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Have I applied the Fourier Transformation correctly to this Dataframe? [EXAFS X-Ray Absorption Dataframe]

I have a dataset with a signal and a 1/distance (Angstrom^-1) column.
This is the dataset (fourier.csv): https://pastebin.com/ucFekzc6
After applying these steps:
import pandas as pd
import numpy as np
from numpy.fft import fft
df = pd.read_csv (r'fourier.csv')
df.plot(x ='1/distance', y ='signal', kind = 'line')
I generated this plot:
To generate the Fast Fourier Transformation data, I used the numpy library for its fft function and I applied it like this:
df['signal_fft'] = fft(df['signal'])
df.plot(x ='1/distance', y ='signal_fft', kind = 'line')
Now the plot looks like this, with the FFT data plotted instead of the initial "signal" data:
What I hoped to generate is something like this (This signal is extremely similar to mine, yet yields a vastly different FFT picture):
Theory Signal before windowing:
Theory FFT:
As you can see, my initial plot looks somewhat similar to graphic (a), but my FFT plot doesn't look anywhere near as clear as graphic (b). I'm still using the 1/distance data for both horizontal axes, but I don't see anything wrong with it, only that it should be interpreted as distance (Angstrom) instead of 1/distance (1/Angstrom) in the FFT plot.
How should I apply FFT in order to get a result that resembles the theoretical FFT curve?
Here's another slide that shows a similar initial signal to mine and a yet again vastly different FFT:
Addendum: I have been asked to provide some additional information on this problem, so I hope this helps.
The origin of the dataset that I have linked is an XAS (X-Ray Absorption Spectroscopy) spectrum of iron oxide. Such an experimentally obtained spectrum looks similar to the one shown in the "Schematic of XAFS data processing" on the top left, i.e. absorbance [a.u.] plotted against the photon energy [eV]. Firstly I processed the spectrum (pre-edge baseline correction + post-edge normalization). Then I converted the data on the x-axis from energy E to wavenumber k (thus dimension 1/Angstrom) and cut off the signal at the L-edge jump, leaving only the signal in the post-edge EXAFS region, referred to as fine structure function χ(k). The mentioned dataset includes k^2 weighted χ(k) (to emphasize oscillations at large k). All of this is not entirely relevant as the only thing I want to do now is a Fourier transformation on this signal ( k^2 χ(k) vs. k). In theory, as we are dealing with photoelectrons and (back)scattering phenomena, the EXAFS region of the XAS spectrum can be approximated using a superposition of many sinusoidal waves such as described in this equation with f(k) being the amplitude and δ(k) the phase shift of the scattered wave.
The aim is to gain an understanding of the chemical environment and the coordination spheres around the absorbing atom. The goal of the Fourier transform is to obtain some sort of signal in dependence of the "radius" R [Angstrom], which could later on be correlated to e.g. an oxygen being in ~2 Angstrom distance to the Mn atom (see "Schematic of XAFS data processing" on the right).
I only want to be able to reproduce the theoretically expected output after the FFT. My main concern is to get rid of the weird output signal and produce something that in some way resembles a curve with somewhat distinct local maxima (as shown in the 4th picture).

I don't have a 100% solution for you, but here's part of the problem.
The fft function you're using assumes that your X values are equally spaced. I checked this assumption by taking the difference between each 1/distance value, and graphing it:
df['1/distance'].diff().plot()
(Y is the difference, X is the index in the dataframe.)
This is supposed to be a constant line.
In order to fix this, one solution is to resample the signal through linear interpolation so that the timestep is constant.
from scipy import interpolate
rs_df = df.drop_duplicates().copy() # Needed because 0 is present twice in dataset
x = rs_df['1/distance']
y = rs_df['signal']
flinear = interpolate.interp1d(x, y, kind='linear')
xnew = np.linspace(np.min(x), np.max(x), rs_df.index.size)
ylinear = flinear(xnew)
rs_df['signal'] = ylinear
rs_df['1/distance'] = xnew
df.plot(x ='1/distance', y ='signal', kind = 'line')
rs_df.plot(x ='1/distance', y ='signal', kind = 'line')
The new line looks visually identical, but has a constant timestep.
I still don't get your intended result from the FFT, so this is only a partial solution.

