For a simulation study I am working on, we are trying to test an algorithm that aims to identify specific culprit factors that predict a binary outcome of interest from a large mixture of possible exposures that are mostly unrelated to the outcome. To test this algorithm, I am trying to simulate the following data:
A binary dependent variable
A set of, say, 1000 variables, most binary and some continuous, that are not associated with the outcome (that is, are completely independent from the binary dependent variable, but that can still be correlated with one another).
A group of 10 or so binary variables which will be associated with the dependent variable. I will a-priori determine the magnitude of the correlation with the binary dependent variable, as well as their frequency in the data.
Generating a random set of binary variables is easy. But is there a way of doing this while ensuring that none of these variables are correlated with the dependent outcome?
Thank you!
"But is there a way of doing this while ensuring that none of these variables are correlated with the dependent outcome?"
With statistical sampling you can't ensure anything, you can only adjust the acceptable risk. Finding an acceptable level of risk may be harder than many people think.
Spurious correlations are a very real phenomenon. Real independent observations will often contain correlations, and if you want to actually test your algorithm to see how it will perform in reality then your tests should produce such phenomena in a manner similar to the real world—you should be generating independent candidate factors and allowing spurious correlations to occur.
If you are performing ~1000 independent tests of candidate factors, and you're targeting a risk level of α = 0.05, you can expect 50 non-significant terms to leak through into your analysis. To avoid this, you need to adjust your testing threshold using something along the lines of a Bonferroni correction. Recall that statistical discriminating power is based on standard error, which is inversely proportional to the square root of the sample size. Bonferroni says that 1000 simultaneous tests need their individual test threshold to be adjusted by a factor of 1000, which in turn means the sample size needs to be a million times larger than when performing a single test for significance.
So in summary I'd say that you shouldn't attempt to ensure lack of correlation, it's going to occur in the real world. You can mitigate the risk of non-predictive factors being included due to spurious correlation by generating massive amounts of data. In practice there will be non-predictors that leak through unless you can obtain enough data, so I'd suggest that your testing should address the rates of occurrence as a function of number of candidate factors and the sample size.
Related
I'm working for quite some time on finding a numerical instance for solution of a 8 variables system of 7 very complicated inequalities plus region specification. Unfortunately I cannot produce a MWE or nothing of the sort since the inputs are really long.
My current method is Mathematica's NMinimize routine, minimizing one of the 7 inequalities subject to every other condition as constraint -- The FindInstance command simply quits the kernel without being able to finish running.
The NMinimize is able to produce output, but besides being slower than would be optimal, produce results that do not obey every constraint.
The thing is that I need to be certain, for each benchmark I run, that if the output doesn't satisfy every constraint it is because such a set of real numbers doesn't exist -- which with my current method I can't be, by experience.
So: is there a foolproof, as efficient as possible, computational method for me to find a single instance of numerical solution to 7 complicated inequalities (involving trigonometric functions) of 8 variables or be sure that such a set doesn't exist?
It could be a Mathematica/python/fortran package, genetic algorithm or anything -- as long as there is clear enough documentation.
You need to give importance multiplier to constraints and the optimization method should not be greedy.
A genetic algorithm combined with multiple-starting points (or simulated annealing for diminishing mutations) tends to converge to global minima (hence not greedy) with more time given to it but there is not guarantee that the heuristic will complete X function in Y time. The more time given to it, the better it converges to global minima.
In genetic algorithm, you can add big constraint penalties like this:
fitness_minima = some_function_output_between_1_and_10 +
constraints_breached?1000.0f:0;
so that the DNAs with no contraint-violations will be favored for the crossover part of GA.
"As efficient as possible" depends on your algorithm. If you can parallelize the algorithm and run it on multiple GPUs, it should give substantial speedup over CPU. Compared to some hours of Mona-Lisa painting by CPU, a parallelized version running on 3 low-end GPUs complete within 10 minutes (https://www.youtube.com/watch?v=QRZqBLJ6brQ). At least some OpenCL/CUDA supporting libraries/frameworks (like Tensorflow) should be able to accelerate your algorithm if you don't want to do the work distribution yourself.
The documentation of MALLET mentions following:
--num-iterations [NUMBER]
The number of sampling iterations should be a trade off between the time taken to complete sampling and the quality of the topic model.
