We Keep Coding

Fast Fourier Plot in Python

I have a vibration data in time domain and want to convert it to frequency domain with fft. However the plot of the FFT only shows a big spike at zero and nothing else.
This is my vibration data: https://pastebin.com/7RK57kJW
My code:
import numpy as np
import matplotlib.pyplot as plt
t = np.arange(3000)
a1_fft= np.fft.fft(a1, axis=0)
freq = np.fft.fftfreq(t.shape[-1])
plt.plot(freq, a1_fft)
My FFT Plot:
What am I doing wrong here? I am pretty sure my data is uniform, which provoces in other cases a similar problem with fft.

The bins of the FFT correspond to the frequencies at 0, df, 2df, 3df, ..., F-2df, F-df, where df is determined by the number of bins and F is 1 cycle per bin.
Notice the zero frequency at the beginning. This is called the DC offset. It's the mean of your data. In the data that you show, the mean is ~1.32, while the amplitude of the sine wave is around 0.04. It's not surprising that you can't see a peak that's 33x smaller than the DC term.
There are some common ways to visualize the data that help you get around this. One common methods is to keep the DC offset but use a log scale, at least for the y-axis:
plt.semilogy(freq, a1_fft)
OR
plt.loglog(freq, a1_fft)
Another thing you can do is zoom in on the bottom 1/33rd or so of the plot. You can do this manually, or by adjusting the span of the displayed Y-axis:
p = np.abs(a1_fft[1:]).max() * [-1.1, 1.1]
plt.ylim(p)
If you are plotting the absolute values already, use
p = np.abs(a1_fft[1:]).max() * [-0.1, 1.1]
Another method is to remove the DC offset. A more elegant way of doing this than what #J. Schmidt suggests is to simply not display the DC term:
plt.plot(freq[1:], a1_fft[1:])
Or for the positive frequencies only:
n = freq.size
plt.plot(freq[1:n//2], a1_fft[1:n//2])
The cutoff at n // 2 is only approximate. The correct cutoff depends on whether the FFT has an even or odd number of elements. For even numbers, the middle bin actual has energy from both sides of the spectrum and often gets special treatment.

The peak at 0 is the DC-gain, which is very high since you didn't normalize your data. Also, the Fourier transform is a complex number, you should plot the absolute value and phase separately. In this code I also plotted only the positive frequencies:
import numpy as np
import matplotlib.pyplot as plt
#Import data
a1 = np.loadtxt('a1.txt')
plt.plot(a1)
#Normalize a1
a1 -= np.mean(a1)
#Your code
t = np.arange(3000)
a1_fft= np.fft.fft(a1, axis=0)
freq = np.fft.fftfreq(t.shape[-1])
#Only plot positive frequencies
plt.figure()
plt.plot(freq[freq>=0], np.abs(a1_fft)[freq>=0])

How to make a PSD plot using `np.fft.fft`?

I want to make a plot of power spectral density versus frequency for a signal using the numpy.fft.fft function. I want to do this so that I can preserve the complex information in the transform and know what I'm doing, as apposed to relying on higher-level functions provided by numpy (like the periodogram function). I'm following Mathwork's nice page about doing PSD analysis using Matlab's fft function: https://www.mathworks.com/help/matlab/ref/fft.html
In this example, I expect the PSD to peak at the frequency I used to construct the signal, which was 100 in this case. I generate the signal using 1000 time points a frequency of 100 inverse time units. I thought that the fft magnitude could be plotted against [0, nt/2] and the peaks would show up where there is the most energy in the frequency. When I did this, things went wrong. I expected my PSD to peak at 100.
How can I make a spectral density plot of frequency vs energy contained in that frequency using np.fft.fft?
Edit
to clarify, in my real problem, I only know that my characteristic frequency is much larger than my sample frequency
import matplotlib.pyplot as plt
import numpy as np
t = np.arange(1000)
sp = np.fft.fft(np.sin(100 * t * np.pi))
trange = np.linspace(0, t[-1] / 2, t.size)
plt.plot(trange, np.abs(sp) / t.size)
plt.show()
This is a sketch I made of the expected output:

What is your sample frequency? This sequence you are generating can represent a infinite number of continuous time signals according to the sample frequency.
The sample frequency needs to be at least twice the maximum signal frequency, as stated by the Sampling Theorem, so, using fs = 250Hz and using a sine of 10 seconds it becomes:
import matplotlib.pyplot as plt
import numpy as np
fs = 250
t = np.arange(0, 10, 1/fs)
sp = np.fft.fft(np.sin(2*np.pi * 100 * t))
trange = np.linspace(0, fs, len(t))
plt.plot(trange, np.abs(sp))
plt.show()
If you run this you will see a peak at 100Hz as expected.

How do you extract time-domain amplitudes from real signal with python FFT tools?

I'm working on a problem where I would like to extract and compare the time domain amplitudes of two different signals at each frequency. The signals are real world, so have noise, and multiple frequencies, so I'm trying to work in the FFT world.
I wrote a function to take the FFT of a dataset, and return the amplitudes. This seems to work okay for a simulated pure sin wave, but when performed on actual datasets, the amplitudes are always attenuated by some amount.
def amplitudePowerSpectrum(time,data):
dt = np.zeros(time.size-1,)
avgdt = np.mean(time[1:-1] - time[0:-2])
sampFreq = 1.0/(avedt)
nyquistFreq = sampFreq/2.0
FFTData = np.abs(scipy.fftpack.fft(data))
## Only care about positive frequencies
FFTData = FFTData[0:len(FFTData)/2]
## This is how we get the power spectrum in terms of time-domain amplitudes
amplitudeSpectrum = FFTData/len(FFTData)
freqsData = scipy.fftpack.fftfreq(data.size, avgdt)
freq = freqsData[0:len(freqsData)/2]
return (freq,amplitudeSpectrum,(sampFreq,nyquistFreq))
Here is a plot of a raw dataset, followed by one of the computed amplitude spectrum.As you can see, there are two specifically different frequencies, with other noise on top.
I'd expect the amplitudes in figure 2 to match the time domain amplitudes in figure 1. But they are attenuated by a pretty decent factor. The end goal is a scale factor between the input (blue) and output (red) signals at each frequency.
First, is obataining time domain amplitudes accurately possible in the Fourrier domain on real datasets? If so, what am I missing? I'm working with python numpy and scipy packages

Python - FFT leads to wrong physical meanings

I am new to Python.
I intend to do Fourier Transform to an array of discrete points, (time, acceleration), and plot the result out.
I copy and paste the sample FFT code, and modify accordingly.
Please see codes:
import numpy as np
import matplotlib.pyplot as plt
# Load the .txt file in
myData = np.loadtxt('twenty_z_up.txt')
# Extract the time and acceleration columns
time = copy(myData[:,0])
# Extract the acceleration columns
zAcc = copy(myData[:,3])
t = np.arange(10080)
sp = np.fft.fft(zAcc)
freq = np.fft.fftfreq(t.shape[-1])
plt.plot(freq, sp.real)
myData is a rectangular matrix with 10080 rows and 10 columns.
Thus, zAcc is the row3 extracted from the matrix.
In the plot drawn by Spyder, most of the harmonics concentrated around 0.
They are all extremely small.
But my data are actually the accelerations of the phone carried by a walking person (including the gravity). So I expect the most significant harmonic happens around 2Hz.
Why is the graph non-sense?
Thanks in advance!
==============UPDATES: My Graphs======================
The first time domain one:
x-axis is in millisecond.
y-axis is in m/s^2, due to earth gravity, it has a DC offset of ~10.

