I am looking for an efficient way to detect plateaus in otherwise very noisy data. The plateaus are always relatively broad A simple example of what this data could look like:
test=np.random.uniform(0.9,1,100)
test[10:20]=0
plt.plot(test)
Note that there can be multiple plateaus (which should all be detected) which can have different values.
I've tried using scipy.signal.argrelextrema, but it doesn't seem to be doing what I want it to:
peaks=argrelextrema(test,np.less,order=25)
plt.vlines(peaks,ymin=0, ymax=1)
I don't need the exact interval of the plateau- a rough range estimate would be enough, as long as that estimate is bigger or equal than the actual plateau range. It should be relatively efficient however.
There is a method scipy.signal.find_peaks that you can try, here is an exmple
import numpy
from scipy.signal import find_peaks
test = numpy.random.uniform(0.9, 1.0, 100)
test[10 : 20] = 0
peaks, peak_plateaus = find_peaks(- test, plateau_size = 1)
although find_peaks only finds peaks, it can be used to find valleys if the array is negated, then you do the following
for i in range(len(peak_plateaus['plateau_sizes'])):
if peak_plateaus['plateau_sizes'][i] > 1:
print('a plateau of size %d is found' % peak_plateaus['plateau_sizes'][i])
print('its left index is %d and right index is %d' % (peak_plateaus['left_edges'][i], peak_plateaus['right_edges'][i]))
it will print
a plateau of size 10 is found
its left index is 10 and right index is 19
This is really just a "dumb" machine learning task. You'll want to code a custom function to screen for them. You have two key characteristics to a plateau:
They're consecutive occurrences of the same value (or very nearly so).
The first and last points deviate strongly from a forward and backward moving average, respectively. (Try quantifying this based on the standard deviation if you expect additive noise, for geometric noise you'll have to take the magnitude of your signal into account too.)
A simple loop should then be sufficient to calculate a forward moving average, stdev of points in that forward moving average, reverse moving average, and stdev of points in that reverse moving average.
Read until you find a point well outside the regular noise (compare to variance). Start buffering those indices into a list.
Keep reading and buffering indices into that list while they have the same value (or nearly the same, if your plateaus can be a little rough; you'll want to use some tolerance plus the standard deviation of your plateaus, or just some tolerance if you expect them all to behave similarly).
If the variance of the points in your buffer gets too high, it's not a plateau, too rough; throw it out and start scanning again from your current position.
If the last value was very different from the previous (on the order of the change that triggered your code to start buffering indices) and in the opposite direction of the original impulse, cap your buffer here; you've got a plateau there.
Now do whatever you want with the points at those indices. Delete them, replace them with a linear interpolation between the two boundary points, whatever.
I could generate some noise and give you some sample code, but this is really something you're going to have to adapt to your application. (For example, there's a shortcoming in this method that a plateau which captures a point on the middle of the "cliff edge" may leave that point when it removes the rest of the plateau. If that's something you're worried about, you'll have to do a little more exploring after you ID the plateau.) You should be able to do this in a single pass over the data, but it might be wise to get some statistics on the whole set first to intelligently tweak your thresholds.
If you have an exact definition of what constitutes a plateau, you can make this a lot less hand-wavey and ML-looking, but so long as you're trying to identify fuzzy pattern, you're gonna have to take a statistics-based approach.
I had a similar problem, and found a simple heuristic solution shared below. I find plateaus as ranges of constant gradient of the signal. You could change the code to also check that the gradient is (close to) 0.
I apply a moving average (uniform_filter_1d) to filter out noise. Also, I calculate the first and second derivative of the signal numerically, so I'm not sure it matches the requirement of efficiency. But it worked perfectly for my signal and might be a good starting point for others.
def find_plateaus(F, min_length=200, tolerance = 0.75, smoothing=25):
'''
Finds plateaus of signal using second derivative of F.
Parameters
----------
F : Signal.
min_length: Minimum length of plateau.
tolerance: Number between 0 and 1 indicating how tolerant
the requirement of constant slope of the plateau is.
smoothing: Size of uniform filter 1D applied to F and its derivatives.
