low pass filter on ongoing samples - lowpass-filter

My application requires to sample a sensor and send the samples out in real time, without saving the samples.
I am required to run a low pass filter on the samples, but here is the problem cause if I want to do it on 20 samples for example then I will not be able to send out the samples each after I got it as required, and when I send them immediately each after I got it I use an ongoing average on 10 samples but that wasn't enough.
what low pass filter algorithm is the best for this application?
what is the accepted way to do this ?

Try the Exponential, or First-order filter:
Y[n] = dY[n-1] + (1-d)X[n]
Y[n] is the current output
Y[n-1] is the previous output
X[n] is the current input
d is the damping factor
The damping factor is a number between 0 and 1. If d = 0, the output is just equal to the input, and no filtering (smoothing) takes place. The closer d is to 1, the greater the amount of filtering (smoothing).
Here is an example: The input signal is a Sine function with amplitude +- 100, added to that is random noise of +-10. To filter out the noise we run the data through the filter equation first with d = 0.5 and then d = 0.8
More details with code and second and third order filters available here >> ST Community- answer

Related

How to fix skew trapezoidal distribution sampling output sample size

I am trying to generate a skewed trapezoidal distribution using inverse transform sampling.
The inputs are the values where the ramps start and end (a, b, c, d) and the sample size.
a=-3;b=-1;c=1;d=8;
SampleSize=10e4;
h=2/(d+c-a-b);
Then I calculate the ratio of the length of ramps and flat components to get sample size for each:
firstramp=round(((b-a)/(d-a)),3);
flat=round((c-b)/(d-a),3);
secondramp=round((d-c)/(d-a),3);
n1=firstramp*SampleSize; %sample size for first ramp
n3=secondramp*SampleSize; %sample size for second ramp
n2=flat*SampleSize;
And then finally I get the histogram from the following code:
quartile1=h/2*(b-a);
quartile2=1-h/2*(d-c);
y1=linspace(0,quartile1,n1);
y2=linspace(quartile1,quartile2,n2);
y3=linspace(quartile2,1,n3);
%inverse cumulative distribution functions
invcdf1=a+sqrt(2*(b-a)/h)*sqrt(y1);
invcdf2=(a+b)/2+y2/h;
invcdf3=d-sqrt(2*(d-c)/h)*sqrt(1-y3);
distr=[invcdf1 invcdf2 invcdf3];
histogram(distr,100)
However the sampling of ramps and flat components are not equal, looks like this:
I fixed this by trial and error, by reducing the sample size of the ramps by half:
n1=0.5*firstramp*SampleSize; %sample size for first ramp
n3=0.5*secondramp*SampleSize; %sample size for second ramp
n2=flat*SampleSize;
This made the distribution look like this:
However this makes the output sample less than what is given in input.
I've also tried different combinations of changing the sample sizes of ramps and flat.
This also works:
n1=0.75*firstramp*SampleSize; %sample size for first ramp
n3=0.75*secondramp*SampleSize; %sample size for second ramp
n2=1.5*flat*SampleSize;
It increases the output samples, but it's still not close.
Any help will be appreciated.
Full code:
a=-3;b=-1;c=1;d=8;
SampleSize=10e4;%*1.33333333333333;
h=2/(d+c-a-b);
firstramp=round(((b-a)/(d-a)),3);
flat=round((c-b)/(d-a),3);
secondramp=round((d-c)/(d-a),3);
n1=firstramp*SampleSize; %sample size for first ramp
n3=secondramp*SampleSize; %sample size for second ramp
n2=flat*SampleSize;
quartile1=h/2*(b-a);
quartile2=1-h/2*(d-c);
y1=linspace(0,quartile1,.75*n1);
y2=linspace(quartile1,quartile2,1.5*n2);
y3=linspace(quartile2,1,.75*n3);
%inverse cumulative distribution functions
invcdf1=a+sqrt(2*(b-a)/h)*sqrt(y1);
invcdf2=(a+b)/2+y2/h;
invcdf3=d-sqrt(2*(d-c)/h)*sqrt(1-y3);
distr=[invcdf1 invcdf2 invcdf3];
histogram(distr,100)
%end
I don't know Matlab so I was hoping somebody else would jump in on this, but since nobody did here goes.
If I'm reading your code correctly what you did is not an inversion. Inversion is 1-1, i.e., one uniform input produces one outcome. You seem to be using a technique known as the "composition method". In composition the overall distribution is comprised of component pieces, each of which is straightforward to generate. You choose which component to generate from based on their proportions/probabilities relative to the whole. For density functions, probability is found as the area under the density curve, so your first mistake was in sampling the components relative to the width of each component rather than using their areas. The correct sampling proportions are 2/13, 4/13, and 7/13 for what you designated the firstramp, flat, and secondramp components, respectively. A second mistake (which is relatively minor) was to assign exact sample sizes to each of the components. Having probability 2/13 does not mean that exactly 2*SampleSize/13 of your samples will be from the firstramp, it means that's the expected sample size for that component. The expected value of a random variate is not necessarily (or even likely to be) the outcome you will actually get.
In pseudocode, the composition approach would be
generate U ~ Uniform(0,1)
if U <= 2/13:
generate and return a value from firstramp
else if U <= 6/13:
generate and return a value from flat
else:
generate and return a value from secondramp
Note that since each of the generate options will use one or more uniforms, and choosing between the options requires a uniform U, this is not an inversion.
If you want an actual inversion, you need to quantify your density, integrate it to get the cumulative distribution function, then apply the inversion technique by setting F(X) = U and solving for X. Since your distribution is made of distinct components, both the density and cumulative density will be piecewise functions.
After deriving the height based on the requirement that the areas of the two triangles and the flat section must add up to 1, I came up with the following for your density:
| (x + 3) / 13 -3 <= x <= -1
|
f(x) = | 2 / 13 -1 <= x <= 1
|
| 2 * (8 - x) / 91 1 <= x <= 8
Integrating this and collecting terms produces the CDF:
| (x + 3)**2 / 26 -3 <= x <= -1
|
F(x) = | (2 + x) * 2 / 13 -1 <= x <= 1
|
| 6 / 13 + [49 - (x - 8)**2] / 91 1 <= x <= 8
Finally, determining the values of F(x) at the break points between the segments and applying inversion yields the following pseudocode algorithm:
generate U ~ Uniform(0,1)
if U <= 2 / 13:
return 2 * sqrt( (13 * U) / 2 ) - 3
else if U <= 6 / 13:
return (13 * U) / 2 - 2:
else:
return 8 - sqrt( 91 * (1 - U) )
Note that this is a true inversion. The outcome is determined by generating a single U, and transforming it in different ways depending on which range it falls in.

