.23
Enter the stopband ripple = 47
Enter the passband frequency = 1300
Enter the stopband frequency = 1550
Enter the sampling frequency = 7800

Waveform:

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 Digital Signal Processing Lab Manual (KEC 553)

8.2 Chebyshev Type 1 Analog High Pass Filter

Program:

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n,wn] = cheb1ord(w1,w2,rp,rs,'s');
[b,a] = cheby1(n,rp,wn,'high','s');
w = 0:0.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Response');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
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 Digital Signal Processing Lab Manual (KEC 553)
subplot(2,1,2);
plot(om/pi,an);
title('Phase Response');
xlabel('Normalised Frequency---- >');
ylabel('Phase in radians---- >');
grid on;

Output:

Enter the passband ripple = 0.29

Enter the stopband ripple = 29
Enter the passband frequency = 900
Enter the stopband frequency = 1300
Enter the sampling frequency = 7500

Waveform:

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 Digital Signal Processing Lab Manual (KEC 553)

8.3 Chebyshev Type 1 Analog Band Pass Filter

Program:

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n] = cheb1ord(w1,w2,rp,rs,'s');
wn = [w1 w2];
[b,a] = cheby1(n,rp,wn,'bandpass','s');
w = 0:0.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Response');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
subplot(2,1,2);
plot(om/pi,an);
title('Phase Response');
xlabel('Normalised Frequency---- >');
ylabel('Phase in radians---- >');
grid on;

Output:

Enter the passband ripple = 0.3

Enter the stopband ripple = 40
Enter the passband frequency = 1400
Enter the stopband frequency = 2000
Enter the sampling frequency = 5000

Waveform:

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 Digital Signal Processing Lab Manual (KEC 553)

8.4 Chebyshev Type 1 Analog Band Stop Filter

Program:

clc;
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 = ');
fs = input('Enter the sampling frequency = ');
w1 = 2*wp/fs;
w2 = 2*ws/fs;
[n] = cheb1ord(w1,w2,rp,rs,'s');
wn = [w1 w2];
[b,a] = cheby1(n,rp,wn,'stop','s');
w = 0:0.01:pi;
[h,om] = freqs(b,a,w);
m = 20*log10(abs(h));
an = angle(h);
subplot(2,1,1);
plot(om/pi,m);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Response');
ylabel('Gain in dB---- >');
xlabel('Normalised Frequency---- >');
grid on;
subplot(2,1,2);
plot(om/pi,an);
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 Digital Signal Processing Lab Manual (KEC 553)
title('Phase Response');
xlabel('Normalised Frequency---- >');
ylabel('Phase in radians---- >');
grid on;

Output:

Enter the passband ripple = 0.15

Enter the stopband ripple = 30
Enter the passband frequency = 2000
Enter the stopband frequency = 2400
Enter the sampling frequency = 7000

Waveform:

RESULTS: Hence Butterworth and chebyshev analog (LP/HP/BP/BR) proto type filters are
implemented and their magnitude responses are visualized.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-9. IIR FILTER DESIGN

AIM: To design and implement IIR (LPF/HPF) filters on DSP processors using CCS and
also to implement in MATLAB.

EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply

THEORY:

The IIR filter can realize both the poles and zeroes of a system because it has a rational
transfer function, described by polynomials in z in both the numerator and the denominator:

b
M
k z k

H (z) k 0
N (2)
 a Z k k
k 1

The difference equation for such a system is described by the following:

M N

y(n)   bk x(n  k)  a k y(n  k) (3)

k 0 k 1

M and N are order of the two polynomials

bk and ak are the filter coefficients. These filter coefficients are generated using FDS (Filter
Design software or Digital Filter design package).

IIR filters can be expanded as infinite impulse response filters. In designing IIR filters, cutoff
frequencies of the filters should be mentioned. The order of the filter can be estimated using
butter worth polynomial. That‟s why the filters are named as butter worth filters. Filter
coefficients can be found and the response can be plotted.