MCVE
We import required dependencies:
import numpy as np
import pandas as pd
from scipy import signal
import matplotlib.pyplot as plt
And we load your dataset:
raw = pd.read_csv("https://pastebin.com/raw/ucFekzc6", sep="\t",
names=["k", "wchi"], header=0)
We clean the dataset a bit as it contains duplicates and a problematic point with null wave number (or infinite distance) and ensure a zero mean signal:
raw = raw.drop_duplicates()
raw = raw.iloc[1:, :]
raw["wchi"] = raw["wchi"] - raw["wchi"].mean()
The signal is about:
As noticed by #NickODell, signal is not equally sampled which is a problem if you aim to perform FFT signal processing.
We can resample your signal to have equally spaced sampling:
N = 65536
k = np.linspace(raw["k"].min(), raw["k"].max(), N)
interpolant = interpolate.interp1d(raw["k"], raw["wchi"], kind="linear")
g = interpolant(k)
Notice for performance concerns FFT does split the signal with the null frequency component at the borders (that's why your FFT signal does not look as it is usually presented in books). This indeed can be corrected by using classic fftshift method or performing ad hoc indexing.
R = 2*np.pi*fft.fftfreq(N, np.diff(k)[0])[:N//2]
G = (1/N)*fft.fft(g)[0:N//2]
Mind the 2π factor which is involved in the units scaling of your transformation.
You also have mentioned a windowing (at least in a picture) that is not referenced anywhere. This kind of filtering may help a lot when performing signal processing as it filter out artifacts and unwanted noise. I leave it up to you.
Least Square Spectral Analysis
An alternative to process your signal is available since the advent of modern Linear Algebra. There is a way to estimate the periodogram of an irregular sampled signal by a method called Least Square Spectral Analysis.
You are looking for the square root of the periodogram of your signal and scipy does implement an easy way to compute it by the Lomb-Scargle method.
To do so, we simply create a frequency vector (in this case they are desired output distances) and perform the regression for each of those distances w.r.t. your signal:
Rhat = np.linspace(raw["R"].min(), raw["R"].max()*2, 5000)
Ghat = signal.lombscargle(raw["k"], raw["wchi"], freqs=Rhat, normalize=True)
Graphically it leads to:
Comparison
If we compare both methodology we can confirm that the major peaks definitely match.
LSSA gives a smoother curve but do not assume it to be more accurate as this is statistical smooth of an interpolated curve. Anyway it fit the bill for your requirement:
I only want to be able to reproduce the theoretically expected output
after the FFT. My main concern is to get rid of the weird output
signal and produce something that in some way resembles a curve with
somewhat distinct local maxima (as shown in the 4th picture).
Conclusions
I think you have enough information to process your signal either by resampling and using FFT or by using LSSA. Both method has advantages and drawbacks.
Of course this needs to be validated with well know cases. Why not to reproduce with the data of the experience of the paper you are working on to check out you can reconstruct figures you posted.
You also need to dig in the signal conditioning before performing post processing (resampling, windowing, filtering).

Area under the peak of a FFT in Python

I'm trying to do some tests before I proceed analyzing some real dataset via FFT, and I've found the following problem.
First, I create a signal as the sum of two cosines and then use rfft to to the transformation (since it has only real values):
import numpy as np
import matplotlib.pyplot as plt
from scipy.fft import rfft, rfftfreq
# Number of sample points
N = 800
# Sample spacing
T = 1.0 / 800.0
x = np.linspace(0.0, N*T, N)
y = 0.5*np.cos(10*2*np.pi*x) + 0.5*np.cos(200*2*np.pi*x)
# FFT
yf = rfft(y)
xf = rfftfreq(N, T)
fig, ax = plt.subplots(1,2,figsize=(15,5))
ax[0].plot(x,y)
ax[1].plot(xf, 2.0/N*np.abs(yf))
As it can be seen from the definition of the signal, I have two oscillations with amplitude 0.5 and frequency 10 and 200. Now, I would expect the FFT spectrum to be something like two deltas at those points, but apparently increasing the frequency broadens the peaks:
From the first peak it can be infered that the amplitude is 0.5, but not for the second. I've tryied to obtain the area under the peak using np.trapz and use that as an estimate for the amplitude, but as it is close to a dirac delta it's very sensitive to the interval I choose. My problem is that I need to get the amplitude as exact as possible for my data analysis.
EDIT: As it seems to be something related with the number of points, I decided to increment (now that I can) the sample frequency. This seems to solve the problem, as it can be seen in the figure:
However, it still seems strange that for a certain number of points and sample frequency, the high frequency peaks broaden...

