I'm looking for the most abundant frequency in a periodic signal.
I'm trying to understand what do I get if I perform a Fourier transformation on a periodic signal and filter for frequencies which have negative fft values.
In other words, what do the axis of plots 2 and 3 (see below) express? I'm plotting frequency (cycles/second) over the fft-transformed signal - what do negative values on the y axis mean, and would it make sense that I'd be interested in only those?
import numpy as np
import scipy
# generate data
time = scipy.linspace(0,120,4000)
acc = lambda t: 10*scipy.sin(2*pi*2.0*t) + 5*scipy.sin(2*pi*8.0*t) + 2*scipy.random.random(len(t))
signal = acc(time)
# get frequencies from decomposed fft
W = np.fft.fftfreq(signal.size, d=time[1]-time[0])
f_signal = np.fft.fft(signal)
# filter signal
# I'm getting only the "negative" part!
cut_f_signal = f_signal.copy()
# filter noisy frequencies
cut_f_signal[(W < 8.0)] = 0
cut_f_signal[(W > 8.2)] = 0
# inverse fourier to get filtered frequency
cut_signal = np.fft.ifft(cut_f_signal)
# plot
plt.subplot(221)
plt.plot(time,signal)
plt.subplot(222)
plt.plot(W, f_signal)
plt.subplot(223)
plt.plot(W, cut_f_signal)
plt.subplot(224)
plt.plot(time, cut_signal)
plt.show()
The FFT of a real-valued input signal will produce a conjugate symmetric result. (That's just the way the math works best.) So, for FFT result magnitudes only of real data, the negative frequencies are just mirrored duplicates of the positive frequencies, and can thus be ignored when analyzing the result.
However if you want to do the inverse and compute the IFFT, you will need to feed the IFFT a conjugate symmetric negative half (or upper half, above Fs/2) of frequency data, or else your IFFT result will end up producing a complex result (e.g. with non-zero imaginary (sqrt(-1)) components, rarely what one want when dealing with base-band real data).
If you want to filter the FFT data and end up with real results from an IFFT, you will need to filter the positive and negative frequencies symmetrically identically to maintain the needed symmetry.
The FFT also produces a complex result, where the value and sign the components (real and imaginary) of each result bin represents the phase as well as the magnitude of the component basis vector (complex sinusoid, or real cosine plus real sine components). Any negative value just represents a phase rotation from if the same result was positive.
As #hotpaw2 already wrote in his comment above, the result of a FFT performed on a real signal in time domain generates complex values in frequency domain.
The input value f_signal of your plot command is a vector of complex values.
plt.subplot(222)
plt.plot(W, f_signal)
This results in meaningless output.
You should plot the absolute values of f_signal.
If you are interested in the phase you should plot the angle, too.
In Matlab this would look like this:
% Plot the absolute values of f_signal
plot(W, abs(f_signal));
% Plot the phase of f_signal
plot(W, (unwrap(angle(f_signal)));
Related
I'm trying to write a simple python script that recovers the amplitude and phase of a sine wave from it's fourier transformation.
I should be able to do this by calculating the magnitude, and direction of the vector defined by the real and imaginary numbers for the fourier transform, for a given frequency, i.e:
Amplitude_at_freq = sqrt(real_component_at_freq^2 + imag_component_at_freq^2)
Phase = arctan(imag_component_at_freq/real_component_at_freq)
Ref: 1 min 45 seconds into this video: https://www.youtube.com/watch?time_continue=106&v=IWQfj05i87g
I've written a simple python script using numpy's fft library to try and reproduce this, but despite writing out my derivation exactly as above, am failing to get the amplitude and phase, although I can recover the original frequency of my test sine wave correctly. This previous post Calculating amplitude from np.fft and this one Why FFT does not retrieve original amplitude when increasing signal length points to the same problem (where amplitude is off by factor of 2). Specifically the solution is to "multiply by 2 (half of spectrum is removed so energy must be preserved)," but I need clarification on what that means. Secondly there's no mention of my issue with recovering the phase change, and the amplitude is calculated differently from what I have here.
# Define amplitude, phase, frequency
_A = 4 # Amplitude
_p = 0 # Phase shift
_f = 8 # Frequency
# Construct a simple signal
t = np.linspace(0, 2*np.pi, 1024 + 1)[:-1]
g = _A * np.sin(_f * t + _p)
# Apply the fourier transform
ff = np.fft.fft(g)
# Get frequency of original signal
ff_ii = np.where(np.abs(ff) > 1.0)[0][0] # Just get one frequency, the other one is just mirrored freq at negative value
print('frequency of:', ff_ii)
# Get the complex vector at that frequency to retrieve amplitude and phase shift
yy = ff[ff_ii]
# Calculate the amplitude
T = t.shape[0] # domain of x; which we will divide height to get freq amplitude
A = np.sqrt(yy.real**2 + yy.imag**2)/T
print('amplitude of:', A)
# Calculate phase shift
phi = np.arctan(yy.imag/yy.real)
print('phase change:', phi)
However, the result I'm getting is:
>> frequency of: 8
>> amplitude of: 2.0
>> phase change: 1.5707963267948957
So the frequency is accurate, but I'm getting an amplitude of 2, when it should be 4, and phase change of pi/2, when it should be zero.
