I am wondering if there is a way to exclude the lag at 0 from ACF/PACF plot? Because people only interest in the positive lags.
I'm not sure why you would want to do that, since anyone would expect the zero-lag coefficient to be there and would likely to read the plot incorrectly if you exclude it.
In any case, I don't see a neat way of doing it for crosscorrelation, since you output will not in general be symmetrical, and therefore you can't just omit something in the middle of the signal. However, if you're performing autocorrelation, and I believe that you are from what you wrote, you can only keep the positive lags, since you signal is symmetrical. Do to that, I'd write
%# Generate some signal
x = rand(1,10);
%# Compute its autocorrelation, whichever flavor you prefer
acf = xcorr(x);
%# Keep only the positive lags
acf = acf( length(x)+1 : end );
%# Plot your result
stem(acf);
Related
I have a signal 's' of voice of which you can see an extract here:
I would like to plot the zero crossing points in the same graph. I have tried with the following code:
zci = #(v) find(v(:).*circshift(v(:), [-1 0]) <= 0); % Returns Zero-Crossing Indices Of Argument Vector
zx = zci(s);
figure
set(gcf,'color','w')
plot(t,s)
hold on
plot(t(zx),s(zx),'o')
But it does not interpole the points in which the sign change, so the result is:
However, I'd like that the highlighted points were as near as possible to zero.
I hope someone can help me. Thanks you for your responses in advanced.
Try this?
w = 1;
crossPts=[];
for k=1:(length(s)-1)
if (s(k)*s(k+1)<0)
crossPts(w) = (t(k)+t(k+1))/2;
w = w + 1;
end
end
figure
set(gcf,'color','w')
plot(t,s)
hold on
plot(t, s)
plot(crossPts, zeros(length(crossPts)), 'o')
Important questions: what is the highest frequency conponent of the signal you are measuring? Can you remeasure this signal? What is your sampling rate? What is this analysis for? (Schoolwork or scholarly research). You may have quite a bit of trouble measuring the zeros of this function with any significance or accurracy because it looks like your waveform has a frequency greater than half of your sampling rate (greater than your Nyquist frequency). Upsampling/interpolating your entire waveform will allow you to find the zeros much more precisely (but with no greater degree of accurracy) but this is a huge no-no in the scientific community. While my method may not look super pretty, it's the most accurate method that doesn't make unsafe assumptions. If you just want it to look pretty, I would recommend interp1 and using the 'Spline' method. You can interpolate the whole waveform and then use the above answer to find more accurate zeros.
Also, you could calculate the zeros on the interpolated waveform and then display it on the raw data.
A remotely possible solution to improve your data;
If you're measuring a human voice, why not try filtering at the range of human speech? This should be fine mathematically and could possibly improve your waveform.
I want to calculate the Fourier series of a signal using FFT on matlab. And I came across the following unexpected issue. Mere example
If I define a grid and then compute the fft as:
M=59;
x= deal(1*(0:M-1)/M);
y=3*cos(2*pi*x);
Yk=fftshift(fft2(y)/(M));
which gives me the exact analytic values expected: Yk(29)=1.5; Yk(31)=1.5; zeros anything else
but if I define the grid as, and repeat the fft calculation:
x=0:1/(M-1):1;
y=3*cos(2*pi*x);
Yk=fftshift(fft2(y)/(M));
got the Yk's values completely screwed up
This is an annoying issue since I have to analyse many signals data that was sampled as in the second method so the Yk's values will be wrong. Is there a way to workaround this? an option to tell something to the fft function about the way the signal was sampled. Have no way to resample the data in the correct way.
The main reason to avoid have spectral leaking, is that I do further operations with these Fourier terms individually Real and Imag parts. And the spectral leaking is messing the final results.
The second form of sampling includes one sample too many in the period of the cosine. This causes some spectral leaking, and adds a small shift to your signal (which leads to non-zero imaginary values). If you drop the last point, you'll cosine will again be sampled correctly, and you'll get rid of both of these effects. Your FFT will have one value less, I don't know if this will affect your analyses in any way.
x = 0:1/(M-1):1;
y = 3*cos(2*pi*x);
Yk = fftshift(fft2(y(1:end-1))/(M-1));
>> max(abs(imag(Yk)))
ans =
1.837610523517500e-16
I am trying to find power spectrum of the signal. The length of the signal is 100000, sample frequency is 1000Hz,and the number of points is 100000. I found the power spectrum using two approaches. The first one is by taking all the length as one part and found power spectrum for it while the second approach is by dividing the signal into 100*1000and find spectrum for each row then get the mean for all rows. My problem is that I must get the same answer in both approaches but I got different answers. I do not know what is the error in my code.
N=100000;
SF=1000;
a=0.1;
b=0.3;
amplitude1=1;
amplitude2=0.5;
t=0:1/SF:100;
f1=SF*a;
f2=SF*b;
A=amplitude1*sin(2*pi*f1*t)+amplitude2*sin(2*pi*f2*t);
Y=2*randn(1,length(A))+A;
bin=[0 :N/2];
fax_Hz=(bin*SF)/N;
FFT=fft(Y);
spectra=2/(SF*length(Y))*(FFT.*conj(FFT));
plot(fax_Hz,spectra(1,1:50001));
D=reshape(Y(1,1:100000),[100,1000]);
M=length(D(1,:));
for i=1:100
FFT_1(i,:)=fft(D(i,:));
S(i,:)=(2/(SF*M))*(FFT_1(i,:).*conj(FFT_1(i,:)));
end
S_f=mean(S);
figure
plot (S_f);
I just update the code. I do not know but when I added noise to signal the two plots looks shifted.
The main problem is with reshape you are working with each row being a separate sequence. Reshape however fills the first column before moving to the second one.