Returns
-------
plateaus: array of plateau left and right edges pairs
dF: (smoothed) derivative of F
d2F: (smoothed) Second Derivative of F
'''
import numpy as np
from scipy.ndimage.filters import uniform_filter1d
# calculate smooth gradients
smoothF = uniform_filter1d(F, size = smoothing)
dF = uniform_filter1d(np.gradient(smoothF),size = smoothing)
d2F = uniform_filter1d(np.gradient(dF),size = smoothing)
def zero_runs(x):
'''
Helper function for finding sequences of 0s in a signal
https://stackoverflow.com/questions/24885092/finding-the-consecutive-zeros-in-a-numpy-array/24892274#24892274
'''
iszero = np.concatenate(([0], np.equal(x, 0).view(np.int8), [0]))
absdiff = np.abs(np.diff(iszero))
ranges = np.where(absdiff == 1)[0].reshape(-1, 2)
return ranges
# Find ranges where second derivative is zero
# Values under eps are assumed to be zero.
eps = np.quantile(abs(d2F),tolerance)
smalld2F = (abs(d2F) <= eps)
# Find repititions in the mask "smalld2F" (i.e. ranges where d2F is constantly zero)
p = zero_runs(np.diff(smalld2F))
# np.diff(p) gives the length of each range found.
# only accept plateaus of min_length
plateaus = p[(np.diff(p) > min_length).flatten()]
return (plateaus, dF, d2F)
Related
I have a dataframe where each column represents a geographic point, and each row represents a minute in a day. The value of each cell is the flow of water at that point in CFS. Below is a graph of one of these time-flow series.
Basically, I need to calculate the absolute value of the max flow at each of these locations during the day, which in this case would be that hump of 187 cfs. However, there are instabilities, so DF.abs().max() returns 1197 cfs. I need to somehow remove the outliers in the calculation. As you can see, there is no pattern to the outliers, but if you look at the graph, no 2 consecutive points in time should have more than an x% change in flow. I should mention that there are 15K of these points, so the fastest solution is the best.
Anyone know how can I accomplish this in python, or at least know the statistical word for what I want to do? Thanks!
In my opinion, the statistical word your are looking for is smoothing or denoising data.
Here is my try:
# Importing packages
import numpy as np
import matplotlib.pyplot as plt
from scipy.signal import savgol_filter
# Creating a curve with a local maximum to simulate "ideal data"
x = np.arange(start=-1, stop=1, step=0.001)
y_ideal = 10**-(x**2)
# Adding some randomly distributed outliers to simulate "real data"
y_real = y_ideal.copy()
np.random.seed(0)
for i in range(50):
x_index = np.random.choice(len(x))
y_real[x_index] = np.random.randint(-3, 5)
# Denoising with Savitzky-Golay (window size = 501, polynomial order = 3)
y_denoised = savgol_filter(y_real, window_length=501, polyorder=3)
# You should optimize these values to fit your needs
# Getting the index of the maximum value from the "denoised data"
max_index = np.where(y_denoised == np.amax(y_denoised))[0]
# Recovering the maximum value and reporting
max_value = y_real[max_index][0]
print(f'The maximum value is around {max_value:.5f}')
Please, keep in mind that:
This solution is approximate.
You should find the optimum parameters of the window_length and polyorder parameters plugged to the savgol_filter() function.
If the region where your maximum is located is noisy, you can use max_value = y_denoised [max_index][0] instead of max_value = y_real[max_index][0].
Note: This solution is based in this other Stack Overflow answer
I'd like to plot (using matplotlib.pyplot) a probability density function (pdf), but hide their left and/or right tail whenever these are fairly close to zero.
E.g., the normal distribution when being apart some multiples of the standard deviation from the mean value.
The pdf is stored in two arrays samplingPts and functionVals,
containing the equidistant sampling point coordinates and the values of the function, respectively.
Both arrays are of type numpy.ndarray and have identical length.
