Graph Lassoalgorithm Sklearn.covariance.graph_lasso() - lasso-regression

I am working on replicating a paper titled “Improving Mean Variance Optimization through Sparse Hedging Restriction”. The authors’ idea is to use Graphical Lasso algorithm to infuse some bias in the estimation process of the inverse of the sample covariance matrix. The graphical lasso algorithm works perfectly fine in R, but when I use python on the same data with the same parameters I get two sorts of errors:
If I use coordinate descent (cd ) mode as a solver, I get a floating point error saying that: FloatingPointError: Non SPD result: the system is too ill-conditioned for this solver. The system is too ill-conditioned for this solver (The thing that bugs me is that I tried this solver on a simulated Positive definite matrix and It game me this error)
If I use the Least Angle Regression (LARS) mode (Which is less stable but recommended for ill-conditioned matrices) I get an overflow error stating OverflowError: int too large to convert to float
To my knowledge, unlike C++ and other languages, python is not restricted by an upper maximum for integer numbers (besides the capacity of the machine itself). Whereas the floats are restricted. I think this might be the source of the later problem. (I have also heard in the past that R is much more robust in terms of dealing ill-conditioned matrices). I would be glad to hear you experience with graph lasso in R or python.
With this email, I have attached a little python code that simulates this problem in a few lines. Any input will be of great appreciation.
Thank you all,
Skander
from sklearn.covariance import graph_lasso
from sklearn.datasets import make_spd_matrix
symetric_PD_mx= make_spd_matrix(100)
glout = graph_lasso(emp_cov=symetric_PD_mx, alpha=0.01,mode="lars")

Related

How does sklearn's standard DBSCAN run so fast?

I've been messing around with alternative implementations of DBSCAN for clustering radar data (like grid-based DBSCAN). Up to this point, I had been using sklearn's standard euclidean DBSCAN and it would run on 26,000 data points in less than a second. However, when I specify my own distance metric, like this:
X = np.column_stack((beam, gate, time_index))
num_pts = X.shape[0]
epsilons = np.array([[beam_eps]*num_pts, [gate_eps] * num_pts, [time_eps] * num_pts]).T
metric = lambda x, y, eps: np.sqrt(np.sum((x/eps - y/eps)**2))
def dist_metric(x, y, eps):
return np.sqrt(np.sum((x - y)**2))
db = DBSCAN(eps=eps, min_samples=minPts, metric=dist_metric, metric_params={'eps': epsilons}).fit(X)
it goes from 0.36 seconds to 92 minutes to run on the same data.
What I did in that code snippet can also be accomplished with just transforming the data beforehand and running standard Euclidean DBSCAN, but I'm trying to implement a reasonably fast version of Grid-based DBSCAN, for which the horizontal epsilon varies based on distance from the radar, so I won't be able to do that.
Part of the slowness in the above distance metric is because of that division by epsilon I think, because it only takes about a minute to run if I use a 'custom metric' that's just Euclidean distance:
metric = lambda x, y: np.sqrt(np.sum((x - y)**2))
How does sklearn's euclidean DBSCAN manage to run so much faster? I've been digging through the code, but haven't made sense of it so far.
Because it uses an index.
Furthermore, it avoids the slow and memory intensive Python interpreter, but does all the work in native code (compiled from Cython). This makes a huge difference when dealing with lots of primitive data such as doubles and ints that the Python interpreter would need to box.
Indexes make all the difference for similarity search. They can reduce the runtime from O(n²) to O(n log n).
But while the ball tree index allows custom metrics, the cost of invoking the python interpreter for every distance computation is very high, so if you really want a custom metric, edit the cython source code and compile sklearn yourself. Or you can use ELKI because the Java JVM can compile extension code into native code when necessary; it does not need to fallback to slow interpreter callbacks like sklearn.
In your case, it will likely be much better to rather preprocess the data. Scale it prior to clustering.

Machine learning: why the cost function does not need to be derivable?

