I'm trying to minimize my function "FunctionToMinimize", which is defined as follows:
FunctionToMinimize[a_, b_, c_, d_] := (2.35*Sqrt[
Variance[1/2*
(a*#1 + b*#2 + c*#3 + d*#4)
]
]
/Mean[1/2*(a*#1 + b*#2 + c*#3 + d*#4)])
&[DataList1[[1 ;; 1000]],DataList2[[1 ;; 1000]],
DataList3[[1 ;; 1000]], DataList4[[1 ;; 1000]]]
The four parameters a,b,c and d are restricted to be somewhere between 0.5 and 1.5. My Problem is now, that if I call
NMinimize[{Funktion[w, x, y, z],
0.75 < w < 1.25 && 0.75 < y < 1.25 && 0.75 < x < 1.25 && 0.75 < z < 1.25},
{w, x, y, z}]
the Mathematica kernel shuts down because it has not enough memory. If I use only the first 100 entries in my DataLists, it will find me results (in 4.1 sec), but if I use DataList[[1;;1000]] or even more entries, the kernel crashes.
Has anybody an idea, why the NMinimize function uses so much memory? I would need to have the minimization for 150'000 events in each list...
Thanks for your answer,
Cheers,
Andreas
I would guess (but haven't in any way checked) that the problem is that on each call to your function, Mathematica is trying to construct a symbolic expression derived from all your data and that occupies much more memory than you'd expect.
Regardless, the good news -- if you haven't long since moved on and forgotten about this problem -- is that you can turn the function into something much simpler.
So, first of all, the 2.35 and the 1/2s just change your function by a constant factor and don't affect where the minimum is, so let's ignore them. Next, your function is always non-negative, so minimizing it is the same as minimizing its square, so let's do that.
So now you're trying to minimize var(aw+bx+cy+dz)/mean(aw+bx+cy+dz)^2 where w,x,y,z are (perhaps quite long) vectors.
Now your numerator and denominator are both just quadratic forms in a,b,c,d whose coefficients depend (in fixed ways) on those vectors. Specifically, suppose your vectors have length N. Then your function is just
[sum(aw+bx+cy+dz)^2/N - sum(aw+bx+cy+dz)^2/N^2] / (sum(aw+bx+cy+dz)^2/N^2)
which you might prefer to write as N sum(aw+bx+cy+dz)^2 / sum(aw+bx+cy+dz)^2 - 1
and in that fraction, e.g., the coefficient of bc in the numerator is 2 sum(xy), and the coefficient in the denominator is 2 sum(x) sum(y).
So you can take your big vectors, compute the relevant coefficients once, and then just ask Mathematica to optimize a function of the form (quadratic / quadratic), which should be pretty painless.
Related
I am a student working with time-series data which we feed into a neural network for classification (my task is to build and train this NN).
We're told to use a band-pass filter of 10 Hz to 150 Hz since anything outside that is not interesting.
After applying the band-pass, I've also down-sampled the data to 300 samples per second (originally it was 768 Hz). My understanding of the Shannon Nyquist sampling theorem is that, after applying the band-pass, any information in the data will be perfectly preserved at this sample-rate.
However, I got into a discussion with my supervisor who claimed that 300 Hz might not be sufficient even if the signal was band-limited. She says that it is only the minimum sample rate, not necessarily the best sample rate.
My understanding of the sampling theorem makes me think the supervisor is obviously wrong, but I don't want to argue with my supervisor, especially in case I'm actually the one who has misunderstood.
Can anyone help to confirm my understanding or provide some clarification? And how should I take this up with my supervisor (if at all).
The Nyquist-Shannon theorem states that the sampling frequency should at-least be twice of bandwidth, i.e.,
fs > 2B
So, this is the minimal criteria. If the sampling frequency is less than 2B then there will be aliasing. There is no upper limit on sampling frequency, but more the sampling frequency, the better will be the reconstruction.
So, I think your supervisor is right in saying that it is the minimal condition and not the best one.
Actually, you and your supervisor are both wrong. The minimum sampling rate required to faithfully represent a real-valued time series whose spectrum lies between 10 Hz and 150 Hz is 140 Hz, not 300 Hz. I'll explain this, and then I'll explain some of the context that shows why you might want to "oversample", as it is referred to (spoiler alert: Bailian-Low Theorem). The supervisor is mixing folklore into the discussion, and when folklore is not properly-contexted, it tends to telephone tag into fakelore. (That's a common failing even in the peer-reviewed literature, by the way). And there's a lot of fakelore, here, that needs to be defogged.
