Advanced Digital Signal Processing

MATLAB Assignment

- Manish Kumar
- U101112FEC186
- Batch - S1

EXPERIMENT 1
Downsampling and Upsampling
Theory:
Upsampling : "Upsampling" is the process of inserting zero-valued samples between original
samples to increase the sampling rate ("zero-stuffing"). Upsampling adds to the original
signal undesired spectral images
Downsampling: "Downsampling" is the process of reducing the sampling rate of the
samples.
Here, upsample(x,n) gives the upsampled signal by a factor of n.
downsample(x,n) gives the downsampled signal by a factor of n.

Matlab Code:
t=1:40;
fs=0.2;
a=sin(2*pi*fs*t);
subplot(311)
stem(a);
axis([0 20 -1 1]);
grid on;
title('ORIGINAL');
%Upsampling
ua=upsample(a,2);
subplot(312)
stem(ua);
axis([0 20 -1 1]);
grid on;
title('UPSAMPLED BY 2');
.

%Downsampling
da=downsample(a,2);
subplot(313)
stem(da);
axis([0 20 -1 1]);
grid on;
title('DOWNSAMPLED BY 2');

Output:

EXPERIMENT 2
Interpolation and Decimation
Theory:
Interpolation : "Interpolation" is the process of upsampling followed by filtering. The
filtering removes the undesired spectral images. The interpolation factor is simply the ratio
of the output rate to the input rate.
.

Decimation : It is filtering (to mitigate aliasing distortion) followed by downsampling.

Here, interp(x,n) gives the interpolated signal by a factor of n.
decimate(x,n) gives the decimated signal by a factor of n.

Matlab Code:
t= 0:(1/10000):((4/1000)-(1/10000))
m = cos(2*pi*1000*t)
subplot(3,1,1)
stem(m)
title('Original signal');
y1 = decimate(m,1);
subplot(3,1,2)
stem(y1)
title('Decimation by 1');
axis([0 5 -1 1]);
y2 = interp(m,4);
subplot(3,1,3)
stem(y2)
title('Interpolation by 4');

Output:

EXPERIMENT 3
Polyphase Decomposition
.

Theory:

Matlab Code:
t= 0:1/100:1;
x= sin(2*pi*t);
t0=1:100;
h=fir1(99,0.2,'low');
subplot(221);
stem(t0,h);
y=filter(h,1,x);
subplot(222);
stem(t,y);
t1=0:0.1:1;
yd= y(1:10:end);
subplot(223);
stem(t1,yd);
hp= zeros(10,10);
for i=1:10
hp(i,1:10)= h(i:10:end);
end
xp(1,1:11)=x(1:10:end);
j=10;
for i=2:10
xp(i,1:11)= [0 x(j:10:end)];
j=j-1;
end
yp=zeros(1,11);
for i=1:10
yp= yp+ filter(hp(i,1:10),1,xp(i,1:11));
end
subplot(224);
stem(t1,yp);

Output:

EXPERIMENT 4
SUB-BAND CODING- QMF
Theory:
QMF is used for splitting a signal into two or more subbands in the frequency domain, so that
each subband signal can be processed in an independent manner and sufficient compression
may be achieved.
Eventually, at some point in the process, the subband signals are recombined so that the
original signal is properly reconstructed

Matlab Code:
n= 10;

6
p = fir1(19,1/2);
for i= length(p)
if(i > (length(p)/2)+1)
p(i+1)= p(i);
if(i == (length(p)/2)+1)
p(i) = 0.28;
else
p(i)=p(i);
end
end
end
p=downsample(p,2);
p=upsample(p,2);
Z = roots(p);
hZ= [];
for i=1:length(Z)
if abs(Z(i))>1
hZ = [hZ,Z(i)];
end
end
[h,d] = zp2tf(hZ',[],1);
%coefficients of H0(z)
h1 = wrev(h);
for i = i:length(h1)
if i ==1 || i==3
h1(i)= -h1(i);
end
end
f0 = wrev(h);
f1 = wrev(h1);
[y, fs] = audioread('1.wav');
yl = filter(h,1,y(:,1));
yu = filter(h1,1,y(:,1));
yrl = filter(f0,1,yl);
yru = filter(f1,1,yu);
y1 = yrl + yru;
fvtool(y(:,1),1,y1,1);
fvtool(h,1,h1,1,p,1);

Output:

EXPERIMENT 5
ADAPTIVE FILTERING USING LMS ALGORITHM
.

Theory:
Least mean squares (LMS) algorithms are a class of adaptibve filters used to mimic a desired
filter by finding the filter coefficients that relate to producing the least mean squares of the
error signal

where
x(n) is the input signal to a linear filter
y(n) is the corresponding output signal
d(n) is an additional input signal to the adaptive filter
e(n) is the error signal that denotes the difference between d(n) and y(n).
The LMS algorithm performs the following operations to update the coefficients of an
adaptive FIR filter:
1. Calculates the output signal y(n) from the FIR filter.
Y(n = x(n)Tw(n)
2. Calculates the error signal e(n) by using the following equation: e(n) = d(n)y(n)
3. Updates the filter coefficients by using the following equation:
w(n+1) = w(n) + *e(n)*x(n)
where is the step size of the adaptive filter

Matlab Code1:
%ADAPTIVE FILTERING for signal estimation
clc;
clear all;
close all;
m=16;
mu=0.05; %step size
x=[];
.

