ECG signal filtering problem

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M
M on 29 Mar 2021
Answered: Sharon on 9 Apr 2024
This is my ECG signal I wanna improve. Firstly i would like to cut and make high frequencies kind of equal:
Secondly I would like to smooth the noise below.
How to create a filter like this using filterdesign? I found some parameters from other questions concerning it but after I put it in filterdesign, it was destroying my signal like doubling its amplitude etc. I'd be grateful for any help

Accepted Answer

Mathieu NOE
Mathieu NOE on 29 Mar 2021
hello
a little tool to do the fft anamysis (and notch filter demo included)
clc
clearvars
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
% load signal
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
%% dummy data
Fs = 10000;
samples = 20000;
t = (0:samples-1)'*1/Fs;
signal = sin(2*pi*50*t)+2*sin(2*pi*440*t)+1e-2*randn(samples,1); % two sine + some noise
%% decimate (if needed)
% NB : decim = 1 will do nothing (output = input)
decim = 5;
if decim>1
signal = decimate(signal,decim);
Fs = Fs/decim;
end
samples = length(signal);
t = (0:samples-1)*1/Fs;
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
% FFT parameters
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
NFFT = Fs; % so delta f = 1 Hz
Overlap = 0.75;
w = hanning(NFFT); % Hanning window / Use the HANN function to get a Hanning window which has the first and last zero-weighted samples.
%% notch filter section %%%%%%
% H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
fc = 50; % notch freq
wc = 2*pi*fc;
Q = 5; % adjust Q factor for wider (low Q) / narrower (high Q) notch
% at f = fc the filter has gain = 0
w0 = 2*pi*fc/Fs;
alpha = sin(w0)/(2*Q);
b0 = 1;
b1 = -2*cos(w0);
b2 = 1;
a0 = 1 + alpha;
a1 = -2*cos(w0);
a2 = 1 - alpha;
% analog notch (for info)
num1=[1/wc^2 0 1];
den1=[1/wc^2 1/(wc*Q) 1];
% digital notch (for info)
num1z=[b0 b1 b2];
den1z=[a0 a1 a2];
freq = (fc-1:0.01:fc+1);
[g1,p1] = bode(num1,den1,2*pi*freq);
g1db = 20*log10(g1+1e-6);
[gd1,pd1] = dbode(num1z,den1z,1/Fs,2*pi*freq);
gd1db = 20*log10(gd1+1e-6);
figure(1);
plot(freq,g1db,'b',freq,gd1db,'+r');
title(' Notch: H(s) = (s^2 + 1) / (s^2 + s/Q + 1)');
legend('analog','digital');
xlabel('Fréquence (Hz)');
ylabel(' dB')
% now let's filter the signal
signal_filtered = filtfilt(num1z,den1z,signal);
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
% display : averaged FFT spectrum before / after notch filter
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
[freq,fft_spectrum] = myfft_peak(signal, Fs, NFFT, Overlap); %
sensor_spectrum_dB = 20*log10(fft_spectrum);% convert to dB scale (ref = 1)
% demo findpeaks
[pks,locs]= findpeaks(sensor_spectrum_dB,'SortStr','descend','NPeaks',2);
[freq,fft_spectrum_filtered] = myfft_peak(signal_filtered, Fs, NFFT, Overlap);
sensor_spectrum_filtered_dB = 20*log10(fft_spectrum_filtered);% convert to dB scale (ref = 1)
figure(2),plot(freq,sensor_spectrum_dB,'b',freq,sensor_spectrum_filtered_dB,'r');grid
title(['Averaged FFT Spectrum / Fs = ' num2str(Fs) ' Hz / Delta f = ' num2str(freq(2)-freq(1)) ' Hz ']);
legend('before notch filter','after notch filter');
xlabel('Frequency (Hz)');ylabel(' dB')
text(freq(locs)+1,pks,num2str(freq(locs)))
function [freq_vector,fft_spectrum] = myfft_peak(signal, Fs, nfft, Overlap)
% FFT peak spectrum of signal (example sinus amplitude 1 = 0 dB after fft).
% Linear averaging
% signal - input signal,
% Fs - Sampling frequency (Hz).
% nfft - FFT window size
% Overlap - buffer percentage of overlap % (between 0 and 0.95)
[samples,channels] = size(signal);
% fill signal with zeros if its length is lower than nfft
if samples<nfft
s_tmp = zeros(nfft,channels);
s_tmp((1:samples)) = signal;
signal = s_tmp;
samples = nfft;
end
% window : hanning
window = hanning(nfft);
window = window(:);
% compute fft with overlap
offset = fix((1-Overlap)*nfft);
spectnum = 1+ fix((samples-nfft)/offset); % Number of windows
% % for info is equivalent to :
% noverlap = Overlap*nfft;
% spectnum = fix((samples-noverlap)/(nfft-noverlap)); % Number of windows
% main loop
fft_spectrum = 0;
for i=1:spectnum
start = (i-1)*offset;
sw = signal((1+start):(start+nfft),:).*(window*ones(1,channels));
fft_spectrum = fft_spectrum + (abs(fft(sw))*4/nfft); % X=fft(x.*hanning(N))*4/N; % hanning only
end
fft_spectrum = fft_spectrum/spectnum; % to do linear averaging scaling
% one sidded fft spectrum % Select first half
if rem(nfft,2) % nfft odd
select = (1:(nfft+1)/2)';
else
select = (1:nfft/2+1)';
end
fft_spectrum = fft_spectrum(select,:);
freq_vector = (select - 1)*Fs/nfft;
end