You can use the following instead.
D=reshape(A(1,1:100000),[1000,100]).';
Normalization is another problem. You can either use ifft instead of fft as it is normalized by default (not sure why). Or alternatively keep your normalization and instead of using mean you should can use sum, maybe that is due to a mistake you might have made. There still seems to be a small discrepancy in the amplitudes, not sure where that is coming from.
At the end to plot use the following:
bin=[0 :N];
fax_Hz=(bin*SF)/N;
FFT=ifft(A);
spectra=FFT.*conj(FFT);
plot(fax_Hz,spectra); hold on
D=reshape(A(1,1:100000),[1000,100]).';
M=length(D(1,:));
for i=1:100
FFT_1(i,:)=ifft(D(i,:));
S(i,:)=FFT_1(i,:).*conj(FFT_1(i,:));
end
S_f=mean(S);
plot(fax_Hz(1:100:end-1), S_f);
Note: the fax_Hz(1:100:end-1) is a hacky way of getting the length of the vectors to be the same.
I am wondering if I am using Fourier Transformation in MATLAB the right way. I want to have all the average amplitudes for frequencies in a song. For testing purposes I am using a free mp3 download of Beethovens "For Elise" which I converted to a 8 kHz mono wave file using Audacity.
My MATLAB code is as follows:
clear all % be careful
% load file
% Für Elise Recording by Valentina Lisitsa
% from http://www.forelise.com/recordings/valentina_lisitsa
% Converted to 8 kHz mono using Audacity
allSamples = wavread('fur_elise_valentina_lisitsa_8khz_mono.wav');
% apply windowing function
w = hanning(length(allSamples));
allSamples = allSamples.*w;
% FFT needs input of length 2^x
NFFT = 2^nextpow2(length(allSamples))
% Apply FFT
fftBuckets=fft(allSamples, NFFT);
fftBuckets=fftBuckets(1:(NFFT/2+1)); % because of symetric/mirrored values
% calculate single side amplitude spectrum,
% normalize by dividing by NFFT to get the
% popular way of displaying amplitudes
% in a range of 0 to 1
fftBuckets = (2*abs(fftBuckets))/NFFT;
% plot it: max possible frequency is 4000, because sampling rate of input
% is 8000 Hz
x = linspace(1,4000,length(fftBuckets));
bar(x,fftBuckets);
The output then looks like this:
Can somebody please tell me if my code is correct? I am especially wondering about the peaks around 0.
For normalizing, do I have to divide by NFFT or length(allSamples)?
For me this doesn't really look like a bar chart, but I guess this is due to the many values I am plotting?
Thanks for any hints!
Depends on your definition of "correct". This is doing what you intended, I think, but it's probably not very useful. I would suggest using a 2D spectrogram instead, as you'll get time-localized information on frequency content.
There is no one correct way of normalising FFT output; there are various different conventions (see e.g. the discussion here). The comment in your code says that you want a range of 0 to 1; if your input values are in the range -1 to 1, then dividing by number of bins will achieve that.
Well, exactly!
I would also recommend plotting the y-axis on a logarithmic scale (in decibels), as that's roughly how the human ear interprets loudness.
Two things that jump out at me:
I'm not sure why you are including the DC (index = 1) component in your plot. Not a big deal, but of course that bin contains no frequency data
I think that dividing by length(allSamples) is more likely to be correct than dividing by NFFT. The reason is that if you want the DC component to be equal to the mean of the input data, dividing by length(allSamples) is the right thing to do.
However, like Oli said, you can't really say what the "correct" normalization is until you know exactly what you are trying to calculate. I tend to use FFTs to estimate power spectra, so I want units like "DAC / rt-Hz", which would lead to a different normalization than if you wanted something like "DAC / Hz".
Ultimately there's no substitute for thinking about exacty what you want to get out of the FFT (including units), and working out for yourself what the correct normalization should be (starting from the definition of the FFT if necessary).
You should also be aware that MATLAB's fft has no requirement to use an array length that is a power of 2 (though doing so will presumably lead to the FFT running faster). Because zero-padding will introduce some ringing, you need to think about whether it is the right thing to do for your application.
Finally, if a periodogram / power spectrum is really what you want, MATLAB provides functions like periodogram, pwelch and others that may be helpful.
I am trying to compare two data sets in MATLAB. To do this I need to filter the data sets by Fourier transforming the data, filtering it and then inverse Fourier transforming it.
When I inverse Fourier transform the data however I get a spike at either end of the red data set (picture shows the first spike), it should be close to zero at the start, like the blue line. I am comparing many data sets and this only happens occasionally.
I have three questions about this phenomenon. First, what may be causing it, secondly, how can I remedy it, and third, will it affect the data further along the time series or just at the beginning and end of the time series as it appears to from the picture.
Any help would be great thanks.
When using DFT you must remember the DFT assumes a Periodic Signal (As a Superposition of Harmonic Functions).
As you can see, the start point is exact continuation of the last point in harmonic function manner.
Did you perform any Zero Padding in the Spectrum Domain?
Anyhow, Windowing might reduce the Overshooting.
Knowing more about the filter and the Original data might be helpful.
If you say spike near zero frequencies, I answer check the DC component.
You seem interested by the shape, so doing
x = x - mean(x)
or
x -= mean(x)
or
x -= x.mean()
(I love numpy!)
will just constrain the dataset to begin with null amplitude at zero-frequency and to go ahead with comapring the spectra's amplitude.
(as a side-note: did you check that you approprately use fftshift and ifftshift? this has always been the source of trouble for me)
Could be the numerical equivalent of Gibbs' phenomenon. If that's correct, there's no way to remedy it except for filtering.