MALLET provides furthermore an example:
// Run the model for 50 iterations and stop (this is for testing only,
// for real applications, use 1000 to 2000 iterations)
model.setNumIterations(50);
It is obvious that too few iterations lead to bad topic models.
However, does increasing the number of Gibbs sampling iterations necessarily benefit the quality of the topic model (measured by perplexity, topic coherence or on a downstream task)?
Or is it possible that the model quality decreases with the --num-iterations set to a too high value?
On a personal project, averaged over 10-fold cross-validation increasing the number of iterations from 100 to 1000 did not impact the average accuracy (measured as Mean Reciprocal Rank) for a downstream task. However, within the cross-validation splits the performance changed significantly, although the random seed was fixed and all other parameters kept the same. What part of background knowledge about Gibbs sampling am I missing to explain this behavior?
I am using a symmetric prior for alpha and beta without hyperparameter optimization and the parallelized LDA implementation provided by MALLET.
The 1000 iteration setting is designed to be a safe number for most collection sizes, and also to communicate "this is a large, round number, so don't think it's very precise". It's likely that smaller numbers will be fine. I once ran a model for 1000000 iterations, and fully half the token assignments never changed from the 1000 iteration model.
Could you be more specific about the cross validation results? Was it that different folds had different MRRs, which were individually stable over iteration counts? Or that individual fold MRRs varied by iteration count, but they balanced out in the overall mean? It's not unusual for different folds to have different "difficulty". Fixing the random seed also wouldn't make a difference if the data is different.
I am trying to build a deep neural networks that takes in a set of documents and predicts the category it belongs.
Since number of documents in each collection is not fixed, my first attempt was to get a mapping of documents from doc2vec and use the average.
The accuracy on training is high as 90% but the testing accuracy is low as 60%.
Is there a better way of representing a collection of documents as a fixed length vector so that the words they have in common are captured?
The description of your process so far is a bit vague and unclear – you may want to add more detail to your question.
Typically, Doc2Vec would convert each doc to a vector, not "a collection of documents".
If you did try to collapse a collection into a single vector – for example, by averaging many doc-vecs, or calculating a vector for a synthetic document with all the sub-documents' words – you might be losing valuable higher-dimensional structure.
To "predict the category" would be a typical "classification" problem, and with a bunch of documents (represented by their per-doc vectors) and known-labels, you could try various kinds of classifiers.
I suspect from your description, that you may just be collapsing a category to a single vector, then classifying new documents by checking which existing category-vector they're closest-to. That can work – it's vaguely a K-Nearest-Neighbors approach, but with every category reduced to one summary vector rather than the full set of known examples, and each classification being made by looking at just one single nearest-neighbor. That forces a simplicity on the process that may not match the "shapes" of the real categories as well as a true KNN classifier, or other classifiers, could achieve.
If accuracy on test data falls far below that observed during training, that can indicate that significant "overfitting" is occurring: the model(s) are essentially memorizing idiosyncrasies of the training data to "cheat" at answers based on arbitrary correlations, rather than learning generalizable rules. Making your model(s) smaller – such as by decreasing the dimensionality of your doc-vectors – may help in such situations, by giving the model less extra state in which to remember peculiarities of the training data. More data can also help - as the "noise" in more numerous varied examples tends of cancel itself out, rather than achieve the sort of misguided importance that can be learned in smaller datasets.
There are other ways to convert a variable-length text into a fixed-length vector, including many based on deeper learning algorithms. But, those can be even more training-data-hungry, and it seems like you may have other factors to improve before trying those in-lieu-of Doc2Vec.
The following is how I understand the point of parameter sharing in RNNs:
In regular feed-forward neural networks, every input unit is assigned an individual parameter, which means that the number of input units (features) corresponds to the number of parameters to learn. In processing e.g. image data, the number of input units is the same over all training examples (usually constant pixel size * pixel size * rgb frames).
However, sequential input data like sentences can come in highly varying lengths, which means that the number of parameters will not be the same depending on which example sentence is processed. That is why parameter sharing is necessary for efficiently processing sequential data: it makes sure that the model always has the same input size regardless of the sequence length, as it is specified in terms of transition from one state to another. It is thus possible to use the same transition function with the same weights (input to hidden weights, hidden to output weights, hidden to hidden weights) at every time step. The big advantage is that it allows generalization to sequence lengths that did not appear in the training set.
My questions are:
Is my understanding of RNNs, as summarized above, correct?