It is not strange , you have leakage of the frequency bins. When you discretize the signal (sampling) needed for the Fourier transfrom , frequency bins are created which are frequency intervals where the the amplitude is calculated. And each bin has wide which is given by the sample_rate / num_points . So , the less the number of bins the more difficult is to assign precise amplitudes to every frequency. Other problems in choosing the best sampling rate exist such as the shannon-nyquist theorem to prevent aliasing. https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem . But depending on the problem sometimes there some custom rates used for sampling. E.g. when dealing with audio a sampling rate of 44,100 Hz is widely used , cause is based on the limits of the human hearing. So it depends also on nature of the data you want to perform analysis as you wrote. Anyway , since this question has also theoretical value , you can also check https://dsp.stackexchange.com for some useful info.

I would comment to George's answer, but yet I cannot.
Maybe a starting point for your research are the properties of the Discrete Fourier Transform.
The signal in the time domain is actual the cosines multiplied by a box window which transforms into the frequency domain as the convolution of the deltas with the sinc function. The sinc functions will smear the spectrum.
However, I am not sure we are observing spectral leakage here, since the window fits exactly to the full period of cosines. The discretization of the bins might still play a role here.

How to interpret this fft graph

I want to apply Fourier transformation using fft function to my time series data to find "patterns" by extracting the dominant frequency components in the observed data, ie. the lowest 5 dominant frequencies to predict the y value (bacteria count) at the end of each time series.
I would like to preserve the smallest 5 coefficients as features, and eliminate the rest.
My code is as below:
df = pd.read_csv('/content/drive/My Drive/df.csv', sep=',')
X = df.iloc[0:2,0:10000]
dft_X = np.fft.fft(X)
print(dft_X)
print(len(dft_X))
plt.plot(dft_X)
plt.grid(True)
plt.show()
# What is the graph about(freq/amplitude)? How much data did it use?
for i in dft_X:
m = i[np.argpartition(i,5)[:5]]
n = i[np.argpartition(i,range(5))[:5]]
print(m,'\n',n)
Here is the output:
But I am not sure how to interpret this graph. To be precise,
1) Does the graph show the transformed values of the input data? I only used 2 rows of data(each row is a time series), thus data is 2x10000, why are there so many lines in the graph?
2) To obtain frequency value, should I use np.fft.fftfreq(n, d=timestep)?
Parameters:
n : int
Window length.
d : scalar, optional
Sample spacing (inverse of the sampling rate). Defaults to 1.
Returns:
f : ndarray
Array of length n containing the sample frequencies.
How to determine n(window length) and sample spacing?
3) Why are transformed values all complex numbers?
Thanks