What techniques are effective to find periodicity in arbitrary data points?

By "arbitrary" I mean that I don't have a signal sampled on a grid that is amenable to taking an FFT. I just have points (e.g. in time) where events happened, and I'd like an estimate of the rate, for example:
p = [0, 1.1, 1.9, 3, 3.9, 6.1 ...]
...could be hits from a process with a nominal periodicity (repetition interval) of 1.0, but with noise and some missed detections.
Are there well known methods for processing such data?
A least square algorithm may do the trick, if correctly initialized. A clustering method can be applied to this end.
As an FFT is performed, the signal is depicted as a sum of sine waves. The amplitude of the frequencies may be depicted as resulting from a least square fit on the signal. Hence, if the signal is unevenly sampled, resolving the same least square problem may make sense if the Fourier transform is to be estimated. If applied to a evenly sampled signal, it boils down to the same result.
As your signal is descrete, you may want to fit it as a sum of Dirac combs. It seems more sound to minimize the sum of squared distance to the nearest Dirac of the Dirac comb. This is a non-linear optimization problem where Dirac combs are described by their period and offset. This non-linear least-square problem can be solved by mean of the Levenberg-Marquardt algorithm. Here is an python example making use of the scipy.optimize.leastsq() function. Moreover, the error on the estimated period and offset can be estimated as depicted in How to compute standard deviation errors with scipy.optimize.least_squares . It is also documented in the documentation of curve_fit() and Getting standard errors on fitted parameters using the optimize.leastsq method in python
Nevertheless, half the period, or the thrid of the period, ..., would also fit, and multiples of the period are local minima that are to be avoided by a refining the initialization of the Levenberg-Marquardt algorithm. To this end, the differences between times of events can be clustered, the cluster featuring the smallest value being that of the expected period. As proposed in Clustering values by their proximity in python (machine learning?) , the clustering function sklearn.cluster.MeanShift() is applied.
Notice that the procedure can be extended to multidimentionnal data to look for periodic patterns or mixed periodic patterns featuring different fundamental periods.
import numpy as np
from scipy.optimize import least_squares
from scipy.optimize import leastsq
from sklearn.cluster import MeanShift, estimate_bandwidth
ticks=[0,1.1,1.9,3,3.9,6.1]
import scipy
print scipy.__version__
def crudeEstimate():
# loooking for the period by looking at differences between values :
diffs=np.zeros(((len(ticks))*(len(ticks)-1))/2)
k=0
for i in range(len(ticks)):
for j in range(i):
diffs[k]=ticks[i]-ticks[j]
k=k+1
#see https://stackoverflow.com/questions/18364026/clustering-values-by-their-proximity-in-python-machine-learning
X = np.array(zip(diffs,np.zeros(len(diffs))), dtype=np.float)
bandwidth = estimate_bandwidth(X, quantile=1.0/len(ticks))
ms = MeanShift(bandwidth=bandwidth, bin_seeding=True)
ms.fit(X)
labels = ms.labels_
cluster_centers = ms.cluster_centers_
print cluster_centers
labels_unique = np.unique(labels)
n_clusters_ = len(labels_unique)
for k in range(n_clusters_):
my_members = labels == k
print "cluster {0}: {1}".format(k, X[my_members, 0])
estimated_period=np.min(cluster_centers[:,0])
return estimated_period
def disttoDiracComb(x):
residual=np.zeros((len(ticks)))
for i in range(len(ticks)):
mindist=np.inf
for j in range(len(x)/2):
offset=x[2*j+1]
period=x[2*j]
#print period, offset
index=np.floor((ticks[i]-offset)/period)
#print 'index', index
currdist=ticks[i]-(index*period+offset)
if currdist>0.5*period:
currdist=period-currdist
index=index+1
#print 'event at ',ticks[i], 'not far from index ',index, '(', currdist, ')'
#currdist=currdist*currdist
#print currdist
if currdist<mindist:
mindist=currdist
residual[i]=mindist
#residual=residual-period*period
#print x, residual
return residual
estimated_period=crudeEstimate()
print 'crude estimate by clustering :',estimated_period
xp=np.array([estimated_period,0.0])
#res_1 = least_squares(disttoDiracComb, xp,method='lm',xtol=1e-15,verbose=1)
p,pcov,infodict,mesg,ier=leastsq(disttoDiracComb, x0=xp,ftol=1e-18, full_output=True)
#print ' p is ',p, 'covariance is ', pcov
# see https://stackoverflow.com/questions/14581358/getting-standard-errors-on-fitted-parameters-using-the-optimize-leastsq-method-i
s_sq = (disttoDiracComb(p)**2).sum()/(len(ticks)-len(p))
pcov=pcov *s_sq
perr = np.sqrt(np.diag(pcov))
#print 'estimated standard deviation on parameter :' , perr
print 'estimated period is ', p[0],' +/- ', 1.96*perr[0]
print 'estimated offset is ', p[1],' +/- ', 1.96*perr[1]
Applied to your sample, it prints :
crude estimate by clustering : 0.975
estimated period is 1.0042857141346768 +/- 0.04035792507868619
estimated offset is -0.011428571139828817 +/- 0.13385206912205957
It sounds like you need to decide what exactly you want to determine. If you want to know the average interval in a set of timestamps, then that's easy (just take the mean or median).
If you expect that the interval could be changing, then you need to have some idea about how fast it is changing. Then you can find a windowed moving average. You need to have an idea of how fast it is changing so that you can select your window size appropriately - a larger window will give you a smoother result, but a smaller window will be more responsive to a faster-changing rate.
If you have no idea whether the data is following any sort of pattern, then you are probably in the territory of data exploration. In that case, I would start by plotting the intervals, to see if a pattern appears to the eye. This might also benefit from applying a moving average if the data is quite noisy.
Essentially, whether or not there is something in the data and what it means is up to you and your knowledge of the domain. That is, in any set of timestamps there will be an average (and you can also easily calculate the variance to give an indication of variability in the data), but it is up to you whether that average carries any meaning.