Is my math wrong, or is my understanding of numpy's fft implementation incorrect?
Fourier analyses a signal as a sum of exp(i.2.pi.f.t) terms, so it sees
A.sin(2.pi.f1.t) as:
-i.A/2.exp(i.2.pi.f1.t)+i.A/2.exp(-i.2.pi.f1.t),
which is mathematically equal. So in Fourier terms, you have both the positive frequency f1 and negative -f1 with complex values -A/2.i and A/2.i respectively. So each 'side' has only half the amplitude, but if you add them together (in the inverse Fourier transform) you get back amplitude A. This split in positive and negative frequency is where your missing factor 2 is if you only look at one (positive or negative) side of the spectrum. This is often done in practice because for real signals, the other half is trivial to derive given one.
Look into the exact mathematics Euler's formula and Fourier transform.
I have a real signal in time given by:
And I am simply trying to compute its power spectrum, which is the Fourier transform of the autocorrelation of the signal, and is also a purely real and positive quantity in this case. To do this, I simply write:
import numpy as np
from scipy.fftpack import fft, arange, rfftfreq, rfft
from pylab import *
lags1, c1, line1, b1 = acorr(((Y_DATA)), usevlines=False, normed=True, maxlags=3998, lw=2)
Power_spectrum = (fft(np.real(c1)))
freqs = np.fft.fftfreq(len(c1), dx)
plt.plot(freqs,Power_spectrum)
plt.xlabel('f (Hz)')
plt.xlim([-20000,20000])
plt.show()
But the output gives:
which has negative-valued output. Although if I simply take the absolute value of the data on the y-axis and plot it (i.e. np.abs(Power_spectrum)), then the output is:
which is exactly what I expect. Although why is this only fixed by taking the absolute value of my power spectrum? I checked my autocorrelation and plotted it—it seems to be working as expected and matches what others have computed.
Although what appears odd is the next step when I take the FFT. The FFT function outputs negative values which is contrary to the theory discussed in the link above and I don't quite understand why. Any thoughts on what is going wrong?
The power spectrum is the FFT of the autocorrelation, but that's not an efficient way to calculate it.
The autocorrelation is probably calculated with an FFT and iFFT, anyway.
The power spectrum is also just the squared magnitude of the FFT coefficients.
Do that instead so that the total work will be one FFT instead of 3.
An fft produces a complex result (real and imaginary components to represent both magnitude and phase of the spectrum). You have to take the (squared) magnitude of the complex vector to get the power spectrum.
im playing with python and scipy to understand windowing, i made a plot to see how windowing behave under FFT, but the result is not what i was specting.
the plot is:
the middle plots are pure FFT plot, here is where i get weird things.
Then i changed the trig. function to get leak, putting a 1 straight for the 300 first items of the array, the result:
the code:
sign_freq=80
sample_freq=3000
num=np.linspace(0,1,num=sample_freq)
i=0
#wave data:
sin=np.sin(2*pi*num*sign_freq)+np.sin(2*pi*num*sign_freq*2)
while i<1000:
sin[i]=1
i=i+1
#wave fft:
fft_sin=np.fft.fft(sin)
fft_freq_axis=np.fft.fftfreq(len(num),d=1/sample_freq)
#wave Linear Spectrum (Rms)
lin_spec=sqrt(2)*np.abs(np.fft.rfft(sin))/len(num)
lin_spec_freq_axis=np.fft.rfftfreq(len(num),d=1/sample_freq)
#window data:
hann=np.hanning(len(num))
#window fft:
fft_hann=np.fft.fft(hann)
#window fft Linear Spectrum:
wlin_spec=sqrt(2)*np.abs(np.fft.rfft(hann))/len(num)
#window + sin
wsin=hann*sin
#window + sin fft:
wsin_spec=sqrt(2)*np.abs(np.fft.rfft(wsin))/len(num)
wsin_spec_freq_axis=np.fft.rfftfreq(len(num),d=1/sample_freq)
fig=plt.figure()
ax1 = fig.add_subplot(431)
ax2 = fig.add_subplot(432)
ax3 = fig.add_subplot(433)
ax4 = fig.add_subplot(434)
ax5 = fig.add_subplot(435)
ax6 = fig.add_subplot(436)
ax7 = fig.add_subplot(413)
ax8 = fig.add_subplot(414)
ax1.plot(num,sin,'r')
ax2.plot(fft_freq_axis,abs(fft_sin),'r')
ax3.plot(lin_spec_freq_axis,lin_spec,'r')
ax4.plot(num,hann,'b')
ax5.plot(fft_freq_axis,fft_hann)
ax6.plot(lin_spec_freq_axis,wlin_spec)
ax7.plot(num,wsin,'c')
ax8.plot(wsin_spec_freq_axis,wsin_spec)
plt.show()
EDIT: as asked in the comments, i plotted the functions in dB scale, obtaining much clearer plots. Thanks a lot #SleuthEye !