9.1 PROGRAM :

% IIR filters LPF & HPF

clc;
clear all;

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 Digital Signal Processing Lab Manual (KEC 553)
close all;
disp('enter the IIR filter design specifications');
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband freq');
ws=input('enter the stopband freq');
fs=input('enter the sampling freq');
w1=2*wp/fs;w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs,'s');
c=input('enter choice of filter 1. LPF 2. HPF \n ');
if(c==1)
disp('Frequency response of IIR LPF is:');
[b,a]=butter(n,wn,'low','s');
end
if(c==2)
disp('Frequency response of IIR HPF is:');
[b,a]=butter(n,wn,'high','s');
end
w=0:.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure,subplot(2,1,1);plot(om/pi,m);
title('magnitude response of IIR filter is:');
xlabel('(a) Normalized freq. -->');
ylabel('Gain in dB-->');
subplot(2,1,2);plot(om/pi,an);
title('phase response of IIR filter is:');
xlabel('(b) Normalized freq. -->');
ylabel('Phase in radians-->');

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 Digital Signal Processing Lab Manual (KEC 553)

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 Digital Signal Processing Lab Manual (KEC 553)

9.2 Program
clc;
clear;
ap=input('Enter PB attn in dB : '); as=input('Enter SB attn in dB : ');
fp=input('Enter PB freq in hz : '); fs=input('Enter SB freq in hz : ');
F=input('Enter the sampling freq : ');
wp=2*pi*fp/F;
ws=2*pi*fs/F;
wp=2*F*tan(wp/2);
ws=2*F*tan(ws/2);
[N,wc]=buttord(wp,ws,ap,as,'s')
c=input('enter choice of filter 1. LPF 2. HPF \n ');
if(c==1)
disp('Frequency response of IIR LPF is:');
[b,a]=butter(n,wn,'low','s');
end
if(c==2)
disp('Frequency response of IIR HPF is:');
[b,a]=butter(n,wn,'high','s');
end
%[b,a]=butter(N,wc,'s')
[bz,az]=bilinear(b,a,F)
freqz(bz,az)
command window
Enter PB attn in dB : 0.15
Enter SB attn in dB : 60
Enter PB freq in hz : 1500
Enter SB freq in hz : 3000
Enter the sampling freq : 7000
enter choice of filter 1. LPF 2. HPF : 1

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 Digital Signal Processing Lab Manual (KEC 553)

Lowpass Filter Response

100

Magnitude (dB)
-100

-200

-300
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

0
Phase (degrees)

-200

-400

-600
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized Frequency ( rad/sample)

Enter PB attn in dB : 0.15

Enter SB attn in dB : 60
Enter PB freq in hz : 1500
Enter SB freq in hz : 3000
Enter the sampling freq : 7000
enter choice of filter 1. LPF 2. HPF : 1

Highpass Filter Response

9.3 IIR filter (LP/HP) Implementation on DSP Processors

#include<stdio.h>
#include<math.h>
float h[100];
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 Digital Signal Processing Lab Manual (KEC 553)
int n,w,c,i,wc;
float mul(float,int);
void main()
{
printf("enter order of the filter");
scanf("%d",&n);
printf("enter wc");
scanf("%d",&wc);
printf("enter the choice 1.lpf 2.hpf");
scanf("%d",&c);
switch(c)
{
case 1:
for(w=0;w<100;w++)
{
h[w]=1/sqrt(1+mul((w/(float)wc),2*n));
printf("h[%d]=%f\n",w,h[w]);
}
break;
case 2:
for(w=0;w<100;w++)
{
h[w]=1/sqrt(1+mul(((float)wc/w),2*n));
printf("h[%d]=%f\n",w,h[w]);
}
break;
default:
printf("enter correct choice");
}
}
float mul(float a,int x)
{
for(i=0;i<x-1;i++)
{
a*=a;
}
return(a);
}

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 Digital Signal Processing Lab Manual (KEC 553)
Output