I'm gonna answer in reverse order of your questions
3) Why are transformed values all complex numbers?
The output of a Fourier Transform is always complex numbers. To get around this fact, you can either apply the absolute value on the output of the transform, or only plot the real part using:
plt.plot(dft_X.real)
2) To obtain frequency value, should I use np.fft.fftfreq(n, d=timestep)?
No, the "frequency values" will be visible on the output of the FFT.
1) Does the graph show the transformed values of the input data? I only used 2 rows of data(each row is a time series), thus data is 2x10000, why are there so many lines in the graph?
Your graph has so many lines because it's making a line for each column of your data set. Apply the FFT on each row separately (or possibly just transpose your dataframe) and then you'll get more actual frequency domain plots.
Follow up
Would using absolute value or real part of the output as features for a later model have different effect than using the original output?
Absolute values are easier to work with usually.
Using real part
Using absolute value
Here's the Octave code that generated this:
Fs = 4000; % Sampling rate of signal
T = 1/Fs; % Period
L = 4000; % Length of signal
t = (0:L-1)*T; % Time axis
freq = 1000; % Frequency of our sinousoid
sig = sin(freq*2*pi*t); % Fill Time-Domain with 1000 Hz sinusoid
f_sig = fft(sig); % Apply FFT
f = Fs*(0:(L/2))/L; % Frequency axis
figure
plot(f,abs(f_sig/L)(1:end/2+1)); % peak at 1kHz)
figure
plot(f,real(f_sig/L)(1:end/2+1)); % main peak at 1kHz)
In my example, you can see the absolute value returned no noise at frequencies other than the sinusoid of frequency 1kHz I generated while the real part had a bigger peak at 1kHz but also had much more noise.
As for effects, I don't know what you mean by that.
is it expected that "frequency values" always be complex numbers
Always? No. The Fourier series represents the frequency coefficients at which the sum of sines and cosines completely equate any continuous periodic function. Sines and cosines can be written in complex forms through Euler's formula. This is the most convenient way to store Fourier coefficients. In truth, the imaginary part of your frequency-domain signal represents the phase of the signal. (i.e if I have 2 sine functions of the same frequency, they can have different complex forms depending on the time shifting). However, most libraries that provide an FFT function will, by default, store FFT coefficients as complex numbers, to facilitate phase and magnitude calculations.
Is it convention that FFT use each column of dataset when plotting a line
I think it is an issue with mathplotlib.plot, not np.fft.
Could you please show me how to apply FFT on each row separately
There are many ways to go around this and I don't want to force you down one path, so I will propose the general solution to iterate over each row of your dataframe and apply the FFT on each specific row. Otherwise, in your case, I believe transposing your output could also work.

How do I get the frequencies from a signal?

I am look for a way to obtain the frequency from a signal. Here's an example:
signal = [numpy.sin(numpy.pi * x / 2) for x in range(1000)]
This Array will represent the sample of a recorded sound (x = miliseconds)
sin(pi*x/2) => 250 Hrz
How can we go from the signal (list of points), to obtaining the frequencies form this array?
Note:
I have read many Stackoverflow threads and watch many youtube videos. I am yet to find an answer. Please use simple words.
(I am Thankfull for every answer)

What you're looking for is known as the Fourier Transform
A bit of background
Let's start with the formal definition:
The Fourier transform (FT) decomposes a function (often a function of time, or a signal) into its constituent frequencies
This is in essence a mathematical operation that when applied over a signal, gives you an idea of how present each frequency is in the time series. In order to get some intuition behind this, it might be helpful to look at the mathematical definition of the DFT:
Where k here is swept all the way up t N-1 to calculate all the DFT coefficients.
The first thing to notice is that, this definition resembles somewhat that of the correlation of two functions, in this case x(n) and the negative exponential function. While this may seem a little bit abstract, by using Euler's formula and by playing a bit around with the definition, the DFT can be expressed as the correlation with both a sine wave and a cosine wave, which will account for the imaginary and the real parts of the DFT.
So keeping in mind that this is in essence computing a correlation, whenever a corresponding sine or cosine from the decomposition of the complex exponential matches with that of x(n), there will be a peak in X(K), meaning that, such frequency is present in the signal.
How can we do the same with numpy?
So having given a very brief theoretical background, let's consider an example to see how this can be implemented in python. Lets consider the following signal:
import numpy as np
import matplotlib.pyplot as plt
Fs = 150.0; # sampling rate
Ts = 1.0/Fs; # sampling interval
t = np.arange(0,1,Ts) # time vector
ff = 50; # frequency of the signal
y = np.sin(2*np.pi*ff*t)
plt.plot(t, y)
plt.xlabel('Time')
plt.ylabel('Amplitude')
plt.show()
Now, the DFT can be computed by using np.fft.fft, which as mentioned, will be telling you which is the contribution of each frequency in the signal now in the transformed domain:
n = len(y) # length of the signal
k = np.arange(n)
T = n/Fs
frq = k/T # two sides frequency range
frq = frq[:len(frq)//2] # one side frequency range
Y = np.fft.fft(y)/n # dft and normalization
Y = Y[:n//2]
Now, if we plot the actual spectrum, you will see that we get a peak at the frequency of 50Hz, which in mathematical terms it will be a delta function centred in the fundamental frequency of 50Hz. This can be checked in the following Table of Fourier Transform Pairs table.
So for the above signal, we would get:
plt.plot(frq,abs(Y)) # plotting the spectrum
plt.xlabel('Freq (Hz)')
plt.ylabel('|Y(freq)|')
plt.show()

