The Scharr-Filter is explained in Scharrs dissertation. However the values given on page 155 (167 in the pdf) are [47 162 47] / 256. Multiplying this with the derivation-filter would yield:
Yet all other references I found use
Which is roughly the same as the ones given by Scharr, scaled by a factor of 32.
Now my guess is that the range can be represented better, but I'm curious if there is an official explanation somewhere.
To get the ball rolling on this question in case no "expert" can be found...
I believe the values [3, 10, 3] ... instead of [47 162 47] / 256 ... are used simply for speed. Recall that this method is competing against the Sobel Operator whose coefficient values are are 0, and positive/negative 1's and 2's.
Even though the divisor in the division, 256 or 512, is a power of 2 and can can be performed by a shift, doing that and multiplying by 47 or 162 is going to take more time. A multiplication by 3 however can in fact be done on some RISC architectures like the IBM POWER series in a single shift-and-add operation. That is 3x = (x << 1) + x. (On these architectures, the shifter and adder are separate units and can be done independently).
I don't find it surprising that Phd paper used the more complicated and probably more precise formula; it needed to prove or demonstrate something, and the author probably wasn't totally certain or concerned that it be used and implemented alongside other methods. The purpose in the thesis was probably to have "perfect rotational symmetry". Afterwards when one decides to implement it, that person I suspect used the approximation formula and gave up a little on perfect rotational symmetry, to gain speed. That person's goal as I said was to have something that was competitive at the expense of little bit of speed for this rotational stuff.
Since I'm guessing you are willing to do work this as it is your thesis, my suggestion is to implement the original algorithm and benchmark it against both the OpenCV Scharr and Sobel code.
The other thing to try to get an "official" answer is: "Use the 'source', Luke!". The code is on github so check it out and see who added the Scharr filter there and contact that person. I won't put the person's name here, but I will say that the code was added 2010-05-11.
Related
This may show my naiveté but it is my understanding that quantum computing's obstacle is stabilizing the qbits. I also understand that standard computers use binary (on/off); but it seems like it may be easier with today's tech to read electric states between 0 and 9. Binary was the answer because it was very hard to read the varying amounts of electricity, components degrade over time, and maybe maintaining a clean electrical "signal" was challenging.
But wouldn't it be easier to try to solve the problem of reading varying levels of electricity so we can go from 2 inputs to 10 and thereby increasing the smallest unit of storage and exponentially increasing the number of paths through the logic gates?
I know I am missing quite a bit (sorry the puns were painful) so I would love to hear why or why not.
Thank you
"Exponentially increasing the number of paths through the logic gates" is exactly the problem. More possible states for each n-ary digit means more transistors, larger gates and more complex CPUs. That's not to say no one is working on ternary and similar systems, but the reason binary is ubiquitous is its simplicity. For storage, more possible states also means we need more sensitive electronics for reading and writing, and a much higher error frequency during these operations. There's a lot of hype around using DNA (base-4) for storage, but this is more on account of the density and durability of the substrate.
You're correct, though that your question is missing quite a bit - qubits are entirely different from classical information, whether we use bits or digits. Classical bits and trits respectively correspond to vectors like
Binary: |0> = [1,0]; |1> = [0,1];
Ternary: |0> = [1,0,0]; |1> = [0,1,0]; |2> = [0,0,1];
A qubit, on the other hand, can be a linear combination of classical states
Qubit: |Ψ> = α |0> + β |1>
where α and β are arbitrary complex numbers such that such that |α|2 + |β|2 = 1.
This is called a superposition, meaning even a single qubit can be in one of an infinite number of states. Moreover, unless you prepared the qubit yourself or received some classical information about α and β, there is no way to determine the values of α and β. If you want to extract information from the qubit you must perform a measurement, which collapses the superposition and returns |0> with probability |α|2 and |1> with probability |β|2.
We can extend the idea to qutrits (though, just like trits, these are even more difficult to effectively realize than qubits):
Qutrit: |Ψ> = α |0> + β |1> + γ |2>
These requirements mean that qubits are much more difficult to realize than classical bits of any base.
I have been trying to solve a problem stated in an exam of coursera. I am not seeking the solution but I need to get the steps and concepts to resolve this.
Can any one share the concept and steps to help me find the solution.