I was playing around with Tensorflow creating a customized loss function and this question about general machine learning arose to my head.
My understanding is that the optimization algorithm needs a derivable cost function to find/approach a minimum, however we can use functions that are non-derivable such as the absolute function (there is no derivative when x=0). A more extreme example, I defined my cost function like this:
def customLossFun(x,y):
return tf.sign(x)
and I expected an error when running the code, but it actually worked (it didn't learn anything but it didn't crash).
Am I missing something?
You're missing the fact that the gradient of the sign function is somewhere manually defined in the Tensorflow source code.
As you can see here:
def _SignGrad(op, _):
"""Returns 0."""
x = op.inputs[0]
return array_ops.zeros(array_ops.shape(x), dtype=x.dtype)
the gradient of tf.sign is defined to be always zero. This, of course, is the gradient where the derivate exists, hence everywhere but not in zero.
The tensorflow authors decided to do not check if the input is zero and throw an exception in that specific case
In order to prevent TensorFlow from throwing an error, the only real requirement is that you cost function evaluates to a number for any value of your input variables. From a purely "will it run" perspective, it doesn't know/care about the form of the function its trying to minimize.
In order for your cost function to provide you a meaningful result when TensorFlow uses it to train a model, it additionally needs to 1) get smaller as your model does better and 2) be bounded from below (i.e. it can't go to negative infinity). It's not generally necessary for it to be smooth (e.g. abs(x) has a kink where the sign flips). Tensorflow is always able to compute gradients at any location using automatic differentiation (https://en.wikipedia.org/wiki/Automatic_differentiation, https://www.tensorflow.org/versions/r0.12/api_docs/python/train/gradient_computation).
Of course, those gradients are of more use if you've chose a meaningful cost function isn't isn't too flat.
Ideally, the cost function needs to be smooth everywhere to apply gradient based optimization methods (SGD, Momentum, Adam, etc). But nothing's going to crash if it's not, you can just have issues with convergence to a local minimum.
When the function is non-differentiable at a certain point x, it's possible to get large oscillations if the neural network converges to this x. E.g., if the loss function is tf.abs(x), it's possible that the network weights are mostly positive, so the inference x > 0 at all times, so the network won't notice tf.abs. However, it's more likely that x will bounce around 0, so that the gradient is arbitrarily positive and negative. If the learning rate is not decaying, the optimization won't converge to the local minimum, but will bound around it.
In your particular case, the gradient is zero all the time, so nothing's going to change at all.
If it didn't learn anything, what have you gained ? Your loss function is differentiable almost everywhere but it is flat almost anywhere so the minimizer can't figure out the direction towards the minimum.
If you start out with a positive value, it will most likely be stuck at a random value on the positive side even though the minima on the left side are better (have a lower value).
Tensorflow can be used to do calculations in general and it provides a mechanism to automatically find the derivative of a given expression and can do so across different compute platforms (CPU, GPU) and distributed over multiple GPUs and servers if needed.
But what you implement in Tensorflow does not necessarily have to be a goal function to be minimized. You could use it e.g. to throw random numbers and perform Monte Carlo integration of a given function.

Implementing a 2D recursive spatial filter using Scipy

Minimally, I would like to know how to achieve what is stated in the title. Specifically, signal.lfilter seems like the only implementation of a difference equation filter in scipy, but it is 1D, as shown in the docs. I would like to know how to implement a 2D version as described by this difference equation. If that's as simple as "bro, use this function," please let me know, pardon my naiveté, and feel free to disregard the rest of the post.
I am new to DSP and acknowledging there might be a different approach to answering my question so I will explain the broader goal and give context for the question in the hopes someone knows how do want I want with Scipy, or perhaps a better way than what I explicitly asked for.
To get straight into it, broadly speaking I am using vectorized computation methods (Numpy/Scipy) to implement a Monte Carlo simulation to improve upon a naive for loop. I have successfully abstracted most of my operations to array computation / linear algebra, but a few specific ones (recursive computations) have eluded my intuition and I continually end up in the digital signal processing world when I go looking for how this type of thing has been done by others (that or machine learning but those "frameworks" are much opinionated). The reason most of my google searches end up on scipy.signal or scipy.ndimage library references is clear to me at this point, and subsequent to accepting the "signal" representation of my data, I have spent a considerable amount of time (about as much as reasonable for a field that is not my own) ramping up the learning curve to try and figure out what I need from these libraries.
My simulation entails updating a vector of data representing the state of a system each period for n periods, and then repeating that whole process a "Monte Carlo" amount of times. The updates in each of n periods are inherently recursive as the next depends on the state of the prior. It can be characterized as a difference equation as linked above. Additionally this vector is theoretically indexed on an grid of points with uneven stepsize. Here is an example vector y and its theoretical grid t:
y = np.r_[0.0024, 0.004, 0.0058, 0.0083, 0.0099, 0.0133, 0.0164]
t = np.r_[0.25, 0.5, 1, 2, 5, 10, 20]
I need to iteratively perform numerous operations to y for each of n "updates." Specifically, I am computing the curvature along the curve y(t) using finite difference approximations and using the result at each point to adjust the corresponding y(t) prior to the next update. In a loop this amounts to inplace variable reassignment with the desired update in each iteration.
y += some_function(y)
Not only does this seem inefficient, but vectorizing things seems intuitive given y is a vector to begin with. Furthermore I am interested in preserving each "updated" y(t) along the n updates, which would require a data structure of dimensions len(y) x n. At this point, why not perform the updates inplace in the array? This is wherein lies the question. Many of the update operations I have succesfully vectorized the "Numpy way" (such as adding random variates to each point), but some appear overly complex in the array world.
Specifically, as mentioned above the one involving computing curvature at each element using its neighbouring two elements, and then imediately using that result to update the next row of the array before performing its own curvature "update." I was able to implement a non-recursive version (each row fails to consider its "updated self" from the prior row) of the curvature operation using ndimage generic_filter. Given the uneven grid, I have unique coefficients (kernel weights) for each triplet in the kernel footprint (instead of always using [1,-2,1] for y'' if I had a uniform grid). This last part has already forced me to use a spatial filter from ndimage rather than a 1d convolution. I'll point out, something conceptually similar was discussed in this math.exchange post, and it seems to me only the third response saliently addressed the difference between mathematical notion of "convolution" which should be associative from general spatial filtering kernels that would require two sequential filtering operations or a cleverly merged kernel.
In any case this does not seem to actually address my concern as it is not about 2D recursion filtering but rather having a backwards looking kernel footprint. Additionally, I think I've concluded it is not applicable in that this only allows for "recursion" (backward looking kernel footprints in the spatial filtering world) in a manner directly proportional to the size of the recursion. Meaning if I wanted to filter each of n rows incorporating calculations on all prior rows, it would require a convolution kernel far too big (for my n anyways). If I'm understanding all this correctly, a recursive linear filter is algorithmically more efficient in that it returns (for use in computation) the result of itself applied over the previous n samples (up to a level where the stability of the algorithm is affected) using another companion vector (z). In my case, I would only need to look back one step at output signal y[n-1] to compute y[n] from curvature at x[n] as the rest works itself out like a cumsum. signal.lfilter works for this, but I can't used that to compute curvature, as that requires a kernel footprint that can "see" at least its left and right neighbors (pixels), which is how I ended up using generic_filter.
It seems to me I should be able to do both simultaneously with one filter namely spatial and recursive filtering; or somehow I've missed the maths of how this could be mathematically simplified/combined (convolution of multiples kernels?).
It seems like this should be a common problem, but perhaps it is rarely relevant to do both at once in signal processing and image filtering. Perhaps this is why you don't use signals libraries solely to implement a fast monte carlo simulation; though it seems less esoteric than using a tensor math library to implement a recursive neural network scan ... which I'm attempting to do right now.
EDIT: For those familiar with the theoretical side of DSP, I know that what I am describing, the process of designing a recursive filters with arbitrary impulse responses, is achieved by employing a mathematical technique called the z-transform which I understand is generally used for two things:
converting between the recursion coefficients and the frequency response
combining cascaded and parallel stages into a single filter
Both are exactly what I am trying to accomplish.
Also, reworded title away from FIR / IIR because those imply specific definitions of "recursion" and may be confusing / misnomer.

