Is there a way to calculate odds ratios for a glmer model?
My model is defined as follows:
results_reduced <- glmer(R0A1 ~ MPHW_Perc + AG_Perc + Shrub_Perc + Dist_PrimaryRoads
+ Dist_SecondaryRoads + (1 | ID), data = secondorder_st,
family = binomial)
I would like to calculate odds ratios for this model with associated confidence intervals; however, the following syntax that I would use in GLM continues to run without stopping:
#odds ratios and 95% CI
exp(cbind(OR = coef(results_reduced), confint(results_reduced)))
Therefore, I assume that the random effect is causing some hang-up issue. Is there a way to output the odds ratio for a mixed-effects model?
The random effects probably aren't the problem. 1st, coef(glmm_model) pulls up a list of your fixed and random effects. You want fixef(glmm_model). 2nd.How large is your dataset? Calculating confidence intervals for glmm models takes time (sometimes minutes), so you might just not be waiting long enough. 3rd, For glmer the default way to calculate confidence intervals is "profile." Depending on your data it will often throw warnings, which might not cooperate when nested in your calls to cbind and exp. You could try confint(..., method ="Wald"), or doing things piece by piece.
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I am implementing program which solve differential equations - 1d diffusion.
I am using Crank - Nicolson method which accuracyshould be second-order for time and second-order for space.
Unfortunately my results are second-order for time and first-order for space.
How is that possible? Or i messed something and it is not possible?
I know f.e. that if it should be second-order and becomes third-order that means that T = Ahp + Bhp+1 and etc. if Ahp equals 0 then Bhp+1 will become dominant and we have p+1 order of accuracy.
I'm doing some performance/load testing of a service. Imagine the test function like this:
bytesPerSecond = test(filesize: 10MB, concurrency: 5)
Using this, I'll populate a table of results for different sizes and levels of concurrency. There are other variables too, but you get the idea.
The test function spins up concurrency requests and tracks throughput. This rate starts off at zero, then spikes and dips until it eventually stabilises on the 'true' value.
However it can take a while for this stability to occur, and there are lot of combinations of input to evaluate.
How can the test function decide when it's performed enough samples? By enough, I suppose I mean that the result isn't going to change beyond some margin if testing continues.
I remember reading an article about this a while ago (from one of the jsperf authors) that discussed a robust method, but I cannot find the article any more.
One simple method would be to compute the standard deviation over a sliding window of values. Is there a better approach?
IIUC, you're describing the classic problem of estimating the confidence interval of the mean with unknown variance. That is, suppose you have n results, x1, ..., xn, where each of the xi is a sample from some process of which you don't know much: not the mean, not the variance, and not the distribution's shape. For some required confidence interval, you'd like to now whether n is large enough so that, with high probability the true mean is within the interval of your mean.
(Note that with relatively-weak conditions, the Central Limit Theorem guarantees that the sample mean will converge to a normal distribution, but to apply it directly you would need the variance.)
So, in this case, the classic solution to determine if n is large enough, is as follows:
Start by calculating the sample mean μ = ∑i [xi] / n. Also calculate the normalized sample variance s2 = ∑i [(xi - μ)2] / (n - 1)
Depending on the size of n:
If n > 30, the confidence interval is approximated as μ ± zα / 2(s / √(n)), where, if necessary, you can find here an explanation on the z and α.
If n < 30, the confidence interval is approximated as μ ± tα / 2(s / √(n)); see again here an explanation of the t value, as well as a table.
If the confidence is enough, stop. Otherwise, increase n.
Stability means rate of change (derivative) is zero or close to zero.
The test function spins up concurrency requests and tracks throughput.
This rate starts off at zero, then spikes and dips until it eventually
stabilises on the 'true' value.
I would track your past throughput values. For example last X values or so. According to this values, I would calculate rate of change (derivative of your throughput). If your derivative is close to zero, then your test is stable. I will stop test.
How to find X? I think instead of constant value, such as 10, choosing a value according to maximum number of test can be more suitable, for example:
X = max(10,max_test_count * 0.01)
I'm trying to implement an Inertial Navigation System using an Indirect Kalman Filter. I've found many publications and thesis on this topic, but not too much code as example. For my implementation I'm using the Master Thesis available at the following link:
https://fenix.tecnico.ulisboa.pt/downloadFile/395137332405/dissertacao.pdf
As reported at page 47, the measured values from inertial sensors equal the true values plus a series of other terms (bias, scale factors, ...).
For my question, let's consider only bias.
So:
Wmeas = Wtrue + BiasW (Gyro meas)
Ameas = Atrue + BiasA. (Accelerometer meas)
Therefore,
when I propagate the Mechanization equations (equations 3-29, 3-37 and 3-41)
I should use the "true" values, or better:
Wmeas - BiasW
Ameas - BiasA
where BiasW and BiasA are the last available estimation of the bias. Right?