In the actual code example in Keras I looked at for LSTMs, they padded the sentences to equal lengths before all. By doing so, doesn't this wash away the whole purpose of parameter sharing in RNNs?
Parameter Sharing
Being able to efficiently process sequences of varying length is not the only advantage of parameter sharing. As you said, you can achieve that with padding. The main purpose of parameter sharing is a reduction of the parameters that the model has to learn. This is the whole purpose of using a RNN.
If you would learn a different network for each time step and feed the output of the first model to the second etc. you would end up with a regular feed-forward network. For a number of 20 time steps, you would have 20 models to learn. In Convolutional Nets, parameters are shared by the Convolutional Filters because when we can assume that there are similar interesting patterns in different regions of the picture (for example a simple edge). This drastically reduces the number of parameters we have to learn. Analogously, in sequence learning we can often assume that there are similar patterns at different time steps. Compare 'Yesterday I ate an apple' and 'I ate an apple yesterday'. These two sentences mean the same, but the 'I ate an apple' part occurs on different time steps. By sharing parameters, you only have to learn what that part means once. Otherwise, you'd have to learn it for every time step, where it could occur in your model.
There is a drawback to sharing the parameters. Because our model applies the same transformation to the input at every time step, it now has to learn a transformation that makes sense for all time steps. So, it has to remember, what word came in which time step, i.e. 'chocolate milk' should not lead to the same hidden and memory state as 'milk chocolate'. But this drawback is small compared to using a large feed-forward network.
Padding
As for padding the sequences: the main purpose is not directly to let the model predict sequences of varying length. Like you said, this can be done by using parameter sharing. Padding is used for efficient training - specifically to keep the computational graph during training low. Without padding, we have two options for training:
We unroll the model for each training sample. So, when we have a sequence of length 7, we unroll the model to 7 time steps, feed the sequence, do back-propagation through the 7 time steps and update the parameters. This seems intuitive in theory. But in practice, this is inefficient, because TensorFlow's computational graphs don't allow recurrency, they are feedforward.
The other option is to create the computational graphs before starting training. We let them share the same weights and create one computational graph for every sequence length in our training data. But when our dataset has 30 different sequence lengths this means 30 different graphs during training, so for large models, this is not feasible.
This is why we need padding. We pad all sequences to the same length and then only need to construct one computational graph before starting training. When you have both very short and very long sequence lengths (5 and 100 for example), you can use bucketing and padding. This means, you pad the sequences to different bucket lengths, for example [5, 20, 50, 100]. Then, you create a computational graph for each bucket. The advantage of this is, that you don't have to pad a sequence of length 5 to 100, as you would waste a lot of time on "learning" the 95 padding tokens in there.
I have a series of functions that are all designed to do the same thing. The same inputs produce the same outputs, but the time that it takes to do them varies by function. I want to determine which one is 'fastest', and I want to have some confidence that my measurement is 'statistically significant'.
Perusing Wikipedia and the interwebs tells me that statistical significance means that a measurement or group of measurements is different from a null hypothesis by a p-value threshold. How would that apply here? What is the null hypothesis between function A being faster than function B?
Once I've got that whole setup defined, how do I figure out when to stop measuring? I'll typically see that a benchmark is run three times, and then the average is reported; why three times and not five or seven? According to this page on Statistical Significance (which I freely admit I do not understand fully), Fisher used 8 as the number of samples that he needed to measure something with 98% confidence; why 8?
I would not bother applying statistics principles to benchmarking results. In general, the term "statistical significance" refers to the likelihood that your results were achieved accidentally, and do not represent an accurate assessment of the true values. In statistics, as a result of simple probability, the likelihood of a result being achieved by chance decreases as the number of measurements increases. In the benchmarking of computer code, it is a trivial matter to increase the number of trials (the "n" in statistics) so that the likelihood of an accidental result is below any arbitrary threshold you care to define (the "alpha" or level of statistical significance).
To simplify: benchmark by running your code a huge number of times, and don't worry about statistical measurements.
Note to potential down-voters of this answer: this answer is somewhat of a simplification of the matter, designed to illustrate the concepts in an accessible way. Comments like "you clearly don't understand statistics" will result in a savage beat-down. Remember to be polite.
You are asking two questions:
How do you perform a test of statistical significance that the mean time of function A is greater than the mean time of function B?
If you want a certain confidence in your answer, how many samples should you take?
