Digital Signal Processor Architecture Lab

Nitish S. Prabhu
1MS10EC071

Table of contents
Analysis of system form it i/o equations .................................................................................................. 1
Impulse response ...................................................................................................................................... 2
Sampling theorem..................................................................................................................................... 3
FIR filter - window method ....................................................................................................................... 5
IIR Low pass filters .................................................................................................................................... 9
IIR Filter - using bilenear transformation .............................................................................................. 112

Analysis of system form its i/o equations

Compute the magnitude and phase plots form the coefficients of the differential equation.
clc
clear all
close all
b=[1,2,3];

% Feedforward coefficients

a=1;

% Feedback coefficients

n=100;
% Plot the Pole-Zero diagram (Fig 1.a)
subplot(4,1,1);
zplane(b,a);
% Plot the frequency response (Fig 1.b)
subplot(4,1,2);
[h,w]=freqz(b,a,n);
plot(h);
% Plot the phase response (Fig 1.c)
subplot(4,1,3);
phase=angle(h);plot(phase);

% Plot the magnitude response (Fig 1.d)

subplot(4,1,4);
mag=abs(h);
plot(mag);

Fig.1 :Plots for a given i/o differential equation

Impulse response
To determine the impulse response of a given system and thus model the behavior of the system
clc
clear all
close all
n=-20:100;

% Discrete time variable

b=1;

% Coefficients

a=[1 -0.1 -0.9];

x=(n==0);

% Impulse function

stem(n,x);
title('Filter Output');
figure
y=filter(b,a,x);
stem(n,y)
title('Unit Impulse function');

Sampling theorem
The setup below explains the requirement of fast sampling rates greater than that of Nyquest rates. The
behavior of a system to sampling rates equal to and greater than that of Nyquest frequency is tested.
clc
close all
clear all
f1= 500;
f2= 2000;
% Define the parameters of the signal
N=256;
n=0:N-1;
fs=2*max(f1,f2);
% Case1: Nyquist rate sampling with 'a' and 'b' as input signals
a=sin(2*pi*(f1)*n);

b=sin(2*pi*(f2)*n);
x=sin(2*pi*(f1/fs)*n)+sin(2*pi*(f2/fs)*n);
nt=n./fs;
% Plot of the sine wave to be sampled
subplot(2,2,1)
plot(nt,x)
xlabel('Time(s)')
ylabel('Amplitude')
title('Sampling at nyquist rate')
grid;
X=fft(x);
mag=abs(X);
fhz=n.*(fs/N);
% The frequecy domain representation
subplot(2,2,3)
plot(fhz(1:N/2),mag(1:N/2));
xlabel('Frequency')
ylabel('Amplitude')
title('Sinewave in freq domain')
grid

% Case2: Sampling at higher rates than Nyquest rate

fh=2*2*max(f1,f2);
z=sin(2*pi*(f1/fh)*n)+sin(2*pi*(f2/fh)*n);
nh=n./fh;
% Plot of the signal to be sampled
subplot(2,2,2)
plot(nh,z);
xlabel('time(seconds)');
ylabel('amplitude');
title('sampling at nyquist rate');
grid;
p=fft(x);
mag1=abs(X);
fhz1=n.*(fh/N);

% The frequency domain representation

subplot(2,2,4)
plot(fhz1(1:N/2),mag(1:N/2));
xlabel('frequency');
ylabel('amplitude');
title('sinewave in freq domain');
grid;

FIR filter - window method

Creating an FIR filter using window method based on the filter specifications like pass band and stop
bang attenuation.
clear all
close all
clc
% Define filter parameters
FS = 8000;

% Max frequency

FN = FS/2;

% Nyquest frequency

fp = 1500;

% Passband frequency

tw = 500;

% Transition Width

fs = fp + tw;
fc = (fp+fs)/2/FN;
tw = (fs - fp)/FS;
A = 5;