For the following, I will use the following conventions.
There's no math layout on Stack Overflow (except what we already have with UTF-8), so ...
a^b denotes a raised to the power b.
∫_I (⋯x⋯) dx denotes an integral of (⋯x⋯) taken over all x ∈ I, with the default I = ℝ.
The support supp φ (or supp_x φ(x) to make the "x" explicit) of a function φ(x) is the smallest closed set containing all the x-es for which φ(x) ≠ 0. For regularly-behaving (e.g. continuously differentiable) functions that means a union of closed intervals and/or half-rays or the whole real line, itself. This figures centrally in the Shannon-Nyquist sampling theorem, as its main condition is that a spectrum have bounded support; i.e. a "finite bandwidth".
For the Fourier transform I will use the version that has the 2π up in the exponent, and for added convenience, I will use the convention 1^x = e^{2πix} = cos(2πx) + i sin(2πx) (which I refer to as the Ramanujan Convention, as it is the convention I frequently used in my previous life oops I mean which Ramanujan secretly used in his life to make the math a whole lot simpler).
The set ℤ = {⋯, -2, -1, 0, +1, +2, ⋯ } is the integers, and 1^{x+z} = 1^x for all z∈ℤ - making 1^x the archetype of a periodic function whose period is 1.
Thus, the Fourier transform f̂(ν) of a function f(t) and its inverse are given by:
f̂(ν) = ∫ f(t) 1^{-νt} dt, f(t) = ∫ f̂(ν) 1^{+νt} dν.
The spectrum of the time series given by the function f(t) is the function f̂(ν) of the cyclic frequency ν, which is what is measured in Hertz (Hz.); t, itself, being measured in seconds. A common convention is to use the angular frequency ω = 2πν, instead, but that muddies the picture.
The most important example, with respect to the issue at hand, is the Fourier transform χ̂_Ω of the interval function given by χ_Ω(t) = 1 if t ∈ [-½Ω,+½Ω] and χ_Ω(t) = 0 else:
χ̂_Ω(t) = ∫_[-½Ω,+½Ω] 1^ν dν
= {1^{+½Ω} - 1^{-½Ω}}/{2πi}
= {2i sin πΩ}/{2πi}
= Ω sinc πΩ
which is where the function sinc x = (sin πx)/(πx) comes into play.
The cardinal form of the sampling theorem is that a function f(t) can be sampled over an equally-spaced sampled domain T ≡ { kΔt: k ∈ ℤ }, if its spectrum is bounded by supp f̂ ⊆ [-½Ω,+½Ω] ⊆ [-1/(2Δt),+1/(2Δt)], with the sampling given as
f(t) = ∑_{t'∈T} f(t') Ω sinc(Ω(t - t')) Δt.
So, this generally applies to [over-]sampling with redundancy factors 1/(ΩΔt) ≥ 1. In the special case where the sampling is tight with ΩΔt = 1, then it reduces to the form
f(t) = ∑_{t'∈T} f(t') sinc({t - t'}/Δt).
In our case, supp f̂ = [10 Hz., 150 Hz.] so the tightest fits are with 1/Δt = Ω = 300 Hz.
This generalizes to equally-spaced sampled domains of the form T ≡ { t₀ + kΔt: k ∈ ℤ } without any modification.
But it also generalizes to frequency intervals supp f̂ = [ν₋,ν₊] of width Ω = ν₊ - ν₋ and center ν₀ = ½ (ν₋ + ν₊) to the following form:
f(t) = ∑_{t'∈T} f(t') 1^{ν₀(t - t')} Ω sinc(Ω(t - t')) Δt.
In your case, you have ν₋ = 10 Hz., ν₊ = 150 Hz., Ω = 140 Hz., ν₀ = 80 Hz. with the condition Δt ≤ 1/140 second, a sampling rate of at least 140 Hz. with
f(t) = (140 Δt) ∑_{t'∈T} f(t') 1^{80(t - t')} sinc(140(t - t')).
where t and Δt are in seconds.
There is a larger context to all of this. One of the main places where this can be used is for transforms devised from an overlapping set of windowed filters in the frequency domain - a typical case in point being transforms for the time-scale plane, like the S-transform or the continuous wavelet transform.
Since you want the filters to be smoothly-windowed functions, without sharp corners, then in order for them to provide a complete set that adds up to a finite non-zero value over all of the frequency spectrum (so that they can all be normalized, in tandem, by dividing out by this sum), then their respective supports have to overlap.
(Edit: Generalized this example to cover both equally-spaced and logarithmic-spaced intervals.)