extracting phase information using numpy fft

I am trying to use a fast fourier transform to extract the phase shift of a single sinusoidal function. I know that on paper, If we denote the transform of our function as T, then we have the following relations:
However, I am finding that while I am able to accurately capture the frequency of my cosine wave, the phase is inaccurate unless I sample at an extremely high rate. For example:
import numpy as np
import pylab as pl
num_t = 100000
t = np.linspace(0,1,num_t)
dt = 1.0/num_t
w = 2.0*np.pi*30.0
phase = np.pi/2.0
amp = np.fft.rfft(np.cos(w*t+phase))
freqs = np.fft.rfftfreq(t.shape[-1],dt)
print (np.arctan2(amp.imag,amp.real))[30]
pl.subplot(211)
pl.plot(freqs[:60],np.sqrt(amp.real**2+amp.imag**2)[:60])
pl.subplot(212)
pl.plot(freqs[:60],(np.arctan2(amp.imag,amp.real))[:60])
pl.show()
Using num=100000 points I get a phase of 1.57173880459.
Using num=10000 points I get a phase of 1.58022110476.
Using num=1000 points I get a phase of 1.6650441064.
What's going wrong? Even with 1000 points I have 33 points per cycle, which should be enough to resolve it. Is there maybe a way to increase the number of computed frequency points? Is there any way to do this with a "low" number of points?
EDIT: from further experimentation it seems that I need ~1000 points per cycle in order to accurately extract a phase. Why?!
EDIT 2: further experiments indicate that accuracy is related to number of points per cycle, rather than absolute numbers. Increasing the number of sampled points per cycle makes phase more accurate, but if both signal frequency and number of sampled points are increased by the same factor, the accuracy stays the same.

Your points are not distributed equally over the interval, you have the point at the end doubled: 0 is the same point as 1. This gets less important the more points you take, obviusly, but still gives some error. You can avoid it totally, the linspace has a flag for this. Also it has a flag to return you the dt directly along with the array.
Do
t, dt = np.linspace(0, 1, num_t, endpoint=False, retstep=True)
instead of
t = np.linspace(0,1,num_t)
dt = 1.0/num_t
then it works :)

The phase value in the result bin of an unrotated FFT is only correct if the input signal is exactly integer periodic within the FFT length. Your test signal is not, thus the FFT measures something partially related to the phase difference of the signal discontinuity between end-points of the test sinusoid. A higher sample rate will create a slightly different last end-point from the sinusoid, and thus a possibly smaller discontinuity.
If you want to decrease this FFT phase measurement error, create your test signal so the your test phase is referenced to the exact center (sample N/2) of the test vector (not the 1st sample), and then do an fftshift operation (rotate by N/2) so that there will be no signal discontinuity between the 1st and last point in your resulting FFT input vector of length N.

This snippet of code might help:
def reconstruct_ifft(data):
"""
In this function, we take in a signal, find its fft, retain the dominant modes and reconstruct the signal from that
Parameters
----------
data : Signal to do the fft, ifft
Returns
-------
reconstructed_signal : the reconstructed signal
"""
N = data.size
yf = rfft(data)
amp_yf = np.abs(yf) #amplitude
yf = yf*(amp_yf>(THRESHOLD*np.amax(amp_yf)))
reconstructed_signal = irfft(yf)
return reconstructed_signal
The 0.01 is the threshold of amplitudes of the fft that you would want to retain. Making the THRESHOLD greater(more than 1 does not make any sense), will give
fewer modes and cause higher rms error but ensures higher frequency selectivity.
(Please adjust the TABS for the python code)