Concerning the update phase of the EKF,
if the measurement equation is
dzV = VelGPS_est - VelGPS_meas
the H matrix should have an identity matrix in corrispondence of the velocity error state variables dx(VEL) and 0 elsewhere. Right?
Said that I'm not sure how I have to propagate the state variable after update phase.
The propagation of the state variable should be (in my opinion):
POSk|k = POSk|k-1 + dx(POS);
VELk|k = VELk|k-1 + dx(VEL);
...
But this didn't work. Therefore I've tried:
POSk|k = POSk|k-1 - dx(POS);
VELk|k = VELk|k-1 - dx(VEL);
that didn't work too... I tried both solutions, even if in my opinion the "+" should be used. But since both don't work (I have some other error elsewhere)
I would ask you if you have any suggestions.
You can see a snippet of code at the following link: http://pastebin.com/aGhKh2ck.
Thanks.
The difficulty you're running into is the difference between the theory and the practice. Taking your code from the snippet instead of the symbolic version in the question:
% Apply corrections
Pned = Pned + dx(1:3);
Vned = Vned + dx(4:6);
In theory when you use the Indirect form you are freely integrating the IMU (that process called the Mechanization in that paper) and occasionally running the IKF to update its correction. In theory the unchecked double integration of the accelerometer produces large (or for cheap MEMS IMUs, enormous) error values in Pned and Vned. That, in turn, causes the IKF to produce correspondingly large values of dx(1:6) as time evolves and the unchecked IMU integration runs farther and farther away from the truth. In theory you then sample your position at any time as Pned +/- dx(1:3) (the sign isn't important -- you can set that up either way). The important part here is that you are not modifying Pned from the IKF because both are running independent from each other and you add them together when you need the answer.
In practice you do not want to take the difference between two enourmous double values because you will lose precision (because many of the bits of the significand were needed to represent the enormous part instead of the precision you want). You have grasped that in practice you want to recursively update Pned on each update. However, when you diverge from the theory this way, you have to take the corresponding (and somewhat unobvious) step of zeroing out your correction value from the IKF state vector. In other words, after you do Pned = Pned + dx(1:3) you have "used" the correction, and you need to balance the equation with dx(1:3) = dx(1:3) - dx(1:3) (simplified: dx(1:3) = 0) so that you don't inadvertently integrate the correction over time.
Why does this work? Why doesn't it mess up the rest of the filter? As it turns out, the KF process covariance P does not actually depend on the state x. It depends on the update function and the process noise Q and so on. So the filter doesn't care what the data is. (Now that's a simplification, because often Q and R include rotation terms, and R might vary based on other state variables, etc, but in those cases you are actually using state from outside the filter (the cumulative position and orientation) not the raw correction values, which have no meaning by themselves).
I have a series of data and need to detect peak values in the series within a certain number of readings (window size) and excluding a certain level of background "noise." I also need to capture the starting and stopping points of the appreciable curves (ie, when it starts ticking up and then when it stops ticking down).
The data are high precision floats.
Here's a quick sketch that captures the most common scenarios that I'm up against visually:
One method I attempted was to pass a window of size X along the curve going backwards to detect the peaks. It started off working well, but I missed a lot of conditions initially not anticipated. Another method I started to work out was a growing window that would discover the longer duration curves. Yet another approach used a more calculus based approach that watches for some velocity / gradient aspects. None seemed to hit the sweet spot, probably due to my lack of experience in statistical analysis.
Perhaps I need to use some kind of a statistical analysis package to cover my bases vs writing my own algorithm? Or would there be an efficient method for tackling this directly with SQL with some kind of local max techniques? I'm simply not sure how to approach this efficiently. Each method I try it seems that I keep missing various thresholds, detecting too many peak values or not capturing entire events (reporting a peak datapoint too early in the reading process).
Ultimately this is implemented in Ruby and so if you could advise as to the most efficient and correct way to approach this problem with Ruby that would be appreciated, however I'm open to a language agnostic algorithmic approach as well. Or is there a certain library that would address the various issues I'm up against in this scenario of detecting the maximum peaks?
my idea is simple, after get your windows of interest you will need find all the peaks in this window, you can just compare the last value with the next , after this you will have where the peaks occur and you can decide where are the best peak.
I wrote one simple source in matlab to show my idea!