x(1)=0;
w=zeros(16,1);
u1=randn(1000,1);
for n=2:1000
x(n)=0.5*(x(n-1)+u1(n)); %original signal
end
d=[];
d=x+cos(0.2.*pi.*(1:1000));
ip =cos(0.2.*pi.*(1:1000)); %noise added to signal
y=[];
e=[];
for i=17:1000
y(i)=w'*(ip(i:-1:(i-m+1)))';
e(i)= d(i)-y(i); %error between i/p and estimated signal
w=w+(mu*(ip(i:-1:(i-m+1)))'*e(i)); %update filter
coefficients
end
error=zeros(1000,1);
error(1)=0;
for i=2:1000
error(i)=error(i-1)+(abs(x(i)-e(i))^2); %energy of error
minimizes as samples increase
end
fvtool(d,1,y,1);
fvtool(error,1);

Output:

10

Matlab Code 2:
clc;
clear all;
close all;
m=16;
mu=0.05; %step size
x=[];
x(1)=0;
w=zeros(16,1);
u1=randn(1000,1);
for n=2:1000
x(n)=0.5*(x(n-1)+u1(n)); %original signal
end
h=fir1(15,0.2);
d=filter(h,1,x);
for i=1:20 %20 iterations
for j=17:1000
y(j)=w'*(x(j:-1:(j-m+1)))';
e(j)= d(j)-y(j); %error between i/p and estimated signal
w=w+(mu*(x(j:-1:(j-m+1)))'*e(j)); %update filter
coefficients
end
end
fvtool(h,1,w,1);

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Output:

EXPERIMENT 6
Linear and Circular Convolution
Theory: One dimensional circular discrete convolution is given by :

Matlab Code:
x = [1 2 3 4 5];
subplot 221
stem(x);
title('x');
axis([0 6 0 10]);
y = [3 6 8];
subplot 222
stem(y);
title('y');
axis([0 6 0 10]);

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c1 = conv(x,y);
subplot 223
stem(c1);
title('linear convolution');
axis([0 10 0 70]);

c2 = cconv(x,y,70);
subplot 224
stem(c2);
title('circular convolution');
axis([0 20 0 70]);

Output :

EXPERIMENT 7
MFCC
Theory:
Mel Frequency Cepstral Coefficents (MFCCs) are a feature widely used in automatic speech
and speaker recognition.
The steps to calculate MFCCs are as follows:
1.
2.
3.
4.
5.

Frame the signal into short frames.

For each frame calculate the periodogram estimate of the power spectrum.
Apply the mel filterbank to the power spectra, sum the energy in each filter.
Take the logarithm of all filterbank energies.
Take the DCT of the log filterbank energies.
6. Keep DCT coefficients 2-13, discard the rest.

Matlab Code:
clear all;

13
close all;
[z,Fs] = audioread('1.wav');
%10ms of sample
m = round(0.01*Fs);
f = fft(z);
log_mag = log(abs(f));
c = ifft(log_mag);
glottal = [zeros(1,m) c(m:end-m)' zeros(1,m)];
vocal = [c(1:m)',zeros(1,length(c)-(2*m)),c(end-m:end)'];
f_glottal = fft(glottal);
f_vocal = fft(vocal);
figure('Name','Cepstrum 1')
subplot(421), plot(z), title('Audio sample of 10ms');
subplot(422), plot(abs(fftshift(f))), title('FFT of sample of 10ms');
subplot(423), plot(log_mag), title('Log magnitude of sample of 10ms');
subplot(424), plot(c), title('Cepstrum of sample of 10ms');
subplot(425), plot(glottal), title('Glottal Value');
subplot(426), plot(vocal), title('Vocal coefficients');
subplot(427), plot(abs(fftshift(f_glottal))), title('FFT of Glottal
Value');
subplot(428), plot(abs(fftshift(f_vocal))), title('FFT of Vocal
coefficients');

Output:

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EXPERIMENT 8
WAVELET TRANSFORM
Theory:
The discrete wavelet transform (DWT) represents a 1-D signal s(t) in terms of shifted versions of a
lowpass scaling function (t) and shifted and dilated versions of a prototype bandpass wavelet
function (t).

Haar Wavelets
For an input represented by a list of
numbers, the Haar wavelet transform may be considered to pair
up input values, storing the difference and passing the sum. This process is repeated recursively, pairing
up the sums to provide the next scale, which leads to
differences and a final sum.

Matlab Code 1:
close all;
clear all;
clc;
[ak,bk] = wfilters('db2');
A = [bk(1) 0 0 0;bk(3) bk(2) bk(1) 0; 0 bk(4) bk(3) bk(2); 0 0 0 bk(1)];
A = 2*A;
[v,b] = eigs(A);
v(1:4,1)= v(1:4,1)/sum(v(1:4,1)); % lambda ~ 1
N = 4;
phi = Scale(v(:,1),bk);
for i = 1:9
subplot(3,3,i)
phi = Scale(phi,bk);
plot(phi);
end
%We get round off errors because we are not multiplying/taking lambda value
= 1
%Try by putting ak also in matrix A

function [ phi ] = Scale( phi,bk )

phi = upsample(phi,2);
bk1 = [bk,zeros(1,length(phi)-length(bk))];
% phi = [phi,zeros(1,length(phi)-length(bk))];
for i = 2:2:length(phi)
for k = 1:2*i-1
if(i<=length(bk))
phi(i) = 2*bk1(k)*phi(2*i-k);
end
end
end

15

Output:
Matlab Code 2:

% wavelet transform
x= randn(1,1000);
y= zeros(1,1000);
for i= 500:550
y(i)= 10.*cos(0.9*pi*i);
end
x=x+y;
subplot(221)
plot(x);
title('x');
[ca,cd]= dwt(x,'haar');
[ca,cd]= dwt(ca,'haar');
subplot(222)
plot(ca);
title('ca');
subplot(223)
plot(cd);
title('cd');

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Output: