Hello guys, so I have this problem, I do have an audio signal, from two seperate sources with some noise. I need to seperate thoose into proper frequencies and print them out, how would I do that ? I already have the spectrum of the signal, that i acquired using FFT. But i get a lot of frequencies grouped in two spots, i don't get it how i get only the frequencies i need ? I added picture of the spectrum above.
MATLAB: How to filter all the frequencies from the audio signal.
audioband pass filterfourierfrequencysignalSignal Processing Toolboxtransformation
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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 ontitle(['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)'; endfft_spectrum = fft_spectrum(select);freq_vector = (select - 1)*Fs/nfft;end
hello
see below , we are generating a signal with 2 frequencies , then do fft for showing their frequencies.
the time plot will not be very useful but I let it for your info
you can easily adapt it to your own needs
the notch filter is a bonus , to show how filter out one disturbing frequency
hope it helps
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
% load signal
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% dummy data
Fs = 1000;samples = 25000;t = (0:samples-1)'*1/Fs;signal = cos(2*pi*50*t)+cos(2*pi*100*t)+1e-3*rand(samples,1); % two sine + some noise
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% FFT parameters
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%NFFT = 1000; %
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 = 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)
% demo findpeaks
df = mean(diff(freq));[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),semilogx(freq,sensor_spectrum_dB,'b',freq,sensor_spectrum_filtered_dB,'r');gridtitle(['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(locs+.02,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 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)'; endfft_spectrum = fft_spectrum(select);freq_vector = (select - 1)*Fs/nfft;end
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