My example are in wave from audio file :-)
waveFile='Chick_eco.wav';
[y, fs, nbits]=wavread(waveFile);
subplot(2,2,1); plot(y); legend('Original signal');
startIndex=15000;
WindowSize=100;
endIndex=startIndex+WindowSize-1;
frame = y(startIndex:endIndex);
nframe=length(frame)
%find the peaks
peaks = zeros(nframe,1);
k=3;
while(k <= nframe - 1)
y1 = frame(k - 1);
y2 = frame(k);
y3 = frame(k + 1);
if (y2 > 0)
if (y2 > y1 && y2 >= y3)
peaks(k)=frame(k);
end
end
k=k+1;
end
peaks2=peaks;
peaks2(peaks2<=0)=nan;
subplot(2,2,2); plot(frame); legend('Get Window Length = 100');
subplot(2,2,3); plot(peaks); legend('Where are the PEAKS');
subplot(2,2,4); plot(frame); legend('Peaks in the Window');
hold on; plot(peaks2, '*');
for j = 1 : nframe
if (peaks(j) > 0)
fprintf('Local=%i\n', j);
fprintf('Value=%i\n', peaks(j));
end
end
%Where the Local Maxima occur
[maxivalue, maxi]=max(peaks)
you can see all the peaks and where it occurs
Local=37
Value=3.266296e-001
Local=51
Value=4.333496e-002
Local=65
Value=5.049438e-001
Local=80
Value=4.286804e-001
Local=84
Value=3.110046e-001
I'll propose a couple of different ideas. One is to use discrete wavelets, the other is to use the geographer's concept of prominence.
Wavelets: Apply some sort of wavelet decomposition to your data. There are multiple choices, with Daubechies wavelets being the most widely used. You want the low frequency peaks. Zero out the high frequency wavelet elements, reconstruct your data, and look for local extrema.
Prominence: Those noisy peaks and valleys are of key interest to geographers. They want to know exactly which of a mountain's multiple little peaks is tallest, the exact location of the lowest point in the valley. Find the local minima and maxima in your data set. You should have a sequence of min/max/min/max/.../min. (You might want to add an arbitrary end points that are lower than your global minimum.) Consider a min/max/min sequence. Classify each of these triples per the difference between the max and the larger of the two minima. Make a reduced sequence that replaces the smallest of these triples with the smaller of the two minima. Iterate until you get down to a single min/max/min triple. In your example, you want the next layer down, the min/max/min/max/min sequence.
Note: I'm going to describe the algorithmic steps as if each pass were distinct. Obviously, in a specific implementation, you can combine steps where it makes sense for your application. For the purposes of my explanation, it makes the text a little more clear.
I'm going to make some assumptions about your problem:
The windows of interest (the signals that you are looking for) cover a fraction of the entire data space (i.e., it's not one long signal).
The windows have significant scope (i.e., they aren't one pixel wide on your picture).
The windows have a minimum peak of interest (i.e., even if the signal exceeds the background noise, the peak must have an additional signal excess of the background).
The windows will never overlap (i.e., each can be examined as a distinct sub-problem out of context of the rest of the signal).
Given those, you can first look through your data stream for a set of windows of interest. You can do this by making a first pass through the data: moving from left to right, look for noise threshold crossing points. If the signal was below the noise floor and exceeds it on the next sample, that's a candidate starting point for a window (vice versa for the candidate end point).
Now make a pass through your candidate windows: compare the scope and contents of each window with the values defined above. To use your picture as an example, the small peaks on the left of the image barely exceed the noise floor and do so for too short a time. However, the window in the center of the screen clearly has a wide time extent and a significant max value. Keep the windows that meet your minimum criteria, discard those that are trivial.
Now to examine your remaining windows in detail (remember, they can be treated individually). The peak is easy to find: pass through the window and keep the local max. With respect to the leading and trailing edges of the signal, you can see n the picture that you have a window that's slightly larger than the actual point at which the signal exceeds the noise floor. In this case, you can use a finite difference approximation to calculate the first derivative of the signal. You know that the leading edge will be somewhat to the left of the window on the chart: look for a point at which the first derivative exceeds a positive noise floor of its own (the slope turns upwards sharply). Do the same for the trailing edge (which will always be to the right of the window).
Result: a set of time windows, the leading and trailing edges of the signals and the peak that occured in that window.
It looks like the definition of a window is the range of x over which y is above the threshold. So use that to determine the size of the window. Within that, locate the largest value, thus finding the peak.
If that fails, then what additional criteria do you have for defining a region of interest? You may need to nail down your implicit assumptions to more than 'that looks like a peak to me'.