More Answers (1)

Sharon
Sharon on 9 Apr 2024

clc clearvars

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% % load signal %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%

%% dummy data Fs = 10000; samples = 20000; t = (0:samples-1)'*1/Fs; signal = sin(2*pi*50*t)+2*sin(2*pi*440*t)+1e-2*randn(samples,1); % two sine + some noise

%% decimate (if needed) % NB : decim = 1 will do nothing (output = input) decim = 5; if decim>1 signal = decimate(signal,decim); Fs = Fs/decim; end samples = length(signal); t = (0:samples-1)*1/Fs;

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% % FFT parameters %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% NFFT = Fs; % so delta f = 1 Hz Overlap = 0.75; w = hanning(NFFT); % Hanning window / Use the HANN function to get a Hanning window which has the first and last zero-weighted samples.

%% notch filter section %%%%%% % H(s) = (s^2 + 1) / (s^2 + s/Q + 1)

fc = 50; % notch freq wc = 2*pi*fc; Q = 5; % adjust Q factor for wider (low Q) / narrower (high Q) notch % at f = fc the filter has gain = 0

w0 = 2*pi*fc/Fs; alpha = sin(w0)/(2*Q);

            b0 =   1;
            b1 =  -2*cos(w0);
            b2 =   1;
            a0 =   1 + alpha;
            a1 =  -2*cos(w0);
            a2 =   1 - alpha;

% analog notch (for info) num1=[1/wc^2 0 1]; den1=[1/wc^2 1/(wc*Q) 1];

% digital notch (for info) num1z=[b0 b1 b2]; den1z=[a0 a1 a2];

freq = (fc-1:0.01:fc+1); [g1,p1] = bode(num1,den1,2*pi*freq); g1db = 20*log10(g1+1e-6);

[gd1,pd1] = dbode(num1z,den1z,1/Fs,2*pi*freq); gd1db = 20*log10(gd1+1e-6);

figure(1); plot(freq,g1db,'b',freq,gd1db,'+r'); title(' Notch: H(s) = (s^2 + 1) / (s^2 + s/Q + 1)'); legend('analog','digital'); xlabel('Fréquence (Hz)'); ylabel(' dB')

% now let's filter the signal signal_filtered = filtfilt(num1z,den1z,signal);

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% % display : averaged FFT spectrum before / after notch filter %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%

[freq,fft_spectrum] = myfft_peak(signal, Fs, NFFT, Overlap); % sensor_spectrum_dB = 20*log10(fft_spectrum);% convert to dB scale (ref = 1)

% demo findpeaks [pks,locs]= findpeaks(sensor_spectrum_dB,'SortStr','descend','NPeaks',2);

[freq,fft_spectrum_filtered] = myfft_peak(signal_filtered, Fs, NFFT, Overlap); sensor_spectrum_filtered_dB = 20*log10(fft_spectrum_filtered);% convert to dB scale (ref = 1)

figure(2),plot(freq,sensor_spectrum_dB,'b',freq,sensor_spectrum_filtered_dB,'r');grid title(['Averaged FFT Spectrum / Fs = ' num2str(Fs) ' Hz / Delta f = ' num2str(freq(2)-freq(1)) ' Hz ']); legend('before notch filter','after notch filter'); xlabel('Frequency (Hz)');ylabel(' dB') text(freq(locs)+1,pks,num2str(freq(locs)))

function [freq_vector,fft_spectrum] = myfft_peak(signal, Fs, nfft, Overlap) % FFT peak spectrum of signal (example sinus amplitude 1 = 0 dB after fft). % Linear averaging % signal - input signal, % Fs - Sampling frequency (Hz). % nfft - FFT window size % Overlap - buffer percentage of overlap % (between 0 and 0.95)

[samples,channels] = size(signal);

% fill signal with zeros if its length is lower than nfft if samples<nfft s_tmp = zeros(nfft,channels); s_tmp((1:samples)) = signal; signal = s_tmp; samples = nfft; end

% window : hanning window = hanning(nfft); window = window(:);

% compute fft with overlap offset = fix((1-Overlap)*nfft); spectnum = 1+ fix((samples-nfft)/offset); % Number of windows % % for info is equivalent to : % noverlap = Overlap*nfft; % spectnum = fix((samples-noverlap)/(nfft-noverlap)); % Number of windows

    % main loop
    fft_spectrum = 0;
    for i=1:spectnum
        start = (i-1)*offset;
        sw = signal((1+start):(start+nfft),:).*(window*ones(1,channels));
        fft_spectrum = fft_spectrum + (abs(fft(sw))*4/nfft);     % X=fft(x.*hanning(N))*4/N; % hanning only 
    end
    fft_spectrum = fft_spectrum/spectnum; % to do linear averaging scaling

% one sidded fft spectrum % Select first half if rem(nfft,2) % nfft odd select = (1:(nfft+1)/2)'; else select = (1:nfft/2+1)'; end fft_spectrum = fft_spectrum(select,:); freq_vector = (select - 1)*Fs/nfft; end

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