% Stop band Attenuation

for A = 15:15:60
if A<=21
N = ceil(2/tw);
wn = boxcar(N);
str = 'Rect wind';
elseif A>21 && A<=44
N = ceil(4/tw);
wn = hanning(N);
str = 'Hanning';
elseif A>44 && A<=53
N = ceil(4/tw);
wn = hamming(N);
str = 'Hamming';
else
N = ceil(4/tw);
wn = blackman(N);
str = 'Blackman';
end
disp(str);
hd = fir1( N-1, fc, boxcar(N));
hn = fir1( N-1,fc, wn);
disp('FIR coefficients');
disp(hn);
[H,f]= freqz(hn,1,512,FS);
mag=(20*log10(abs(H)));
figure,plot(f,mag), grid on
xlabel ('Freq(Hz)');

ylabel('Mag resp(dB)');
title(strcat(str,' Window'));
end

Rectangular window :
FIR coefficients
Columns 1 through 7
0.0126

0.0188

-0.0066

-0.0246

-0.0026

0.0281

0.0153

0.0224

0.0537

-0.0067

-0.0878

-0.0358

0.3916

0.1815

-0.0358

-0.0878

-0.0067

-0.0318

-0.0281

0.0153

0.0281

-0.0026

Columns 8 through 14
-0.0281

-0.0318

Columns 15 through 21
0.1815

0.3916

Columns 22 through 28
0.0537

0.0224

Columns 29 through 32
-0.0246

-0.0066

0.0188

0.0126

Fig.1 :Frequency response for rectangular window

Fig.2 : Frequency response for Hanning window

Fig.3 : Frequency response for Hamming winndow

Fig.4 : Frequency response for Blackman window

IIR Low pass filters

%IIR LPF -- Butterworth characteristic, Impulse invariant transformation
clc
clear all
close all
%N=2, fs=1.28kHz, fc=150Hz
%Filter order
N=2;
%Sampling frequency
fs=1280;
%Cut off frequency
fc=150;
wc=2*pi*fc;
%Analog filter design
[b,a]=butter(N,wc,'s');
disp('Coefficients of analog filter');
disp(b);
disp(a);
[z,p,k]=butter(N,wc,'s');
disp('zeros, poles & gain of analog filter');
disp(z);
disp(p);
disp(k);
%Convert to discrete filter
[bz,az]=impinvar(b,a,fs);
disp('Coefficients of the discrete filter');
disp(bz);
disp(az);
%Plot magnitude response
subplot(2,1,1);
[H,f]=freqz(bz,az,512,fs);
plot(f,20*log10(abs(H)));
grid;
xlabel('Frequency(Hz)');
ylabel('Magnitude response(dB)');

%plot pole-zero diagram

subplot(2,1,2);
zplane(bz,az);
%Determine poles and zeros
zz=roots(bz);
pz=roots(az);
disp('Zeros and poles of the discrete filter');
disp(zz);
disp(pz);

Coefficients of analog filter

1.0e+05 *
0

8.8826

0.0133

8.8826

1.0e+05 *
0.0000

Zeros, poles & gain of analog filter

1.0e+02 *
-6.6643 + 6.6643i
-6.6643 - 6.6643i
8.8826e+05
Coefficients of the discrete filter
0
0.3078
0
1.0000

-1.0308

0.3530

Zeros and poles of the discrete filter

0
0.5154 + 0.2955i
0.5154 - 0.2955i

IIR Filter - using bilinear transformation

To Design a simple low pass digital IIR filter with Butterworth characteristics to meet the following
specifications Cut off frequency=150Hz, sampling frequency 1.28kHz and filter order 2.
clc
clear all
close all
%IIR LPF -- Butterworth characteristic, Bilinear transformation
%N=2, fs=1.28kHz, fc=150Hz
N=2;

%Filter order

fs=1280;

%Sampling frequency

fc=150;

%Cut off frequency

wc=2*pi*fc;
%Analog filter design -- Normalised Butterworth filter
[b,a]=butter(N,1,'s');
[z,p,k]=butter(N,1,'s');

%Convert to discrete filter

wc=tan(2*pi*(fc/fs)/2);

%Prewarping

%Low pass to low pass transformation

[b1,a1]=lp2lp(b,a,wc);
[bz,az]=bilinear(b1,a1,0.5);

%2*Fs=1, Fs=0.5

disp('coefficients of the discrete filter ');

disp(bz);
disp(az);
%Plot magnitude response
subplot(2,1,1);
[H,f]=freqz(bz,az,512,fs);
plot(f,20*log10(abs(H)));
grid;
xlabel('Frequency(Hz)');
ylabel('Magnitude response(dB)');
%plot pole-zero diagram
subplot(2,1,2);
zplane(bz,az);
%determine poles and zeros
[z,p,k]=tf2zp(bz,az);
disp('zeros, poles and gain of the discrete filter');
disp(z);
disp(p);
disp(k);

coefficients of the discrete filter

0.0878
0.1756
0.0878
1.0000

-1.0048

0.3561

Zeros, poles and gain of the discrete filter

-1
-1
0.5024 + 0.3220i
0.5024 - 0.3220i
0.0878