How to create a synthetic dataset

I want to run some Machine Learning clustering algorithms on some big data.
The problem is that I'm having troubles to find interesting data for this purpose on the web.Also, usually this data might be inconvenient to use because the format won't fit for me.
I need a txt file which each line represents a mathematical vector, each element seperated by space, for example:
1 2.2 3.1
1.12 0.13 4.46
1 2 54.44
Therefore, I decided to first run those algorithms on some synthetic data which I'll create by my self. How can I do this in a smart way with numpy?
In smart way, I mean that it won't be generated uniformly, because it's a little bit boring. How can I generate some interesting clusters?
I want to have 5GB / 10GB of data at the moment.
You need to define what you mean by "clusters", but I think what you are asking for is several random-parameter normal distributions combined together, for each of your coordinate values.
From http://docs.scipy.org/doc/numpy-1.10.0/reference/generated/numpy.random.randn.html#numpy.random.randn:
For random samples from N(\mu, \sigma^2), use:
sigma * np.random.randn(...) + mu
And use <range> * np.random.rand(<howmany>) for each of sigma and mu.
There is no one good answer for such question. What is interesting? For clustering, unfortunately, there is no such thing as an interesting or even well posed problem. Clustering as such has no well defineid evaluation, consequently each method is equally good/bad, as long as it has well defined internal objective. So k-means will always be good one to minimize inter-cluster euclidean distance and will struggle with sparse data, non-convex, imbalanced clusters. DBScan will always be the best in greedy density based sense and will strugle with diverse density clusters. GMM will be always great fitting on gaussian mixtures, and will strugle with clusters which are not gaussians (for example lines, squares etc.).
From the question one could deduce that you are at the very begining of work with clustering and so need "just anything more complex than uniform", so I suggest you take a look at datasets generators, in particular accesible in scikit-learn (python) http://scikit-learn.org/stable/datasets/ or in clusterSim (R) http://www.inside-r.org/packages/cran/clusterSim/docs/cluster.Gen or clusterGeneration (R) https://cran.r-project.org/web/packages/clusterGeneration/clusterGeneration.pdf

LDPC behaviour as density of parity-check matrix increases

My assignment is to implement a Loopy Belief Propagation algorithm for Low-density Parity-check Code. This code uses a parity-check matrix H which is rather sparse (say 750-by-1000 binary matrix with an average of about 3 "ones" per each column). The code to generate the parity-check matrix is taken from here
Anyway, one of the subtasks is to check the reliability of LDPC code when the density of the matrix H increases. So, I fix the channel at 0.5 capacity, fix my code speed at 0.35 and begin to increase the density of the matrix. As the average number of "ones" in a column goes from 3 to 7 in steps of 1, disaster happens. With 3 or 4 the code copes perfectly well. With higher density it begins to fail: not only does it sometimes fail to converge, it oftentimes converges to the wrong codeword and produces mistakes.
So my question is: what type of behaviour is expected of an LDPC code as its sparse parity-check matrix becomes denser? Bonus question for skilled mind-readers: in my case (as the code performance degrades) is it more likely because the Loopy Belief Propagation algo has no guarantee on convergence or because I made a mistake implementing it?
After talking to my TA and other students I understand the following:
According to Shannon's theorem, the reliability of the code should increase with the density of the parity check matrix. That is simply because more checks are made.
However, since we use Loopy Belief Propagation, it struggles a lot when there are more and more edges in the graph forming more and more loops. Therefore, the actual performance degrades.
Whether or not I made a mistake in my code based solely on this behaviour cannot be established. However, since my code does work for sparse matrices, it is likely that the implementation is fine.