UPDATE:
I was expecting a down-vote and its not unusual, as its the most easiest thing people can do. I am seeking the direction to solve the problem as I wasn't able to get the idea to solve it after watching the videos on Coursera. I hope someone sensible out there can share a direction and step to achieve the mentioned goal.
Mean Normalization
Mean normalization, also known as 'standardization', is one of the most popular techniques of feature scaling.
Andrew Ng describes it in the 12a slide of lecture 4:
How to resolve the problem
The problem asks you to standardize the first feature in the third example: midterm = 94;
well, we have just to resolve the equation!
Just for clarity, the notation:
μ (mu) = "avg value of x in training set", in other words: the mean of the x1 column.
σ (sigma) = "range (max-min)", literaly σ = max-min (of the x1 column).
So:
μ = ( 89 + 72 + 94 +69 )/4 = 81
σ = ( 94 - 69 ) = 25
x_std = (94 - 81)/25 = 0.52
Result: 0.52
Best regards,
Marco.
The first step of solving this question is to identify what is , from the content of the lecture, it refers to the first feature of the third training case. Which is the unsquared version of the midterm score in the third row of the table.
Secondly, you need to understand the concept of normalization. The reason why we need normalization is that the value of some features among all training examples may much larger than the value of other features, which may make the cost function have pretty bad shape and this will make it harder gradient descent to find the minimum. In order to solve this, we want to make all features have nearly the same scale, and make the range of the feature to be centered at zero.
In this question, we want to scale every feature to a scale of 1, in order to do this, you need to find the max and min value of the feature among all training cases. Then squeeze the range of the feature to 0 and 1. The second step is to find the center value of the feature (average value in this case) and move the center value of the feature to 0.
I think this is pretty much all hints I can give you, you will totally be able to calculate the answer to this question by yourself from this point.
I'd be grateful if people could help me find an efficient way (probably low memory algorithm) to tackle the following problem.
I need to find the stationary distribution x of a transition matrix P. The transition matrix is an extremely large, extremely sparse matrix, constructed such that all the columns sum to 1. Since the stationary distribution is given by the equation Px = x, then x is simply the eigenvector of P associated with eigenvalue 1.
I'm currently using GNU Octave to both generate the transition matrix, find the stationary distribution, and plot the results. I'm using the function eigs(), which calculates both eigenvalues and eigenvectors, and it is possible to return just one eigenvector, where the eigenvalue is 1 (I actually had to specify 1.1, to prevent an error). Construction of the transition matrix (using a sparse matrix) is fairly quick, but finding the eigenvector gets increasingly slow as I increase the size, and I'm running out of memory before I can examine even moderately sized problems.
My current code is
[v l] = eigs(P, 1, 1.01);
x = v / sum(v);
Given that I know that 1 is the eigenvalue, I'm wondering if there is either a better method to calculate the eigenvector, or a way that makes more efficient use of memory, given that I don't really need an intermediate large dense matrix. I naively tried
n = size(P,1); % number of states
Q = P - speye(n,n);
x = Q\zeros(n,1); % solve (P-I)x = 0
which fails, since Q is singular (by definition).
I would be very grateful if anyone has any ideas on how I should approach this, as it's a calculation I have to perform a great number of times, and I'd like to try it on larger and more complex models if possible.
As background to this problem, I'm solving for the equilibrium distribution of the number of infectives in a cattle herd in a stochastic SIR model. Unfortunately the transition matrix is very large for even moderately sized herds. For example: in an SIR model with an average of 20 individuals (95% of the time the population is between 12 and 28 individuals), P is 21169 by 21169 with 20340 non-zero values (i.e. 0.0005% dense), and uses up 321 Kb (a full matrix of that size would be 3.3 Gb), while for around 50 individuals P uses 3 Mb. x itself should be pretty small. I suspect that eigs() has a dense matrix somewhere, which is causing me to run out of memory, so I should be okay if I can avoid using full matrices.
Power iteration is a standard way to find the dominant eigenvalue of a matrix. You pick a random vector v, then hit it with P repeatedly until you stop seeing it change very much. You want to periodically divide v by sqrt(v^T v) to normalise it.
The rate of convergence here is proportional to the separation between the largest eigenvalue and the second largest eigenvalue. Each iteration takes just a couple of matrix multiplies.
