MATLAB: Time to frequency domain

fftfilter

Hi guys I have to convert the signal generated from accelerator(recorded in xls file) which is in time series to freq domian. So far I did this

% read data

data = xlsread('X');

%Frequency Analysis

time = data(:,1); % sampling time

signal = data(:,2); % signal data in Time-Domain

Ts=time;

Fs=20000; % sampling frequency

Now I want to convert this time signal to frequency signal with filtering . What should I do to get Frequency domain and filtering. Thank you in advance.

Best Answer

Read your data and plot your signal with this:

% read data
data = xlsread('X');
%Frequency Analysis
time = data(:,1);                                   % Time Vector
signal = data(:,2);                                 % Signal data in Time-Domain
N = length(signal);                                 % Number Of Samples
Ts = mean(diff(time));                              % Sampling Interval
Fs = 1/Ts;                                          % Sampling Frequency
Fn = Fs/2;                                          % Nyquist Frequency
FT_Signal = fft(signal)/N;                          % Normalized Fourier Transform Of Data
Fv = linspace(0, 1, fix(N/2)+1)*Fn;                 % Frequency Vector (For ‘plot’ Call)
Iv = 1:length(Fv);                                  % Index Vector (Matches ‘Fv’)
figure(1)
plot(Fv, abs(FT_signal(Iv))*2)
grid

The ‘filtering’ step requires that you define the characteristics you want for the filter, and then design it, and filter your signal. You can filter it in the frequency-domain with the fftfilt (link) function, however it requires that you give it a finite-impulse-response or FIR filter. There are several ways to design your filter, the easiest being the designfilt (link) function.

You can also filter your signal in the time domain with either a FIR filter or an infinite-impulse-respone or IIR filter. Design your filter with the designfilt function, then use the filtfilt function to filter the signal with your filter.

See the documentation for the various functions to understand what they do and how to use them.

Related Solutions

MATLAB: 50 Hz interference and noise reduction from ECG.

hello

If you want only to remove the 50 Hz tone , use a notch filter as shown in example below. I used your data and guessed the sampling frequency from the fact that the peak was at 50 Hz.

So sampling at 10 kHz is pretty high for an ECG signal, you could easily go down by factor 10.

%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%



% load signal
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
%% data
signal = importdata('Sinal1.txt');
dt = 1e-4; % 0.1 milli seconds
samples = length(signal);
Fs = 1/dt; % sampling frequency (Hz)
%% decimate (if needed)
% NB : decim = 1 will do nothing (output = input)
decim = 1;
if decim>1
    signal = decimate(signal,decim);
    Fs = Fs/decim;
end
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
% FFT parameters
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
NFFT = Fs;    % 
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 = 7;         % 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 = linspace(fc-1,fc+1,200);
[g1,p1] = bode(num1,den1,2*pi*freq);
g1db = 20*log10(g1);
[gd1,pd1] = dbode(num1z,den1z,1/Fs,2*pi*freq);
gd1db = 20*log10(gd1);
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)

[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),semilogx(freq,sensor_spectrum_dB,'b',freq,sensor_spectrum_filtered_dB,'r');grid on
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')
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 overlap % (between 0 and 0.95)
signal = signal(:);
samples = length(signal);
% fill signal with zeros if its length is lower than nfft
if samples<nfft
    s_tmp = zeros(nfft,1);
    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;
        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

MATLAB: How to divide the signal to AC and DC component.

After you calculate

Y = fft(signal,NFFT)/L;

then Y(1) is the average DC component, and Y(2:end) have to do with the AC components.

Best Answer

Related Solutions

MATLAB: 50 Hz interference and noise reduction from ECG.

MATLAB: How to divide the signal to AC and DC component.

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