You do get two spikes at (approximately) 2Hz. Your sampling period is around 2.8 ms (as best as I can infer from your first plot), giving +/-2Hz the normalized frequency of +/-0.056, which is about where your spikes are. fft.fftfreq by default returns the normalized frequency (which scales the sampling period). You can set the d argument to be the sampling period, and you'll get a vector containing the actual frequency.
Your huge spike in the middle is obviously the DC offset (which you can trivially remove by subtracting the mean).

As others said, we need to see the data, post it somewhere. Just to check, try first fixing the timestep size in fftfreq, then plot this synthetic signal, and then plot your signal to see how they compare:
timestep=1./50.#Assume sampling at 50Hz. Change this accordingly.
N=10080#the number of samples
T=N*timestep
t = np.linspace(0,T,N)#needed only to generate xAcc_synthetic
freq=2.#peak a frequency at 2Hz
#generate synthetic signal at 2Hz and add some noise to it
xAcc_synthetic = sin((2*np.pi)*freq*t)+np.random.rand(N)*0.2
sp_synthetic = np.fft.fft(xAcc_synthetic)
freq = np.fft.fftfreq(t.size,d=timestep)
print max(abs(freq))==(1/timestep)/2.#simple check highest freq.
plt.plot(freq, abs(sp_synthetic))
xlabel('Hz')
Now, at the x axis equal to 2 you actually have a physical frequency of 2Hz, and you may spot the more pronounced peak you are looking for. Moreover, you may want to have a look also at yAcc and zAcc.

Clipping FFT Matrix

Audio processing is pretty new for me. And currently using Python Numpy for processing wave files. After calculating FFT matrix I am getting noisy power values for non-existent frequencies. I am interested in visualizing the data and accuracy is not a high priority. Is there a safe way to calculate the clipping value to remove these values, or should I use all FFT matrices for each sample set to come up with an average number ?
regards
Edit:
from numpy import *
import wave
import pymedia.audio.sound as sound
import time, struct
from pylab import ion, plot, draw, show
fp = wave.open("500-200f.wav", "rb")
sample_rate = fp.getframerate()
total_num_samps = fp.getnframes()
fft_length = 2048.
num_fft = (total_num_samps / fft_length ) - 2
temp = zeros((num_fft,fft_length), float)
for i in range(num_fft):
tempb = fp.readframes(fft_length);
data = struct.unpack("%dH"%(fft_length), tempb)
temp[i,:] = array(data, short)
pts = fft_length/2+1
data = (abs(fft.rfft(temp, fft_length)) / (pts))[:pts]
x_axis = arange(pts)*sample_rate*.5/pts
spec_range = pts
plot(x_axis, data[0])
show()
Here is the plot in non-logarithmic scale, for synthetic wave file containing 500hz(fading out) + 200hz sine wave created using Goldwave.

Simulated waveforms shouldn't show FFTs like your figure, so something is very wrong, and probably not with the FFT, but with the input waveform. The main problem in your plot is not the ripples, but the harmonics around 1000 Hz, and the subharmonic at 500 Hz. A simulated waveform shouldn't show any of this (for example, see my plot below).
First, you probably want to just try plotting out the raw waveform, and this will likely point to an obvious problem. Also, it seems odd to have a wave unpack to unsigned shorts, i.e. "H", and especially after this to not have a large zero-frequency component.
I was able to get a pretty close duplicate to your FFT by applying clipping to the waveform, as was suggested by both the subharmonic and higher harmonics (and Trevor). You could be introducing clipping either in the simulation or the unpacking. Either way, I bypassed this by creating the waveforms in numpy to start with.
Here's what the proper FFT should look like (i.e. basically perfect, except for the broadening of the peaks due to the windowing)
Here's one from a waveform that's been clipped (and is very similar to your FFT, from the subharmonic to the precise pattern of the three higher harmonics around 1000 Hz)
Here's the code I used to generate these
from numpy import *
from pylab import ion, plot, draw, show, xlabel, ylabel, figure
sample_rate = 20000.
times = arange(0, 10., 1./sample_rate)
wfm0 = sin(2*pi*200.*times)
wfm1 = sin(2*pi*500.*times) *(10.-times)/10.
wfm = wfm0+wfm1
# int test
#wfm *= 2**8
#wfm = wfm.astype(int16)
#wfm = wfm.astype(float)
# abs test
#wfm = abs(wfm)
# clip test
#wfm = clip(wfm, -1.2, 1.2)
fft_length = 5*2048.
total_num_samps = len(times)
num_fft = (total_num_samps / fft_length ) - 2
temp = zeros((num_fft,fft_length), float)
for i in range(num_fft):
temp[i,:] = wfm[i*fft_length:(i+1)*fft_length]
pts = fft_length/2+1
data = (abs(fft.rfft(temp, fft_length)) / (pts))[:pts]
x_axis = arange(pts)*sample_rate*.5/pts
spec_range = pts
plot(x_axis, data[2], linewidth=3)
xlabel("freq (Hz)")
ylabel('abs(FFT)')
show()