It appears the plot which is problematic is the one generated by:
ax5.plot(fft_freq_axis,fft_hann)
resulting in the graph:
instead of the expected graph from Wikipedia.
There are a number of issues with the way the plot is constructed. The first is that this command essentially attempts to plot a complex-valued array (fft_hann). You may in fact be getting the warning ComplexWarning: Casting complex values to real discards the imaginary part as a result. To generate a graph which looks like the one from Wikipedia, you would have to take the magnitude (instead of the real part) with:
ax5.plot(fft_freq_axis,abs(fft_hann))
Then we notice that there is still a line striking through our plot. Looking at np.fft.fft's documentation:
The values in the result follow so-called “standard” order: If A = fft(a, n), then A[0] contains the zero-frequency term (the sum of the signal), which is always purely real for real inputs. Then A[1:n/2] contains the positive-frequency terms, and A[n/2+1:] contains the negative-frequency terms, in order of decreasingly negative frequency.
[...]
The routine np.fft.fftfreq(n) returns an array giving the frequencies of corresponding elements in the output.
Indeed, if we print the fft_freq_axis we can see that the result is:
[ 0. 1. 2. ..., -3. -2. -1.]
To get around this problem we simply need to swap the lower and upper parts of the arrays with np.fft.fftshift:
ax5.plot(np.fft.fftshift(fft_freq_axis),np.fft.fftshift(abs(fft_hann)))
Then you should note that the graph on Wikipedia is actually shown with amplitudes in decibels. You would then need to do the same with:
ax5.plot(np.fft.fftshift(fft_freq_axis),np.fft.fftshift(20*np.log10(abs(fft_hann))))
We should then be getting closer, but the result is not quite the same as can be seen from the following figure:
This is due to the fact that the plot on Wikipedia actually has a higher frequency resolution and captures the value of the frequency spectrum as its oscillates, whereas your plot samples the spectrum at fewer points and a lot of those points have near zero amplitudes. To resolve this problem, we need to get the frequency spectrum of the window at more frequency points.
This can be done by zero padding the input to the FFT, or more simply setting the parameter n (desired length of the output) to a value much larger than the input size:
N = 8*len(num)
fft_freq_axis=np.fft.fftfreq(N,d=1/sample_freq)
fft_hann=np.fft.fft(hann, N)
ax5.plot(np.fft.fftshift(fft_freq_axis),np.fft.fftshift(20*np.log10(abs(fft_hann))))
ax5.set_xlim([-40, 40])
ax5.set_ylim([-50, 80])
I am using the FFT function in NumPy to do some signal processing. I have array called signal
which has one data point for each hour and has a total of 576 data points. I use the following code on signal to look at its fourier transform.
t = len(signal)
ft = fft(signal,n=t)
mgft=abs(ft)
plot(mgft[0:t/2+1])
I see two peaks but I am unsure as to what the units of the x axis are i.e., how they map onto hours? Any help would be appreciated.
Given sampling rate FSample and transform blocksize N, you can calculate the frequency resolution deltaF, sampling interval deltaT, and total capture time capT using the relationships:
deltaT = 1/FSample = capT/N
deltaF = 1/capT = FSample/N
Keep in mind also that the FFT returns value from 0 to FSample, or equivalently -FSample/2 to FSample/2. In your plot, you're already dropping the -FSample/2 to 0 part. NumPy includes a helper function to calculate all this for you: fftfreq.
For your values of deltaT = 1 hour and N = 576, you get deltaF = 0.001736 cycles/hour = 0.04167 cycles/day, from -0.5 cycles/hour to 0.5 cycles/hour. So if you have a magnitude peak at, say, bin 48 (and bin 528), that corresponds to a frequency component at 48*deltaF = 0.0833 cycles/hour = 2 cycles/day.