Until now I use a quick and dirty hack by just cutting down the arrays:
# Define shortened arrays by dropping indices whose
# corresponding value.__abs__() is below a given threshold
threshold = 0.005
samplingPts_shortened = samplingPts[scipy.absolute(functionVals) > threshold]
functionVals_shortened = functionVals[scipy.absolute(functionVals) > threshold]
Very dirty indeed, it cannot be the final clue because the pdf possibly has two or more humps and may be close to zero in between, whence the "in-between sampling points" were eliminated as well. But they should remain and be present in the plot.
In addition it is not at all memory saving.
So my question is how to implement sound code which,
given two arrays as of above representing the function, cuts down these arrays at both ends until the function values begin to notably emerge from zero?
Why not look from the beginning of the samplingPts for when functionVals increases above the threshold, and cut it off there. Then look from the end of samplingPts backwards for when functionVals increases above the threshold and cut it off there too?
Something like:
for i in range(len(samplingPts)):
if scipy.absolute(functionVals[i]) > threshold:
break
samplingPts = samplingPts[i:]
functionVals = functionVals[i:]
for i in range(len(samplingPts)-1, 0, -1)):
if scipy.absolute(functionVals[i]) > threshold:
break
samplingPts = samplingPts[:i+1]
functionVals = functionVals[:i+1]
I have a set of measured radii (t+epsilon+error) at an equally spaced angles.
The model is circle of radius (R) with center at (r, Alpha) with added small noise and some random error values which are much bigger than noise.
The problem is to find the center of the circle model (r,Alpha) and the radius of the circle (R). But it should not be too much sensitive to random error (in below data points at 7 and 14).
Some radii could be missing therefore the simple mean would not work here.
I tried least square optimization but it significantly reacts on error.
Is there a way to optimize least deltas but not the least squares of delta in Python?
Model:
n=36
R=100
r=10
Alpha=2*Pi/6
Data points:
[95.85, 92.66, 94.14, 90.56, 88.08, 87.63, 88.12, 152.92, 90.75, 90.73, 93.93, 92.66, 92.67, 97.24, 65.40, 97.67, 103.66, 104.43, 105.25, 106.17, 105.01, 108.52, 109.33, 108.17, 107.10, 106.93, 111.25, 109.99, 107.23, 107.18, 108.30, 101.81, 99.47, 97.97, 96.05, 95.29]
It seems like your main problem here is going to be removing outliers. There are a couple of ways to do this, but for your application, your best bet is to probably just to remove items based on their distance from the median (Since the median is much less sensitive to outliers than the mean.)
If you're using numpy that would looks like this:
def remove_outliers(data_points, margin=1.5):
nd = np.abs(data_points - np.median(data_points))
s = nd/np.median(nd)
return data_points[s<margin]
After which you should run least squares.
If you're not using numpy you can do something similar with native python lists:
def median(points):
return sorted(points)[len(points)/2] # evaluates to an int in python2
def remove_outliers(data_points, margin=1.5):
m = median(data_points)
centered_points = [abs(point - m) for point in data_points]
centered_median = median(centered_points)
ratios = [datum/centered_median for datum in centered_points]
return [point for i, point in enumerate(data_points) if ratios[i]>margin]
If you're looking to just not count outliers as highly you can just calculate the mean of your dataset, which is just a linear equivalent of the least-squares optimization.
If you're looking for something a little better I might suggest throwing your data through some kind of low pass filter, but I don't think that's really needed here.
A low-pass filter would probably be the best, which you can do as follows: (Note, alpha is a number you will have to fiddle with to get your desired output.)
def low_pass(data, alpha):
new_data = [data[0]]
for i in range(1, len(data)):
new_data.append(alpha * data[i] + (1 - alpha) * new_data[i-1])
return new_data
At which point your least squares optimization should work fine.
Replying to your final question
Is there a way to optimize least deltas but not the least squares of delta in Python?
Yes, pick an optimization method (for example downhill simplex implemented in scipy.optimize.fmin) and use the sum of absolute deviations as a merit function. Your dataset is small, I suppose that any general purpose optimization method will converge quickly. (In case of non-linear least squares fitting it is also possible to use general purpose optimization algorithm, but it's more common to use the Levenberg-Marquardt algorithm which minimizes sums of squares.)