FFT's because they are windowed and sampled cause aliasing and sampling in the frequency domain as well. Filtering in the time domain is just multiplication in the frequency domain so you may want to just apply a filter which is just multiplying each frequency by a value for the function for the filter you are using. For example multiply by 1 in the passband and by zero every were else. The unexpected values are probably caused by aliasing where higher frequencies are being folded down to the ones you are seeing. The original signal needs to be band limited to half your sampling rate or you will get aliasing. Of more concern is aliasing that is distorting the area of interest because for this band of frequencies you want to know that the frequency is from the expected one.
The other thing to keep in mind is that when you grab a piece of data from a wave file you are mathmatically multiplying it by a square wave. This causes a sinx/x to be convolved with the frequency response to minimize this you can multiply the original windowed signal with something like a Hanning window.

It's worth mentioning for a 1D FFT that the first element (index [0]) contains the DC (zero-frequency) term, the elements [1:N/2] contain the positive frequencies and the elements [N/2+1:N-1] contain the negative frequencies. Since you didn't provide a code sample or additional information about the output of your FFT, I can't rule out the possibility that the "noisy power values at non-existent frequencies" aren't just the negative frequencies of your spectrum.
EDIT: Here is an example of a radix-2 FFT implemented in pure Python with a simple test routine that finds the FFT of a rectangular pulse, [1.,1.,1.,1.,0.,0.,0.,0.]. You can run the example on codepad and see that the FFT of that sequence is
[0j, Negative frequencies
(1+0.414213562373j), ^
0j, |
(1+2.41421356237j), |
(4+0j), <= DC term
(1-2.41421356237j), |
0j, v
(1-0.414213562373j)] Positive frequencies
Note that the code prints out the Fourier coefficients in order of ascending frequency, i.e. from the highest negative frequency up to DC, and then up to the highest positive frequency.

I don't know enough from your question to actually answer anything specific.
But here are a couple of things to try from my own experience writing FFTs:
Make sure you are following Nyquist rule
If you are viewing the linear output of the FFT... you will have trouble seeing your own signal and think everything is broken. Make sure you are looking at the dB of your FFT magnitude. (i.e. "plot(10*log10(abs(fft(x))))" )
Create a unitTest for your FFT() function by feeding generated data like a pure tone. Then feed the same generated data to Matlab's FFT(). Do a absolute value diff between the two output data series and make sure the max absolute value difference is something like 10^-6 (i.e. the only difference is caused by small floating point errors)
Make sure you are windowing your data
If all of those three things work, then your fft is fine. And your input data is probably the issue.
Check the input data to see if there is clipping http://www.users.globalnet.co.uk/~bunce/clip.gif
Time doamin clipping shows up as mirror images of the signal in the frequency domain at specific regular intervals with less amplitude.