In general, you should apply a window function to your time domain data before calculating the FFT, to reduce spectral leakage. The Hann window is almost never a bad choice. You can also use the rfft function to skip the -FSample/2, 0 part of the output. So then, your code would be:
ft = np.fft.rfft(signal*np.hanning(len(signal)))
mgft = abs(ft)
xVals = np.fft.fftfreq(len(signal), d=1.0) # in hours, or d=1.0/24 in days
plot(xVals[:len(mgft)], mgft)
Result of fft transformation doesn't map to HOURS, but to frequencies contained in your dataset. It would be beneficial to have your transformed graph so we can see where the spikes are.
You might be having spike at the beginning of the transformed buffer, since you didn't do any windowing.
In general, the dimensional units of frequency from an FFT are the same as the dimensional units of the sample rate attributed to the data fed to the FFT, for example: per meter, per radian, per second, or in your case, per hour.
The scaled units of frequency, per FFT result bin index, are N / theSampleRate, with the same dimensional units as above, where N is the length of the full FFT (you might only be plotting half of this length in the case of strictly real data).
Note that each FFT result peak bin represents a filter with a non-zero bandwidth, so you might want to add some uncertainty or error bounds to the result points you map onto frequency values. Or even use an interpolation estimation method, if needed and appropriate for the source data.
I have a 3D datacube, with two spatial dimensions and the third being a multi-band spectrum at each point of the 2D image.
H[x, y, bands]
Given a wavelength (or band number), I would like to extract the 2D image corresponding to that wavelength. This would be simply an array slice like H[:,:,bnd]. Similarly, given a spatial location (i,j) the spectrum at that location is H[i,j].
I would also like to 'smooth' the image spectrally, to counter low-light noise in the spectra. That is for band bnd, I choose a window of size wind and fit a n-degree polynomial to the spectrum in that window. With polyfit and polyval I can find the fitted spectral value at that point for band bnd.
Now, if I want the whole image of bnd from the fitted value, then I have to perform this windowed-fitting at each (i,j) of the image. I also want the 2nd-derivative image of bnd, that is, the value of the 2nd-derivative of the fitted spectrum at each point.
Running over the points, I could polyfit-polyval-polyder each of the x*y spectra. While this works, this is a point-wise operation. Is there some pytho-numponic way to do this faster?
If you do least-squares polynomial fitting to points (x+dx[i],y[i]) for a fixed set of dx and then evaluate the resulting polynomial at x, the result is a (fixed) linear combination of the y[i]. The same is true for the derivatives of the polynomial. So you just need a linear combination of the slices. Look up "Savitzky-Golay filters".
EDITED to add a brief example of how S-G filters work. I haven't checked any of the details and you should therefore not rely on it to be correct.
So, suppose you take a filter of width 5 and degree 2. That is, for each band (ignoring, for the moment, ones at the start and end) we'll take that one and the two on either side, fit a quadratic curve, and look at its value in the middle.
So, if f(x) ~= ax^2+bx+c and f(-2),f(-1),f(0),f(1),f(2) = p,q,r,s,t then we want 4a-2b+c ~= p, a-b+c ~= q, etc. Least-squares fitting means minimizing (4a-2b+c-p)^2 + (a-b+c-q)^2 + (c-r)^2 + (a+b+c-s)^2 + (4a+2b+c-t)^2, which means (taking partial derivatives w.r.t. a,b,c):
4(4a-2b+c-p)+(a-b+c-q)+(a+b+c-s)+4(4a+2b+c-t)=0
-2(4a-2b+c-p)-(a-b+c-q)+(a+b+c-s)+2(4a+2b+c-t)=0
(4a-2b+c-p)+(a-b+c-q)+(c-r)+(a+b+c-s)+(4a+2b+c-t)=0
or, simplifying,
22a+10c = 4p+q+s+4t
10b = -2p-q+s+2t
10a+5c = p+q+r+s+t
so a,b,c = p-q/2-r-s/2+t, (2(t-p)+(s-q))/10, (p+q+r+s+t)/5-(2p-q-2r-s+2t).
And of course c is the value of the fitted polynomial at 0, and therefore is the smoothed value we want. So for each spatial position, we have a vector of input spectral data, from which we compute the smoothed spectral data by multiplying by a matrix whose rows (apart from the first and last couple) look like [0 ... 0 -9/5 4/5 11/5 4/5 -9/5 0 ... 0], with the central 11/5 on the main diagonal of the matrix.
So you could do a matrix multiplication for each spatial position; but since it's the same matrix everywhere you can do it with a single call to tensordot. So if S contains the matrix I just described (er, wait, no, the transpose of the matrix I just described) and A is your 3-dimensional data cube, your spectrally-smoothed data cube would be numpy.tensordot(A,S).
This would be a good point at which to repeat my warning: I haven't checked any of the details in the few paragraphs above, which are just meant to give an indication of how it all works and why you can do the whole thing in a single linear-algebra operation.