If you are interested when minimizing absolute deviations instead of squares has theoretical justification see Numerical Recipes, chapter Robust Estimation.
From practical side, the sum of absolute deviations may not have unique minimum.
In the trivial case of two points, say, (0,5) and (1,9) and constant function y=a, any value of a between 5 and 9 gives the same sum (4). There is no such problem when deviations are squared.
If minimizing absolute deviations would not work, you may consider heuristic procedure to identify and remove outliers. Such as RANSAC or ROUT.
Operators used to examine the spectrum, knowing the location and width of each peak and judge the piece the spectrum belongs to. In the new way, the image is captured by a camera to a screen. And the width of each band must be computed programatically.
Old system: spectroscope -> human eye
New system: spectroscope -> camera -> program
What is a good method to compute the width of each band, given their approximate X-axis positions; given that this task used to be performed perfectly by eye, and must now be performed by program?
Sorry if I am short of details, but they are scarce.
Program listing that generated the previous graph; I hope it is relevant:
import Image
from scipy import *
from scipy.optimize import leastsq
# Load the picture with PIL, process if needed
pic = asarray(Image.open("spectrum.jpg"))
# Average the pixel values along vertical axis
pic_avg = pic.mean(axis=2)
projection = pic_avg.sum(axis=0)
# Set the min value to zero for a nice fit
projection /= projection.mean()
projection -= projection.min()
#print projection
# Fit function, two gaussians, adjust as needed
def fitfunc(p,x):
return p[0]*exp(-(x-p[1])**2/(2.0*p[2]**2)) + \
p[3]*exp(-(x-p[4])**2/(2.0*p[5]**2))
errfunc = lambda p, x, y: fitfunc(p,x)-y
# Use scipy to fit, p0 is inital guess
p0 = array([0,20,1,0,75,10])
X = xrange(len(projection))
p1, success = leastsq(errfunc, p0, args=(X,projection))
Y = fitfunc(p1,X)
# Output the result
print "Mean values at: ", p1[1], p1[4]
# Plot the result
from pylab import *
#subplot(211)
#imshow(pic)
#subplot(223)
#plot(projection)
#subplot(224)
#plot(X,Y,'r',lw=5)
#show()
subplot(311)
imshow(pic)
subplot(312)
plot(projection)
subplot(313)
plot(X,Y,'r',lw=5)
show()
Given an approximate starting point, you could use a simple algorithm that finds a local maxima closest to this point. Your fitting code may be doing that already (I wasn't sure whether you were using it successfully or not).
Here's some code that demonstrates simple peak finding from a user-given starting point:
#!/usr/bin/env python
from __future__ import division
import numpy as np
from matplotlib import pyplot as plt
# Sample data with two peaks: small one at t=0.4, large one at t=0.8
ts = np.arange(0, 1, 0.01)
xs = np.exp(-((ts-0.4)/0.1)**2) + 2*np.exp(-((ts-0.8)/0.1)**2)
# Say we have an approximate starting point of 0.35
start_point = 0.35
# Nearest index in "ts" to this starting point is...
start_index = np.argmin(np.abs(ts - start_point))
# Find the local maxima in our data by looking for a sign change in
# the first difference
# From http://stackoverflow.com/a/9667121/188535
maxes = (np.diff(np.sign(np.diff(xs))) < 0).nonzero()[0] + 1
# Find which of these peaks is closest to our starting point
index_of_peak = maxes[np.argmin(np.abs(maxes - start_index))]
print "Peak centre at: %.3f" % ts[index_of_peak]
# Quick plot showing the results: blue line is data, green dot is
# starting point, red dot is peak location
plt.plot(ts, xs, '-b')
plt.plot(ts[start_index], xs[start_index], 'og')
plt.plot(ts[index_of_peak], xs[index_of_peak], 'or')
plt.show()
This method will only work if the ascent up the peak is perfectly smooth from your starting point. If this needs to be more resilient to noise, I have not used it, but PyDSTool seems like it might help. This SciPy post details how to use it for detecting 1D peaks in a noisy data set.