These questions regard a set of data with lists of tasks performed in succession and the total time required to complete them. I've been wondering whether it would be possible to determine useful things about the tasks' lengths, either as they are or with some initial guesstimation based on appropriate domain knowledge. I've come to think graph theory would be the way to approach this problem in the abstract, and have a decent basic grasp of the stuff, but I'm unable to know for certain whether I'm on the right track. Furthermore, I think it's a pretty interesting question to crack. So here we go:
Is it possible to determine the weights of edges in a directed weighted graph, given a list of walks in that graph with the lengths (summed weights) of said walks? I recognize the amount and quality of permutations on the routes taken by the walks will dictate the quality of any possible answer, but let's assume all possible walks and their lengths are given. If a definite answer isn't possible, what kind of things can be concluded about the graph? How would you arrive at those conclusions?
What if there were several similar walks with possibly differing lengths given? Can you calculate a decent average (or other illustrative measure) for each edge, given enough permutations on different routes to take? How will discounting some permutations from the available data set affect the calculation's accuracy?
Finally, what if you had a set of initial guesses as to the weights and had to refine those using the walks given? Would that improve upon your guesstimation ability, and how could you apply the extra information?
EDIT: Clarification on the difficulties of a plain linear algebraic approach. Consider the following set of walks:
a = 5
b = 4
b + c = 5
a + b + c = 8
A matrix equation with these values is unsolvable, but we'd still like to estimate the terms. There might be some helpful initial data available, such as in scenario 3, and in any case we can apply knowledge of the real world - such as that the length of a task can't be negative. I'd like to know if you have ideas on how to ensure we get reasonable estimations and that we also know what we don't know - eg. when there's not enough data to tell a from b.
Seems like an application of linear algebra.
You have a set of linear equations which you need to solve. The variables being the lengths of the tasks (or edge weights).
For instance if the tasks lengths were t1, t2, t3 for 3 tasks.
And you are given
t1 + t2 = 2 (task 1 and 2 take 2 hours)
t1 + t2 + t3 = 7 (all 3 tasks take 7 hours)
t2 + t3 = 6 (tasks 2 and 3 take 6 hours)
Solving gives t1 = 1, t2 = 1, t3 = 5.
You can use any linear algebra techniques (for eg: http://en.wikipedia.org/wiki/Gaussian_elimination) to solve these, which will tell you if there is a unique solution, no solution or an infinite number of solutions (no other possibilities are possible).
If you find that the linear equations do not have a solution, you can try adding a very small random number to some of the task weights/coefficients of the matrix and try solving it again. (I believe falls under Perturbation Theory). Matrices are notorious for radically changing behavior with small changes in the values, so this will likely give you an approximate answer reasonably quickly.
Or maybe you can try introducing some 'slack' task in each walk (i.e add more variables) and try to pick the solution to the new equations where the slack tasks satisfy some linear constraints (like 0 < s_i < 0.0001 and minimize sum of s_i), using Linear Programming Techniques.
Assume you have an unlimited number of arbitrary characters to represent each edge. (a,b,c,d etc)
w is a list of all the walks, in the form of 0,a,b,c,d,e etc. (the 0 will be explained later.)
i = 1
if #w[i] ~= 1 then
replace w[2] with the LENGTH of w[i], minus all other values in w.
repeat forever.
Example:
0,a,b,c,d,e 50
0,a,c,b,e 20
0,c,e 10
So:
a is the first. Replace all instances of "a" with 50, -b,-c,-d,-e.
New data:
50, 50
50,-b,-d, 20
0,c,e 10
And, repeat until one value is left, and you finish! Alternatively, the first number can simply be subtracted from the length of each walk.
I'd forget about graphs and treat lists of tasks as vectors - every task represented as a component with value equal to it's cost (time to complete in this case.
In tasks are in different orderes initially, that's where to use domain knowledge to bring them to a cannonical form and assign multipliers if domain knowledge tells you that the ratio of costs will be synstantially influenced by ordering / timing. Timing is implicit initial ordering but you may have to make a function of time just for adjustment factors (say drivingat lunch time vs driving at midnight). Function might be tabular/discrete. In general it's always much easier to evaluate ratios and relative biases (hardnes of doing something). You may need a functional language to do repeated rewrites of your vectors till there's nothing more that romain knowledge and rules can change.
With cannonical vectors consider just presence and absence of task (just 0|1 for this iteratioon) and look for minimal diffs - single task diffs first - that will provide estimates which small number of variables. Keep doing this recursively, be ready to back track and have a heuristing rule for goodness or quality of estimates so far. Keep track of good "rounds" that you backtraced from.
When you reach minimal irreducible state - dan't many any more diffs - all vectors have the same remaining tasks then you can do some basic statistics like variance, mean, median and look for big outliers and ways to improve initial domain knowledge based estimates that lead to cannonical form. If you finsd a lot of them and can infer new rules, take them in and start the whole process from start.
Yes, this can cost a lot :-)