One example of such a set would be filters that have end-point frequencies taken from the set
Π = { p₀ (α + 1)ⁿ + β {(α + 1)ⁿ - 1} / α: n ∈ {0,1,2,⋯} }
So, for interval n (counting from n = 0), you would have ν₋ = p_n and ν₊ = p_{n+1}, where the members of Π are enumerated
p_n = p₀ (α + 1)ⁿ + β {(α + 1)ⁿ - 1} / α,
Δp_n = p_{n+1} - p_n = α p_n + β = (α p₀ + β)(α + 1)ⁿ,
n ∈ {0,1,2,⋯}
The center frequency of interval n would then be ν₀ = p_n + ½ Δp₀ (α + 1)ⁿ and the width would be Ω = Δp₀ (α + 1)ⁿ, but the actual support for the filter would overlap into a good part of the neighboring intervals, so that when you add up the filters that cover a given frequency ν the sum doesn't drop down to 0 as ν approaches any of the boundary points. (In the limiting case α → 0, this produces an equally-spaced frequency domain, suitable for an equalizer, while in the case β → 0, it produces a logarithmic scale with base α + 1, where octaves are equally-spaced.)
The other main place where you may apply this is to time-frequency analysis and spectrograms. Here, the role of a function f and its Fourier transform f̂ are reversed and the role of the frequency bandwidth Ω is now played by the (reciprocal) time bandwidth 1/Ω. You want to break up a time series, given by a function f(t) into overlapping segments f̃(q,λ) = g(λ)* f(q + λ), with smooth windowing given by the functions g(λ) with bounded support supp g ⊆ [-½ 1/Ω, +½ 1/Ω], and with interval spacing Δq much larger than the time sampling Δt (the ratio Δq/Δt is called the "hop" factor). The analogous role of Δt is played, here, by the frequency interval in the spectrogram Δp = Ω, which is now constant.
Edit: (Fixed the numbers for the Audacity example)
The minimum sampling rate for both supp_λ g and supp_λ f(q,λ) is Δq = 1/Ω = 1/Δp, and the corresponding redundancy factor is 1/(ΔpΔq). Audacity, for instance, uses a redundancy factor of 2 for its spectrograms. A typical value for Δp might be 44100/2048 Hz., while the time-sampling rate is Δt = 1/(2×3×5×7)² second (corresponding to 1/Δt = 44100 Hz.). With a redundancy factor of 2, Δq would be 1024/44100 second and the hop factor would be Δq/Δt = 1024.
If you try to fit the sampling windows, in either case, to the actual support of the band-limited (or time-limited) function, then the windows won't overlap and the only way to keep their sum from dropping to 0 on the boundary points would be for the windowing functions to have sharp corners on the boundaries, which would wreak havoc on their corresponding Fourier transforms.
The Balian-Low Theorem makes the actual statement on the matter.
https://encyclopediaofmath.org/wiki/Balian-Low_theorem
And a shout-out to someone I've been talking with, recently, about DSP-related matters and his monograph, which provides an excellent introductory reference to a lot of the issues discussed here.
A Friendly Guide To Wavelets
Gerald Kaiser
Birkhauser 1994
He said it's part of a trilogy, another installment of which is forthcoming.
I'm writing a k-means algorithm. At each step, I want to compute the distance of my n points to k centroids, without a for loop, and for d dimensions.
The problem is I have a hard time splitting on my number of dimensions with the Matlab functions I know. Here is my current code, with x being my n 2D-points and y my k centroids (also 2D-points of course), and with the points distributed along dimension 1, and the spatial coordinates along the dimension 2:
dist = #(a,b) (a - b).^2;
dx = bsxfun(dist, x(:,1), y(:,1)'); % x is (n,1) and y is (1,k)
dy = bsxfun(dist, x(:,2), y(:,2)'); % so the result is (n,k)
dists = dx + dy; % contains the square distance of each points to the k centroids
[_,l] = min(dists, [], 2); % we then argmin on the 2nd dimension
How to vectorize furthermore ?
First edit 3 days later, searching on my own
Since asking this question I made progress on my own towards vectorizing this piece of code.
The code above runs in approximately 0.7 ms on my example.
I first used repmat to make it easy to do broadcasting:
dists = permute(permute(repmat(x,1,1,k), [3,2,1]) - y, [3,2,1]).^2;
dists = sum(dists, 2);
[~,l] = min(dists, [], 3);
As expected it is slightly slower since we replicate the matrix, it runs at 0.85 ms.