So assume at this point you've found the centre of the peak. Now for the width: there are several methods you could use, but the easiest is probably the "full width at half maximum" (FWHM). Again, this is simple and therefore fragile. It will break for close double-peaks, or for noisy data.
The FWHM is exactly what its name suggests: you find the width of the peak were it's halfway to the maximum. Here's some code that does that (it just continues on from above):
# FWHM...
half_max = xs[index_of_peak]/2
# This finds where in the data we cross over the halfway point to our peak. Note
# that this is global, so we need an extra step to refine these results to find
# the closest crossovers to our peak.
# Same sign-change-in-first-diff technique as above
hm_left_indices = (np.diff(np.sign(np.diff(np.abs(xs[:index_of_peak] - half_max)))) > 0).nonzero()[0] + 1
# Add "index_of_peak" to result because we cut off the left side of the data!
hm_right_indices = (np.diff(np.sign(np.diff(np.abs(xs[index_of_peak:] - half_max)))) > 0).nonzero()[0] + 1 + index_of_peak
# Find closest half-max index to peak
hm_left_index = hm_left_indices[np.argmin(np.abs(hm_left_indices - index_of_peak))]
hm_right_index = hm_right_indices[np.argmin(np.abs(hm_right_indices - index_of_peak))]
# And the width is...
fwhm = ts[hm_right_index] - ts[hm_left_index]
print "Width: %.3f" % fwhm
# Plot to illustrate FWHM: blue line is data, red circle is peak, red line
# shows FWHM
plt.plot(ts, xs, '-b')
plt.plot(ts[index_of_peak], xs[index_of_peak], 'or')
plt.plot(
[ts[hm_left_index], ts[hm_right_index]],
[xs[hm_left_index], xs[hm_right_index]], '-r')
plt.show()
It doesn't have to be the full width at half maximum — as one commenter points out, you can try to figure out where your operators' normal threshold for peak detection is, and turn that into an algorithm for this step of the process.
A more robust way might be to fit a Gaussian curve (or your own model) to a subset of the data centred around the peak — say, from a local minima on one side to a local minima on the other — and use one of the parameters of that curve (eg. sigma) to calculate the width.
I realise this is a lot of code, but I've deliberately avoided factoring out the index-finding functions to "show my working" a bit more, and of course the plotting functions are there just to demonstrate.
Hopefully this gives you at least a good starting point to come up with something more suitable to your particular set.
Late to the party, but for anyone coming across this question in the future...
Eye movement data looks very similar to this; I'd base an approach off that used by Nystrom + Holmqvist, 2010. Smooth the data using a Savitsky-Golay filter (scipy.signal.savgol_filter in scipy v0.14+) to get rid of some of the low-level noise while keeping the large peaks intact - the authors recommend using an order of 2 and a window size of about twice the width of the smallest peak you want to be able to detect. You can find where the bands are by arbitrarily removing all values above a certain y value (set them to numpy.nan). Then take the (nan)mean and (nan)standard deviation of the remainder, and remove all values greater than the mean + [parameter]*std (I think they use 6 in the paper). Iterate until you're not removing any data points - but depending on your data, certain values of [parameter] may not stabilise. Then use numpy.isnan() to find events vs non-events, and numpy.diff() to find the start and end of each event (values of -1 and 1 respectively). To get even more accurate start and end points, you can scan along the data backward from each start and forward from each end to find the nearest local minimum which has value smaller than mean + [another parameter]*std (I think they use 3 in the paper). Then you just need to count the data points between each start and end.
This won't work for that double peak; you'd have to do some extrapolation for that.
The best method might be to statistically compare a bunch of methods with human results.
You would take a large variety data and a large variety of measurement estimates (widths at various thresholds, area above various thresholds, different threshold selection methods, 2nd moments, polynomial curve fits of various degrees, pattern matching, and etc.) and compare these estimates to human measurements of the same data set. Pick the estimate method that correlates best with expert human results. Or maybe pick several methods, the best one for each of various heights, for various separations from other peaks, and etc.
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.