Hpf: h[13]=0.004570

enter order of the filter 2 h[14]=0.006146

enter wc 50 h[15]=0.008100

enter the choice 1.lpf 2.hpf2 h[16]=0.010485

h[0]=0.000000 h[17]=0.013362

h[1]=0.000000 h[18]=0.016794

h[2]=0.000003 h[19]=0.020847

h[3]=0.000013 h[20]=0.025592

h[4]=0.000041 h[21]=0.031102

h[5]=0.000100 h[22]=0.037455

h[6]=0.000207 h[23]=0.044730

h[7]=0.000384 h[24]=0.053010

h[8]=0.000655 h[25]=0.062378

h[9]=0.001050 h[26]=0.072922

h[10]=0.001600 h[27]=0.084725

h[11]=0.002343

h[12]=0.003318

Lpf: h[12]=0.999995

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 Digital Signal Processing Lab Manual (KEC 553)
enter order of the filter2 h[13]=0.999990

enter wc50 h[14]=0.999981

enter the choice 1.lpf 2.hpf1 h[15]=0.999967

h[0]=1.000000 h[16]=0.999945

h[1]=1.000000 h[17]=0.999911

h[2]=1.000000 h[18]=0.999859

h[3]=1.000000 h[19]=0.999783

h[4]=1.000000 h[20]=0.999672

h[5]=1.000000 h[21]=0.999516

h[6]=1.000000 h[22]=0.999298

h[7]=1.000000 h[23]=0.998999

h[8]=1.000000 h[24]=0.998594

h[9]=0.999999 h[25]=0.998053

h[10]=0.999999 h[26]=0.997338

h[11]=0.999997

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 Digital Signal Processing Lab Manual (KEC 553)

RESULT: IIR Filter (LP/HP) has been implemented and their magnitude responses are
visualized through DSP processor and MATLAB.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-10. IIR FILTER DESIGN

AIM: To design and implement IIR (BP/BR) filters on DSP processors using CCS and also
to implement in MATLAB.

EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply

THEORY:

The IIR filter can realize both the poles and zeroes of a system because it has a rational
transfer function, described by polynomials in z in both the numerator and the denominator:

b
M
k z k

H (z) k 0
N (2)
 a Z k k
k 1

The difference equation for such a system is described by the following:

M N

y(n)   bk x(n  k)  a k y(n  k) (3)

k 0 k 1

M and N are order of the two polynomials

bk and ak are the filter coefficients. These filter coefficients are generated using FDS (Filter
Design software or Digital Filter design package).

IIR filters can be expanded as infinite impulse response filters. In designing IIR filters, cutoff
frequencies of the filters should be mentioned. The order of the filter can be estimated using
butter worth polynomial. That‟s why the filters are named as butter worth filters. Filter
coefficients can be found and the response can be plotted.

10.1 IIR filter (BP/BR) Implementation on DSP Processors

Program
#include<stdio.h>
#include<math.h>
float h[100];
int n,w,c,i,wc,wl,wu,w1,w2;
Department of ECE, Dronacharya Group of Institutions, Greater Noida. 69 | P a g e
 Digital Signal Processing Lab Manual (KEC 553)
float mul(float,int);
void main()
{
printf("enter order of the filter");
scanf("%d",&n);
printf("enter wl");
scanf("%d%d",&wl,&wu);
printf("enter the choice 1.bpf 2.brf");
scanf("%d",&c);
w1=wu-wl;
w2=wu*wl;
switch(c)
{
case 1:
for(w=0;w<100;w++)
{
h[w]=1/sqrt(1+mul(((w*w)-w2)/(float)(w1),2*n));
printf("h[%d]=%f\n",w,h[w]);
}
break;
case 2:
for(w=0;w<100;w++)
{
h[w]=1/sqrt(1+mul((w*w1)/(float)((w*w)-
w2),2*n));
printf("h[%d]=%f\n",w,h[w]);
}
break;
default:
printf("enter correct choice");
}
}
float mul(float a,int x)
{
for(i=0;i<x-1;i++)
{
a*=a;
}
return(a);
}