KALMAN filter doesn't respond to changes

I am implementing a Kalman filter for the first time to get voltage values from a source. It works and it stabilizes at the source voltage value but if then the source changes the voltage the filter doesn't adapt to the new value.
I use 3 steps:
Get the Kalman gain
KG = previous_error_in_estimate / ( previous_error_in_estimate + Error_in_measurement )
Get current estimation
Estimation = previous_estimation + KG*[measurement - previous_estimation]
Calculate the error in estimate
Error_in_estimate = [1-KG]*previous_error_in_estimate
The thing is that, as 0 <= KG <= 1, Error_in_estimate decreases more and more and that makes KG to also decrease more and more ( error_in_measurement is a constant ), so at the end the estimation only depends on the previous estimation and the current measurement is not taken into account.
This prevents the filter from adapt himself to measurement changes.
How can I do to make that happen?
Thanks
EDIT:
Answering to Claes:
I am not sure that the Kalman filter is valid for my problem since I don't have a system model, I just have a bunch of readings from a quite noisy sensor measuring a not very predictable variable.
To keep things simple, imagine reading a potentiometer ( a variable resistor ) changed by the user, you can't predict or model the user's behavior.
I have implemented a very basic SMA ( Simple Moving Average ) algorithm and I was wondering if there is a better way to do it.
Is the Kalman filter valid for a problem like this?
If not, what would you suggest?
2ND EDIT
Thanks to Claes for such an useful information
I have been doing some numerical tests in MathLab (with no real data yet) and doing the convolution with a Gaussian filter seems to give the most accurate result.
With the Kalman filter I don't know how to estimate the process and measurement variances, is there any method for that?. Only when I decrease quite a lot the measurement variance the kalman filter seems to adapt. In the previous image the measurement variance was R=0.1^2 (the one in the original example). This is the same test with R=0.01^2
Of course, these are MathLab tests with no real data. Tomorrow I will try to implement this filters in the real system with real data and see if I can get similar results
A simple MA filter is probably sufficient for your example. If you would like to use the Kalman filter there is a great example at the SciPy cookbook
I have modified the code to include a step change so you can see the convergence.
# Kalman filter example demo in Python
# A Python implementation of the example given in pages 11-15 of "An
# Introduction to the Kalman Filter" by Greg Welch and Gary Bishop,
# University of North Carolina at Chapel Hill, Department of Computer
# Science, TR 95-041,
# http://www.cs.unc.edu/~welch/kalman/kalmanIntro.html
# by Andrew D. Straw
import numpy as np
import matplotlib.pyplot as plt
plt.rcParams['figure.figsize'] = (10, 8)
# intial parameters
n_iter = 400
sz = (n_iter,) # size of array
x1 = -0.37727*np.ones(n_iter/2) # truth value 1
x2 = -0.57727*np.ones(n_iter/2) # truth value 2
x = np.concatenate((x1,x2),axis=0)
z = x+np.random.normal(0,0.1,size=sz) # observations (normal about x, sigma=0.1)
Q = 1e-5 # process variance
# allocate space for arrays
xhat=np.zeros(sz) # a posteri estimate of x
P=np.zeros(sz) # a posteri error estimate
xhatminus=np.zeros(sz) # a priori estimate of x
Pminus=np.zeros(sz) # a priori error estimate
K=np.zeros(sz) # gain or blending factor
R = 0.1**2 # estimate of measurement variance, change to see effect
# intial guesses
xhat[0] = 0.0
P[0] = 1.0
for k in range(1,n_iter):
# time update
xhatminus[k] = xhat[k-1]
Pminus[k] = P[k-1]+Q
# measurement update
K[k] = Pminus[k]/( Pminus[k]+R )
xhat[k] = xhatminus[k]+K[k]*(z[k]-xhatminus[k])
P[k] = (1-K[k])*Pminus[k]
plt.figure()
plt.plot(z,'k+',label='noisy measurements')
plt.plot(xhat,'b-',label='a posteri estimate')
plt.plot(x,color='g',label='truth value')
plt.legend()
plt.title('Estimate vs. iteration step', fontweight='bold')
plt.xlabel('Iteration')
plt.ylabel('Voltage')
And the output is:

How to create a scoring system using two variables

I have an application (Node/Angular) that I'm creating where I'm trying to rank users based on overall performance across two metrics. There are two metrics used to track the users we are using are the following:
Units Produced (ranges between 0 - 6000)
Rate of production = [ Units Produced ] / [ Labor Hours ] (ranges between 0 - 100)
However, ranking users explicitly by either of these variables doesn't make sense, because it creates some strange incentives/behaviors.
For instance, it is possible to have a really high Rate of Production, but a super low number of total number of units produced by working really hard over a short period of time. Alternatively, you can have a very high number of Units Produced, but it may be due to the fact that they worked overtime, and thus were able to produce more units than anyone else just due to the fact that they had longer to work, and they could have a low Rate of Production.
Does anyone have experience designing these types of scoring systems? How have you handled it?
First, I would recommend to bring them on the same scale. E.g. divide Units produced by 60.
Then, if you are fine with equal weights, there are three common simple choices:
Add the scores
Multiply the scores (equal to adding logs of each)
Take the minimum of the two scores
Which of the ones is best, depends on to what extent you want it to be a measure of combined good results. In your case, I would recommend you to multiply and put a scale on the resulting product.
If you want to go a little more complex and weigh or play around with how much to reward separate vs joint scores, you can use the following formula:
V = alpha * log_b[Units Produced / 60] + (1-alpha) * log_b[Rate of Production],
where alpha determines the weighting of one vs the other and the base of the logarithmic function determines to what extent a joint success is rewarded.
I did something very similar I found it valuable to break them into leagues or tiers, for example using Units Produced as a base.
Novice = 100 Units Produced
Beginner = 500 Units Produced
Advanced = 2000 Units Produced
Expert = 4000 Units Produced
Putting this into a useable object
var levels = [
{id: 1, name: "Novice", minUnits: 100, maxUnits: 599 },
{id: 2, name: "Beginner", minUnits: 500, maxUnits: 1999 },
{id: 3, name: "Advanced", minUnits: 2000, maxUnits: 3999 },
{id: 4, name: "Expert", minUnits: 4000, maxUnits: 6000 }
]
You can then use your Rate of production to multiply by a weighted value inside the levels, you can determine what this is. You can play with the values to make it as hard or as easy as you want.
You can do a combination with
SCORE = 200/( K_1/x_1 + K_2/x_2 )
// x_1 : Score 1
// x_2 : Score 2
// K_1 : Maximum of Score 1
// K_2 : Maximum of Score 2
Of course be carefull when dividing by zero. If either x_1 or x_2 are zero then SCORE=0. If x_1=K_1 and x_2=K_2 then SCORE=100 (maximum)
Otherwise the score is somewhere in between. If x_1/K_1 = x_2/K_2 = z then SCORE = 100*z
This weighs the lower score more such that you get rewarded when raising one of the two scores (unlike a minimum of the two scenarios) but not as much as raising both.