From this example it was pretty easy to use bsxfun for the whole thing, but it turned out to be extremely slow, running in 150 ms so more than 150 times slower than the repmat version:
dist = #(a, b) (a - b).^2;
dists = permute(bsxfun(dist, permute(x, [3, 2, 1]), y), [3, 2, 1]);
dists = sum(dists, 2);
[~,l] = min(dists, [], 3);
Why is it so slow ? Isn't vectorizing always an improvement on speed, since it uses vector instructions on the CPU ? I mean of course simple for loops could be optimized to use it aswell, but how can vectorizing make the code slower ? Did I do it wrong ?
Using a for loop
For the sake of completeness, here's the for loop version of my code, surprisingly the fastest running in 0.4 ms, not sure why..
for i=1:k
dists(:,i) = sum((x - y(i,:)).^2, 2);
endfor
[~,l] = min(dists, [], 2);
Note: This answer was written when the question was also tagged MATLAB. Links to Octave documentation added after the MATLAB tag was removed.
You can use the pdist2MATLAB/Octave function to calculate pairwise distances between two sets of observations.
This way, you offload the bother of vectorization to the people who wrote MATLAB/Octave (and they have done a pretty good job of it)
X = rand(10,3);
Y = rand(5,3);
D = pdist2(X, Y);
D is now a 10x5 matrix where the i, jth element is the distance between the ith X and jth Y point.
You can pass it the kind of distance you want as the third argument -- e.g. 'euclidean', 'minkowski', etc, or you could pass a function handle to your custom function like so:
dist = #(a,b) (a - b).^2;
D = pdist2(X, Y, dist);
As saastn mentions, pdist2(..., 'smallest', k) makes things easier in k-means. This returns just the smallest k values from each column of pdist2's result. Octave doesn't have this functionality, but it's easily replicated using sort()MATLAB/Octave.
D_smallest = sort(D);
D_smallest = D_smallest(1:k, :);
Given input signal x (e.g. a voltage, sampled thousand times per second couple of minutes long), I'd like to calculate e.g.
/ this is not q
y[3] = -3*x[0] - x[1] + x[2] + 3*x[3]
y[4] = -3*x[1] - x[2] + x[3] + 3*x[4]
. . .
I'm aiming for variable window length and weight coefficients. How can I do it in q? I'm aware of mavg and signal processing in q and moving sum qidiom
In the DSP world it's called applying filter kernel by doing convolution. Weight coefficients define the kernel, which makes a high- or low-pass filter. The example above calculates the slope from last four points, placing the straight line via least squares method.
Something like this would work for parameterisable coefficients:
q)x:10+sums -1+1000?2f
q)f:{sum x*til[count x]xprev\:y}
q)f[3 1 -1 -3] x
0n 0n 0n -2.385585 1.423811 2.771659 2.065391 -0.951051 -1.323334 -0.8614857 ..
Specific cases can be made a bit faster (running 0 xprev is not the best thing)
q)g:{prev[deltas x]+3*x-3 xprev x}
q)g[x]~f[3 1 -1 -3]x
1b
q)\t:100000 f[3 1 1 -3] x
4612
q)\t:100000 g x
1791
There's a kx white paper of signal processing in q if this area interests you: https://code.kx.com/q/wp/signal-processing/
This may be a bit old but I thought I'd weigh in. There is a paper I wrote last year on signal processing that may be of some value. Working purely within KDB, dependent on the signal sizes you are using, you will see much better performance with a FFT based convolution between the kernel/window and the signal.
However, I've only written up a simple radix-2 FFT, although in my github repo I do have the untested work for a more flexible Bluestein algorithm which will allow for more variable signal length. https://github.com/callumjbiggs/q-signals/blob/master/signal.q
If you wish to go down the path of performing a full manual convolution by a moving sum, then the best method would be to break it up into blocks equal to the kernel/window size (which was based on some work Arthur W did many years ago)
q)vec:10000?100.0
q)weights:30?1.0
q)wsize:count weights
q)(weights$(((wsize-1)#0.0),vec)til[wsize]+) each til count v
32.5931 75.54583 100.4159 124.0514 105.3138 117.532 179.2236 200.5387 232.168.
If your input list not big then you could use the technique mentioned here:
https://code.kx.com/q/cookbook/programming-idioms/#how-do-i-apply-a-function-to-a-sequence-sliding-window
That uses 'scan' adverb. As that process creates multiple lists which might be inefficient for big lists.
Other solution using scan is:
q)f:{sum y*next\[z;x]} / x-input list, y-weights, z-window size-1
q)f[x;-3 -1 1 3;3]
This function also creates multiple lists so again might not be very efficient for big lists.