Output

Bpf

enter order of the filter2 h[25]=0.000017

enter wl,wu 20 h[26]=0.000023

70 h[27]=0.000031

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 Digital Signal Processing Lab Manual (KEC 553)

enter the choice 1.bpf 2.brf h[28]=0.000043

1 h[29]=0.000064

h[0]=0.000002 h[30]=0.000100

h[1]=0.000002 h[31]=0.000168

h[2]=0.000002 h[32]=0.000313

h[3]=0.000002 h[33]=0.000668

h[4]=0.000002 h[34]=0.001763

h[5]=0.000002 h[35]=0.006664

h[6]=0.000002 h[36]=0.053349

h[7]=0.000002 h[37]=0.989259

h[8]=0.000002 h[38]=0.857608

h[9]=0.000002 h[39]=0.029144

h[10]=0.000002 h[40]=0.003906

h[11]=0.000002 h[41]=0.001002

h[12]=0.000003 h[42]=0.000356

h[13]=0.000003 h[43]=0.000154

h[14]=0.000003 h[44]=0.000076

h[15]=0.000003 h[45]=0.000041

h[16]=0.000004 h[46]=0.000024

h[17]=0.000004 h[47]=0.000015

h[18]=0.000005 h[48]=0.000009

h[19]=0.000005 h[49]=0.000006

h[20]=0.000006 h[50]=0.000004

h[21]=0.000007 h[51]=0.000003

h[22]=0.000009 h[52]=0.000002

h[23]=0.000011

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 Digital Signal Processing Lab Manual (KEC 553)
h[24]=0.000014

Brf

enter order of the filter2 h[24]=0.217024

enter wl20 70 h[25]=0.146176

enter the choice 1.bpf 2.brf2 h[26]=0.095759

h[0]=1.000000 h[27]=0.060918

h[1]=1.000000 h[28]=0.037455

h[2]=1.000000 h[29]=0.022084

h[3]=1.000000 h[30]=0.012345

h[4]=1.000000 h[31]=0.006435

h[5]=0.999999 h[32]=0.003050

h[6]=0.999997 h[33]=0.001262

h[7]=0.999990 h[34]=0.000424

h[8]=0.999968 h[35]=0.000100

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 Digital Signal Processing Lab Manual (KEC 553)

h[9]=0.999908 h[36]=0.000011

h[10]=0.999761 h[37]=0.000000

h[11]=0.999416 h[38]=0.000000

h[12]=0.998647 h[39]=0.000015

h[13]=0.996992 h[40]=0.000100

h[14]=0.993536 h[41]=0.000353

h[15]=0.986501 h[42]=0.000903

h[16]=0.972576 h[43]=0.001902

h[17]=0.946013 h[44]=0.003523

h[18]=0.898180 h[45]=0.005954

h[19]=0.819645 h[46]=0.009391
h[20]=0.707107 h[47]=0.014044
h[21]=0.571175 h[48]=0.020125
h[22]=0.433355 h[49]=0.027855
h[23]=0.312576

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 Digital Signal Processing Lab Manual (KEC 553)

10.2 Program
clc;
clear;
ap=input('Enter PB attn in dB : ');
as=input('Enter SB attn in dB : ');
fp=input('Enter PB freq in hz : ');
fs=input('Enter SB freq in hz : ');
F=input('Enter the sampling freq : ');
wp=2*pi*fp/F;
ws=2*pi*fs/F;
wp=2*F*tan(wp/2);
ws=2*F*tan(ws/2);
[N,wc]=buttord(wp,ws,ap,as,'s')
c=input('enter choice of filter 1. BPF 2. BRF \n ');
if(c==1)
disp('Frequency response of IIR LPF is:');
[b,a]=butter(N,wc,'bandpass','s');
end
if(c==2)
disp('Frequency response of IIR HPF is:');
[b,a]=butter(N,wc,'stop','s');
end
%[b,a]=butter(N,wc,'s')
[bz,az]=bilinear(b,a,F)
freqz(bz,az)

Comand wind
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 Digital Signal Processing Lab Manual (KEC 553)

Enter PB attn in dB : 0.15

Enter SB attn in dB : 60
Enter PB freq in hz : [1500 4500]
Enter SB freq in hz : [2500 3000]
Enter the sampling freq : 10000
N= 5
wc = 1.0e+04 *[ 1.2856 4.2824]
enter choice of filter 1. BPF 2. BRF : 1

