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Matlab Code STFT

This document analyzes a heart sound signal from a loaded data set. It performs signal processing steps like framing, windowing, filtering and fast Fourier transform. It summarizes the murmur level, applies butterworth highpass filters, and visualizes the original, filtered and spectrogram signals. The user can play the original and filtered signals.

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ARNajeeb
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89 views

Matlab Code STFT

This document analyzes a heart sound signal from a loaded data set. It performs signal processing steps like framing, windowing, filtering and fast Fourier transform. It summarizes the murmur level, applies butterworth highpass filters, and visualizes the original, filtered and spectrogram signals. The user can play the original and filtered signals.

Uploaded by

ARNajeeb
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as TXT, PDF, TXT or read online on Scribd
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clear all;

clc;
load Murmurset;
f=input('please enter the date sequence :');
f2=int8(f);
fileheart='heart';
str1=date
str1datenum(date, 'dd-mmm-yyyy')
k=data(f) .Murmur;
frqsample=data(f) .fs;

fileheart= sprintf('%s_%i_%i_%i',fileheart,f2,str1)
fprintf('====================================================\n');
fprintf('Welcome to heard sound PCG Analysis\n');
fprintf('You have to requested data set number %i\n',f);
fprintf('Murmur lever is %i\n',k);
swtich k
case 2
frpintf('Murmur intensity:
case 3
fprintf('Murmur intensity:
case 4
fprintf('Murmur intensity:
case 5
fprintf('Murmur intensity:
le\n')
end

no Murmur \n')
week Murmur \n')
strong Murmur \n')
Murmur noted in Journal but not audib

fprintf('Frequency sampling from data structure :%i \n',freqsample);


fprintf('Creating wave file \n');
wavewrite (0.95*data(f) .data/max(abs(data(f) .data)),4000,fileheart);
fprintf('Created wave file %s \n',fileheart);
[y,fs,nbits] = wavread(fileheart);
reply=input('Play heart signal? [before filter] Y/N: ','s');
if reply == 'Y'
sound (y,fs,nbits)
end
fprintf ('frequency sampling from heart file : %i\n',fs);
maxfreq = max(y);
fprintf('Maximum intensity :%f5 \n',maxfreq);

n=length(y); %signal length in terms of sampling points !!!


time = (0:(N-1))'/fs; %Time increament for original signal
fameSoze = floor((256/fs)*fs);%frame size
overlap = floor(((256/fs)*fs)/2); %overlap of frame in sampling points
opt = 'nodelay';
z=[]; %array (strong)
for i=1:size(y,2) %i will repeat until 32000(signal length)
%====== Framming/segmenting the speach signal=======% to get quiasi stationary
%signal

x=y(:,i)l
[out,z,opt]=buffer[z;x],frameSize, overlap, opt);
%===== Windowing the speach signal=====%
windowed(:,i) = hamming(frameSize).*out(;,i);
end

figure('Name', 'Figure 1','Numbertitle','off');


subplot(3,1,1)
plot(out(:));
xlabel('Time (s)')l
ylabel('Amplitude');
grid on
axis tight
title('Block Framming');
subplot(3,1,2)
plot(windowed(1:end));
xlabel('Frame Size')l
ylabel('Amplitude');
grid on
axis tight
title('Windowing');
%==========Performing the Fast Fourier Transfor(FFT)==========%
[Pwr_spectrum,xf,freq]=heartFFT(y,fs);
subplot(3,1,3)
plot(frq,Pwr_specturm);
xlabel('Frequency (Hz)')l
ylabel('Power (dB)');
grid on
axis tight
title('PSD');
%applying butterworth highpass filter
fNorml=100/(fs/s);
[b, a]=butter(8,fNorml, 'high');
x1=filter(b,a,y);
reply1=input('Play Filtered Heart Signal? Y/N:
if reply1 == 'Y';
sound (x1,fs,nbits)
end

','s');

fNorm2=25/(fs/2);
[b,a] butter(8,fnorm2,'high');
x2=filter(b,a,y);
figure ('Name', 'Figure 2', 'Numbertitle', 'off');
subplot(3,1,1)
plote(time,y);grid on
xlabel('Time (s)');ylabel ('Amplitude');
titel(Orignal Signal');

subplot(3,1,2)
plote(time,xq);grid on
xlabel('Time (s)');ylabel ('Amplitude');
titel(Visualize Murmur');
subplot(3,1,3)
plote(time,x2);grid on
xlabel('Time (s)');ylabel ('Amplitude');
titel(Visualize sls2');

figure ('Name', 'Frequecny Response of Filter', 'Numbertitle', 'off');


freqz(b,a);
figure ('Name', 'Spectrogram', 'Numbertitle', 'off');
subplot(2,1,1)
specgram(y,256,fs);
title('Orignal spectrogram');
subplot(2,1,2)
specgram(xq,256,fs);
title('Filtered spectrogram');

Heart FFt Call Function


function [Pwr_specturm,xf,freq]=heartFFT(Signal, samplingFrequency)
N=length(signal); %length of signal in sampling points
t= (0:(N-1))/samplingFrequency; %Time increament t=N/fs
freq=(0:(N-1)/2)*(samplingFrequency/N); %frequency resolution k*fs/N
xf=fft(Signal);
xf=xf(1:lenght(freq));
Pwr_spectrum=(20*log10(abs(xf)));%.^2)/N %power of signal

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