Other option is to use indices to fetch target items from the input list and perform the calculation. This will operate only on input list.
q) f:{[l;w;i]sum w*l i+til 4} / w- weight, l- input list, i-current index
q) f[x;-3 -1 1 3]#'til count x
This is a very basic function. You can add more variables to it as per your requirements.
So I have something that looks like the following:
However, I am having real trouble integrating the data on the other side of this decision line to get my errors.
In general, given analytic form of the decision boundary you could compute the integrals exactly. However, why not use monte carlo which is fast, simple and generic (will work for any distributions and decision boundaries). All you have to do is repeatedly sample from your gaussians, check if the sampled point is on the correct side (N_c) or incorrect (N_i) and in the limit you will get your integrals from
INTEGRAL_of_distributions_being_on_correct_side ~ N_c / (N_c + N_i)
INTEGRAL_of_distributions_being_on_incorrect_side ~ N_i / (N_c + N_i)
thus in pseudo code:
N_c = 0
N_i = 0
for i=1 to N do
y ~ P({-, +}) # sample distribution
x ~ P(X|y) # sample point from given class
if side_of_decision(x) == y then
N_c += 1
else
N_i += 1
end
end
return N_c, N_i
In your case P({-, +}) is probably just 50-50 chance and P(X|-) and P(X|+) are your two Gaussians.
This has become quite a frustrating question, but I've asked in the Coursera discussions and they won't help. Below is the question:
I've gotten it wrong 6 times now. How do I normalize the feature? Hints are all I'm asking for.
I'm assuming x_2^(2) is the value 5184, unless I am adding the x_0 column of 1's, which they don't mention but he certainly mentions in the lectures when talking about creating the design matrix X. In which case x_2^(2) would be the value 72. Assuming one or the other is right (I'm playing a guessing game), what should I use to normalize it? He talks about 3 different ways to normalize in the lectures: one using the maximum value, another with the range/difference between max and mins, and another the standard deviation -- they want an answer correct to the hundredths. Which one am I to use? This is so confusing.
...use both feature scaling (dividing by the
"max-min", or range, of a feature) and mean normalization.
So for any individual feature f:
f_norm = (f - f_mean) / (f_max - f_min)
e.g. for x2,(midterm exam)^2 = {7921, 5184, 8836, 4761}
> x2 <- c(7921, 5184, 8836, 4761)
> mean(x2)
6676
> max(x2) - min(x2)
4075
> (x2 - mean(x2)) / (max(x2) - min(x2))
0.306 -0.366 0.530 -0.470
Hence norm(5184) = 0.366
(using R language, which is great at vectorizing expressions like this)
I agree it's confusing they used the notation x2 (2) to mean x2 (norm) or x2'
EDIT: in practice everyone calls the builtin scale(...) function, which does the same thing.
It's asking to normalize the second feature under second column using both feature scaling and mean normalization. Therefore,
(5184 - 6675.5) / 4075 = -0.366
Usually we normalize all of them to have zero mean and go between [-1, 1].
You can do that easily by dividing by the maximum of the absolute value and then remove the mean of the samples.
"I'm assuming x_2^(2) is the value 5184" is this because it's the second item in the list and using the subscript _2? x_2 is just a variable identity in maths, it applies to all rows in the list. Note that the highest raw mid-term exam result (i.e. that which is not squared) goes down on the final test and the lowest raw mid-term result increases the most for the final exam result. Theta is a fixed value, a coefficient, so somewhere your normalisation of x_1 and x_2 values must become (EDIT: not negative, less than 1) in order to allow for this behaviour. That should hopefully give you a starting basis, by identifying where the pivot point is.
I had the same problem, in my case the thing was that I was using as average the maximum x2 value (8836) minus minimum x2 value (4761) divided by two, instead of the sum of each x2 value divided by the number of examples.
For the same training set, I got the question as
Q. What is the normalized feature x^(3)_1?
Thus, 3rd training ex and 1st feature makes out to 94 in above table.
Now, normalized form is
x = (x - mean(x's)) / range(x)
Values are :
x = 94
mean(89+72+94+69) / 4 = 81
range = 94 - 69 = 25
Normalized x = (94 - 81) / 25 = 0.52
I'm taking this course at the moment and a really trivial mistake I made first time I answered this question was using comma instead of dot in the answer, since I did by hand and in my country we use comma to denote decimals. Ex:(0,52 instead of 0.52)
So in the second time I tried I used dot and works fine.