Enter PB attn in dB : 0.15

Enter SB attn in dB : 60
Enter PB freq in hz : [1500 4500]
Enter SB freq in hz : [2500 3000]
Enter the sampling freq : 10000
N= 5
wc = 1.0e+04 *[ 1.2856 4.2824]
enter choice of filter 1. BPF 2. BRF : 2

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 Digital Signal Processing Lab Manual (KEC 553)

RESULT: IIR Filter (BP/BR) has been implemented and their magnitude responses are
visualized through DSP processor and MATLAB.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-11. FIR FILTERS USING FREQUENCY SAMPLING

METHOD
AIM: To Design of FIR filters using frequency sampling method

EQUIPMENTS:
Operating System – Windows XP
Constructor - Simulator
Software - CCStudio 3 & MATLAB 7.5
Hardware - DSK6713 kit, USB probe,5V DC supply
THEORY:
FIR Filter Design by Frequency Sampling
The technique described below is simpler than some other approaches. Herein, it is assumed here
that the filter length, M, is odd.
The basic method is to specify the desired magnitude of the frequency response, |Hd(F)|, at a
set of frequencies Fk = k * dF = k /M. The unit sample response is then found via the inverse
DFT. The troublesome point with this method is properly setting up the desired magnitude
|Hd(F)| with even symmetry, and appropriate phase, <Hd(F), (with odd symmetry). These
requirements are needed to ensure the resulting filter is causal, has a constant delay (linear
phase shift), and has real coefficients.
If a zero phase shift was specified for each <Hd(F) then the resulting h(n) would be centered
at the origin. Hence a minimum delay of g = (M-1)/2 samples is needed to make the filter
causal. By the time-shift property of the DTFT, this corresponds to multiplication of |Hd(F)|
by exp(-j*g*2*pi*F). Note the linear phase variation given in the exponential. The derivative
of Hd with respect to w yields the group delay for the filter, which is the constant, g, (in
samples). Real coefficients are assured because of the uniform spacing of Fk and because
symmetric magnitudes are assigned to Hd(Fk) for positive and negative values of Fk.
Recall that with the method of FIR design by windowing the only inputs were the cutoff
frequency and the filter length. The resulting error |H(F)| - |Hd(F)| was not known at any
given frequency. In this technique frequency response will be exact at the specified
frequencies, F = Fk, but will be unknown at other frequencies.

General Procedure:

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 Digital Signal Processing Lab Manual (KEC 553)
1) Given |Hd(F)| for 0 <= F < 1.
2) Select the filter length = M.
3) Find the frequency spacing, dF, for the sampling of Hd(F): dF = 1 / M (cycles / sample)
Hd(F) will be specified for F = k * dF, with k = 0 , M-1, thus giving samples at frequencies
across the range of 0 <= F < 1.
4) Find the group delay for the filter = g = (M-1) / 2.
5) Setup symmetric |Hd(k)|. Setup <Hd(k) = -g * 2 * pi * dF * k.
6) Find h(n) = IDFT{ Hd(k) }. This is an M-point IDFT.
7) Find actual frequency response, H(k) = DFT{ h(n) }. Use a large value of N,
zero padding h(n) to yield an acceptable frequency resolution.
8) Assess filter by examining plot of magnitude of frequency response.
9) Repeat as needed, going to step #2.

Program

fs=8e3;
ts=1/fs; ns=512;
t=[0:ts:ts*(ns-1)];
f1=500; f2=1800; f3=2000; f4=3200;
x1=sin(2*pi*f1*t);
x2=sin(2*pi*f2*t);
x3=sin(2*pi*f3*t);
x4=sin(2*pi*f4*t);
x=x1+x2+x3+x4;
grid on;
n=16;
w=[0.4 0.6];
w1=0.4;
a=1;
c=input('enter 1.lpf 2.hpf 3.bpf 4.brf');
if(c==1)
b=fir1(n,w1,'low');
end;
if(c==2)
w=0.5;
b=fir1(n,w,'high');
end;
if(c==3)
w=[0.4 0.6];
b=fir1(n,w,'DC-0');
end;
if(c==4)
w=[0.4 0.6];

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 Digital Signal Processing Lab Manual (KEC 553)
b=fir1(n,w,'DC-1');
end;
freqz(b,a);
pause;figure;
subplot(2,1,1);
npts=200;
plot(t(1:npts),x(1:npts));
title('time plot of input and output');
xlabel('time');
ylabel('input signal');
y=filter(b,a,x);
subplot(2,1,2);
plot(t(1:npts),y(1:npts));
xlabel('time');
ylabel('filtered signal');
pause;figure;
subplot(2,1,1);
xfftmag=(abs(fft(x,ns)));
xfftmagh=xfftmag(1:length(xfftmag)/2);
f=(1:1:length(xfftmagh))*fs/ns;
plot(f,xfftmagh);
title('input and output spectrum');
xlabel('frequency');ylabel('input spectrum');
subplot(2,1,2);
yfftmag=(abs(fft(y,ns)));
yfftmagh=yfftmag(1:length(yfftmag)/2);
f=(1:1:length(yfftmagh))*fs/ns;
plot(f,yfftmagh);
xlabel('frequency');ylabel('filtered spectrum');
Low Pass Filter
command window display
Enter 1.lpf 2.hpf 3.bpf 4.brf 1
OUTPUT GRAPHS

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 Digital Signal Processing Lab Manual (KEC 553)

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 Digital Signal Processing Lab Manual (KEC 553)

High Pass Filter

command window display
Enter 1.lpf 2.hpf 3.bpf 4.brf 2

OUTPUT GRAPHS

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 Digital Signal Processing Lab Manual (KEC 553)

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 Digital Signal Processing Lab Manual (KEC 553)

Band Pass Filter

command window display
Enter 1.lpf 2.hpf 3.bpf 4.brf 3

OUTPUT

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 Digital Signal Processing Lab Manual (KEC 553)

GRAPHS

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 Digital Signal Processing Lab Manual (KEC 553)

Band Reject Filter

command window display
Enter 1.lpf 2.hpf 3.bpf 4.brf 4

OUTPUT GRAPHS

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 Digital Signal Processing Lab Manual (KEC 553)

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 Digital Signal Processing Lab Manual (KEC 553)

RESULT: FIR Filter (LP?BP?BP/BR) using frequency sampling methed has been
implemented and their magnitude responses are visualized through DSP processor and
MATLAB.

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 Digital Signal Processing Lab Manual (KEC 553)

Experiment-12 Linear to circular convolution

Program code:

#include<stdio.h>
#include<math.h>
int x[30],h[30],y[30],x2[30],a[30],m,n,i,j;
main()
{
printf("enter length of two sequences");
scanf("%d%d",&m,&n);
printf("enter the first sequence \n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf("enter the second sequence \n");
for(i=0;i<n;i++)
scanf("%d",&h[i]);
for(i=0;i<m+n-1;i++)
{
y[i]=0;
for(j=0;j<=i;j++)
{
y[i]=y[i]+(x[j]*h[i-j]);
}
}
for(i=0;i<n;i++)
y[i]=y[n+i]+y[i];
for(i=0;i<n;i++)
printf("Result is \n y[%d]=%d",i,y[i]);
}

Output:

enter length of two sequences4

4
enter the first sequence
1
2
3
4
enter the second sequence
1
2
1
4
Result is
y[0]=20
y[1]=20
y[2]=24
y[3]=16

Waveforms:

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 Digital Signal Processing Lab Manual (KEC 553)

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 Digital Signal Processing Lab Manual (KEC 553)

Result:

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 Digital Signal Processing Lab Manual (KEC 553)

This lab manual has been updated by

Faculty Name

Dr. Kaustubh Kumar Shukla

[email protected]

Cross checked By
HOD ECE/EEE/ECZ

Verified By
Director, DGI Greater Noida

Please spare some time to provide your valuable feedback.

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