Fastest approximate counting algorithm

Whats the fastest way to get an approximate count of number of rows of an input file or std out data stream. FYI, this is a probabilistic algorithm, I can't find many examples online.
The data could just be one or 2 columns coming from an awk script of csv file! Lets say i want an aprox groupby on one of the columns. I would use a database group by but the number of rows are over 6-7 billion. I would like the first approx result In under 3 to 4 seconds. Then run a bayes or something after decisions are made on the prior. Any ideas on a really rough initial group count?
If you can provide the algorithm example in python, or java that would be very helpful.
#Ben Allison's answer is a good way if you want to count the total lines. Since you mentioned the Bayes and the prior, I will add an answer in that direction to calculate the percentage of different groups. (see my comments on your question. I guess if you have an idea of the total and if you want to do a groupby, to estimate the percentage of different groups makes more sense).
The recursive Bayesian update:
I will start by assuming you have only two groups (extensions can be made to make it work for multiple groups, see later explanations for that.), group1 and group2.
For m group1s out of the first n lines(rows) you processed, we denote the event as M(m,n). Obviously you will see n-m group2s because we assume they are the only two possible groups. So you know the conditional probability of the event M(m,n) given the percentage of group1 (s), is given by the binomial distribution with n trials. We are trying to estimate s in a bayesian way.
The conjugate prior for binomial is beta distribution. So for simplicity, we choose Beta(1,1) as the prior (of course, you can pick your own parameters here for alpha and beta), which is a uniform distribution on (0,1). Therefor, for this beta distribution, alpha=1 and beta=1.
The recursive update formulas for a binomial + beta prior are as below:
if group == 'group1':
alpha = alpha + 1
else:
beta = beta + 1
The posterior of s is actually also a beta distribution:
s^(m+alpha-1) (1-s)^(n-m+beta-1)
p(s| M(m,n)) = ----------------------------------- = Beta (m+alpha, n-m+beta)
B(m+alpha, n-m+beta)
where B is the beta function. To report the estimate result, you can rely on Beta distribution's mean and variance, where:
mean = alpha/(alpha+beta)
var = alpha*beta/((alpha+beta)**2 * (alpha+beta+1))
The python code: groupby.py
So a few lines of python to process your data from stdin and estimate the percentage of group1 would be something like below:
import sys
alpha = 1.
beta = 1.
for line in sys.stdin:
data = line.strip()
if data == 'group1':
alpha += 1.
elif data == 'group2':
beta += 1.
else:
continue
mean = alpha/(alpha+beta)
var = alpha*beta/((alpha+beta)**2 * (alpha+beta+1))
print 'mean = %.3f, var = %.3f' % (mean, var)
The sample data
I feed a few lines of data to the code:
group1
group1
group1
group1
group2
group2
group2
group1
group1
group1
group2
group1
group1
group1
group2
The approximate estimation result
And here is what I get as results:
mean = 0.667, var = 0.056
mean = 0.750, var = 0.037
mean = 0.800, var = 0.027
mean = 0.833, var = 0.020
mean = 0.714, var = 0.026
mean = 0.625, var = 0.026
mean = 0.556, var = 0.025
mean = 0.600, var = 0.022
mean = 0.636, var = 0.019
mean = 0.667, var = 0.017
mean = 0.615, var = 0.017
mean = 0.643, var = 0.015
mean = 0.667, var = 0.014
mean = 0.688, var = 0.013
mean = 0.647, var = 0.013
The result shows that group1 is estimated to have 64.7% percent up to the 15th row processed (based on our beta(1,1) prior). You might notice that the variance keeps shrinking because we have more and more observation points.
Multiple groups
Now if you have more than 2 groups, just change the underline distribution from binomial to multinomial, and then the corresponding conjugate prior would be Dirichlet. Everything else you just make similar changes.
Further notes
You said you would like the approximate estimate in 3-4 seconds. In this case, you just sample a portion of your data and feed the output to the above script, e.g.,
head -n100000 YOURDATA.txt | python groupby.py
That's it. Hope it helps.
If it's reasonable to assume the data are IID (so there's no bias such as certain types of records occur in certain parts of the stream), then just subsample and scale up the counts by approximate size.
Take say the first million records (this should be processable in a couple of seconds). Its size is x units (MB, chars, whatever you care about). The full stream has size y where y >> x. Now, derive counts for whatever you care about from your sample x, and simply scale them by the factor y/*x* for approximate full-counts. An example: you want to know roughly how many records have column 1 with value v in the full stream. The first million records have a file size of 100MB, while the total file size is 10GB. In the first million records, 150,000 of them have value v for column 1. So, you assume that in the full 10GB of records, you'll see 150,000 * (10,000,000,000 / 100,000,000) = 15,000,000 with that value. Any statistics you compute on the sample can simply be scaled by the same factor to produce an estimate.
If there is bias in the data such that certain records are more or less likely to be in certain places of the file then you should select your sample records at random (or evenly spaced intervals) from the total set. This is going to ensure an unbiased, representative sample, but probably incur a much greater I/O overhead.

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