There are fancier-pants ways to do this ("PageRank" is one good thing to search for here) that improve speed for really huge sparse matrices, but I don't know that they're necessary or useful here.
Your approach seems like a good one. However, what you're calling x, is the null space of Q. null(Q) would work if it supported sparse matrices, but it doesn't. There's a bunch of stuff on the web for finding the null space of a sparse matrix. For example:
http://www.mathworks.co.uk/matlabcentral/newsreader/view_thread/249467
http://www.mathworks.com/matlabcentral/fileexchange/42922-null-space-for-sparse-matrix/content/nulls.m
http://www.mathworks.com/matlabcentral/fileexchange/11120-null-space-of-a-sparse-matrix
It seems the best solution is to use the Power Iteration method, as suggested by tmyklebu.
The method is to iterate x = Px; x /= sum(x), until x converges. I'm assuming convergence if the d1 norm between successive iterations is less than 1e-5, as that seems to give good results.
Convergence can take a while, since the largest two eigenvalues are fairly close (the number of iterations needed to converge can vary considerably, from around 200 to 2000 depending on the model used and population sizes, but it gets there in the end). However, the memory requirements are low, and it's very easy to implement.
I am a bit confused about the namings in the SVM. I am using this library LibSVM. There are so many parameters that can be set. Does anyone know which of these is the slack variable?
thx
The "slack variable" is C in c-svm and nu in nu-SVM. These both serve the same function in their respective formulations - controlling the tradeoff between a wide margin and classifier error. In the case of C, one generally test it in orders of magnitude, say 10^-4, 10^-3, 10^-2,... to 1, 5 or so. nu is a number between 0 and 1, generally from .1 to .8, which controls the ratio of support vectors to data points. When nu is .1, the margin is small, the number of support vectors will be a small percentage of the number of data points. When nu is .8, the margin is very large and most of the points will fall in the margin.
The other things to consider are your choice of kernel (linear, RBF, sigmoid, polynomial) and the parameters for the chosen kernel. Generally one has to do a lot of experimenting to find the best combination of parameters. However, be careful of over-fitting to your dataset.
Burges wrote a great tutorial: A Tutorial on Support Vector Machines for Pattern
Recognition
But if you mostly just want to know how to USE it and less about how it works, read "A Practical Guide to Support Vector Classication" by Chih-Wei Hsu, Chih-Chung Chang, and Chih-Jen Lin (authors of libsvm)
First decide which type of SVM are u intending to use: C-SVC, nu-SVC , epsilon-SVR or nu-SVR. In my opinion u need to vary C and gamma most of the time... the rest are usually fixed..
I have a scenario where I have several thousand instances of data. The data itself is represented as a single integer value. I want to be able to detect when an instance is an extreme outlier.
For example, with the following example data:
a = 10
b = 14
c = 25
d = 467
e = 12
d is clearly an anomaly, and I would want to perform a specific action based on this.
I was tempted to just try an use my knowledge of the particular domain to detect anomalies. For instance, figure out a distance from the mean value that is useful, and check for that, based on heuristics. However, I think it's probably better if I investigate more general, robust anomaly detection techniques, which have some theory behind them.
Since my working knowledge of mathematics is limited, I'm hoping to find a technique which is simple, such as using standard deviation. Hopefully the single-dimensioned nature of the data will make this quite a common problem, but if more information for the scenario is required please leave a comment and I will give more info.
Edit: thought I'd add more information about the data and what I've tried in case it makes one answer more correct than another.
The values are all positive and non-zero. I expect that the values will form a normal distribution. This expectation is based on an intuition of the domain rather than through analysis, if this is not a bad thing to assume, please let me know. In terms of clustering, unless there's also standard algorithms to choose a k-value, I would find it hard to provide this value to a k-Means algorithm.
The action I want to take for an outlier/anomaly is to present it to the user, and recommend that the data point is basically removed from the data set (I won't get in to how they would do that, but it makes sense for my domain), thus it will not be used as input to another function.
So far I have tried three-sigma, and the IQR outlier test on my limited data set. IQR flags values which are not extreme enough, three-sigma points out instances which better fit with my intuition of the domain.
Information on algorithms, techniques or links to resources to learn about this specific scenario are valid and welcome answers.
What is a recommended anomaly detection technique for simple, one-dimensional data?