The most common answer to the first question is that you either want to compute a confidence interval or perform a t-test. It's not different than any other scientific experiment with random variation. To compute the 95% confidence interval of the mean response time for function A simply take the mean and add 1.96 times the standard error to either side. The standard error is the square root of the variance divided by N. That is,
95% CI = mean +/- 1.96 * sqrt(sigma2/N))
where sigma2 is the variance of speed for function A and N is the number of runs you used to calculate mean and variance.
Your second question relates to statistical power analysis and the design of experiments. You describe a sequential setup where you are asking whether to continue sampling. The design of sequential experiments is actually a very tricky problem in statistics, since in general you are not allowed to calculate confidence intervals or p-values and then draw additional samples conditional on not reaching your desired significance. If you wish to do this, it would be wiser to set up a Bayesian model and calculate your posterior probability that speed A is greater than speed B. This, however, is massive overkill.
In a computing environment it is generally pretty trivial to achieve a very small confidence interval both because drawing large N is easy and because the variance is generally small -- one function obviously wins.
Given that Wikipedia and most online sources are still horrible when it comes to statistics, I recommend buying Introductory Statistics with R. You will learn both the statistics and the tools to apply what you learn.
The research you site sounds more like a highly controlled environment. This is purely a practical answer that has proven itself time and again to be effective for performance testing.
If you are benchmarking code in a modern, multi-tasking, multi-core, computing environment, the number of iterations required to achieve a useful benchmark goes up as the length of time of the operation to be measured goes down.
So, if you have an operation that takes ~5 seconds, you'll want, typically, 10 to 20 iterations. As long as the deviation across the iterations remains fairly constant, then your data is sound enough to draw conclusions. You'll often want to throw out the first iteration or two because the system is typically warming up caches, etc...
If you are testing something in the millisecond range, you'll want 10s of thousands of iterations. This will eliminate noise caused by other processes, etc, firing up.
Once you hit the sub-millisecond range -- 10s of nanoseconds -- you'll want millions of iterations.
Not exactly scientific, but neither is testing "in the real world" on a modern computing system.
When comparing the results, consider the difference in execution speed as percentage, not absolute. Anything less than about 5% difference is pretty close to noise.
Do you really care about statistical significance or plain old significance? Ultimately you're likely to have to form a judgement about readability vs performance - and statistical significance isn't really going to help you there.
A couple of rules of thumb I use:
Where possible, test for enough time to make you confident that little blips (like something else interrupting your test for a short time) won't make much difference. Usually I reckon 30 seconds is enough for this, although it depends on your app. The longer you test for, the more reliable the test will be - but obviously your results will be delayed :)
Running a test multiple times can be useful, but if you're timing for long enough then it's not as important IMO. It would alleviate other forms of error which made a whole test take longer than it should. If a test result looks suspicious, certainly run it again. If you see significantly different results for different runs, run it several more times and try to spot a pattern.
The fundamental question you're trying to answer is how likley is it that what you observe could have happened by chance? Is this coin fair? Throw it once: HEADS. No it's not fair it always comes down heads. Bad conclusion! Throw it 10 times and get 7 Heads, now what do you conclude? 1000 times and 700 heads?
For simple cases we can imagine how to figure out when to stop testing. But you have a slightly different situation - are you really doing a statistical analysis?
How much control do you have of your tests? Does repeating them add any value? Your computer is deterministic (maybe). Eistein's definition of insanity is to repeat something and expect a different outcome. So when you run your tests do you get repeatable answers? I'm not sure that statistical analyses help if you are doing good enough tests.
For what you're doing I would say that the first key thing is to make sure that you really are measuring what you think. Run every test for long enough that any startup or shutdown effects are hidden. Useful performance tests tend to run for quite extended periods for that reason. Make sure that you are not actually measuing the time in your test harness rather than the time in your code.
You have two primary variables: how many iterations of your method to run in one test? How many tests to run?
Wikipedia says this
In addition to expressing the
variability of a population, standard
deviation is commonly used to measure
confidence in statistical conclusions.
For example, the margin of error in
polling data is determined by
calculating the expected standard
deviation in the results if the same
poll were to be conducted multiple
times. The reported margin of error is
typically about twice the standard
deviation.
Hence if your objective is to be sure that one function is faster than another you could run a number of tests of each, compute the means and standard deviations. My expectation is that if your number of iterations within any one test is high then the standard deviation is going to be low.
If we accept that defintion of margin of error, you can see whether the two means are further apart than their total margin's of error.