Check out the three-sigma rule:
mu = mean of the data
std = standard deviation of the data
IF abs(x-mu) > 3*std THEN x is outlier
An alternative method is the IQR outlier test:
Q25 = 25th_percentile
Q75 = 75th_percentile
IQR = Q75 - Q25 // inter-quartile range
IF (x < Q25 - 1.5*IQR) OR (Q75 + 1.5*IQR < x) THEN x is a mild outlier
IF (x < Q25 - 3.0*IQR) OR (Q75 + 3.0*IQR < x) THEN x is an extreme outlier
this test is usually employed by Box plots (indicated by the whiskers):
EDIT:
For your case (simple 1D univariate data), I think my first answer is well suited.
That however isn't applicable to multivariate data.
#smaclell suggested using K-means to find the outliers. Beside the fact that it is mainly a clustering algorithm (not really an outlier detection technique), the problem with k-means is that it requires knowing in advance a good value for the number of clusters K.
A better suited technique is the DBSCAN: a density-based clustering algorithm. Basically it grows regions with sufficiently high density into clusters which will be maximal set of density-connected points.
DBSCAN requires two parameters: epsilon and minPoints. It starts with an arbitrary point that has not been visited. It then finds all the neighbor points within distance epsilon of the starting point.
If the number of neighbors is greater than or equal to minPoints, a cluster is formed. The starting point and its neighbors are added to this cluster and the starting point is marked as visited. The algorithm then repeats the evaluation process for all the neighbors recursively.
If the number of neighbors is less than minPoints, the point is marked as noise.
If a cluster is fully expanded (all points within reach are visited) then the algorithm proceeds to iterate through the remaining unvisited points until they are depleted.
Finally the set of all points marked as noise are considered outliers.
There are a variety of clustering techniques you could use to try to identify central tendencies within your data. One such algorithm we used heavily in my pattern recognition course was K-Means. This would allow you to identify whether there are more than one related sets of data, such as a bimodal distribution. This does require you having some knowledge of how many clusters to expect but is fairly efficient and easy to implement.
After you have the means you could then try to find out if any point is far from any of the means. You can define 'far' however you want but I would recommend the suggestions by #Amro as a good starting point.
For a more in-depth discussion of clustering algorithms refer to the wikipedia entry on clustering.
This is an old topic but still it lacks some information.
Evidently, this can be seen as a case of univariate outlier detection. The approaches presented above have several pros and cons. Here are some weak spots:
Detection of outliers with the mean and sigma has the obvious disadvantage of dependence of mean and sigma on the outliers themselves.
The case of the small sample limit (see question for example) is not adequately covered by, 3 sigma, K-Means, IQR etc.
And I could go on... However the statistical literature offers a simple metric: the median absolute deviation. (Medians are insensitive to outliers)
Details can be found here: https://www.sciencedirect.com/book/9780128047330/introduction-to-robust-estimation-and-hypothesis-testing
I think this problem can be solved in a few lines of python code like this:
import numpy as np
import scipy.stats as sts
x = np.array([10, 14, 25, 467, 12]) # your values
np.abs(x - np.median(x))/(sts.median_abs_deviation(x)/0.6745) #MAD criterion
Subsequently you reject values above a certain threshold (97.5 percentile of the distribution of data), in case of an assumed normal distribution the threshold is 2.24. Here it translates to:
array([ 0.6745 , 0. , 1.854875, 76.387125, 0.33725 ])
or the 467 entry being rejected.
Of course, one could argue, that the MAD (as presented) also assumes a normal dist. Therefore, why is it that argument 2 above (small sample) does not apply here? The answer is that MAD has a very high breakdown point. It is easy to choose different threshold points from different distributions and come to the same conclusion: 467 is the outlier.
Both three-sigma rule and IQR test are often used, and there are a couple of simple algorithms to detect anomalies.
The three-sigma rule is correct
mu = mean of the data
std = standard deviation of the data
IF abs(x-mu) > 3*std THEN x is outlier
The IQR test should be:
Q25 = 25th_percentile
Q75 = 75th_percentile
IQR = Q75 - Q25 // inter-quartile range
If x > Q75 + 1.5 * IQR or x < Q25 - 1.5 * IQR THEN x is a mild outlier
If x > Q75 + 3.0 * IQR or x < Q25 – 3.0 * IQR THEN x is a extreme outlier