Someone will answer a series of questions and will mark each important (I), very important (V), or extremely important (E). I'll then match their answers with answers given by everyone else, compute the percent of the answers in each bucket that are the same, then combine the percentages to get a final score.
For example, I answer 10 questions, marking 3 as extremely important, 5 as very important, and 2 as important. I then match my answers with someone else's, and they answer the same to 2/3 extremely important questions, 4/5 very important questions, and 2/2 important questions. This results in percentages of 66.66 (extremely important), 80.00 (very important), and 100.00 (important). I then combine these 3 percentages to get a final score, but I first weigh each percentage to reflect the importance of each bucket. So the result would be something like: score = E * 66.66 + V * 80.00 + I * 100.00. The values of E, V, and I (the weights) are what I'm trying to figure out how to calculate.
The following are the constraints present:
1 + X + X^2 = X^3
E >= X * V >= X^2 * I > 0
E + V + I = 1
E + 0.9 * V >= 0.9
0.9 > 0.9 * E + 0.75 * V >= 0.75
E + I < 0.75
When combining the percentages, I could give important a weight of 0.0749, very important a weight of .2501, and extremely important a weight of 0.675, but this seems arbitrary, so I'm wondering how to go about calculating the optimal value for each weight. Also, how do I calculate the optimal weights if I ignore all constraints?
As far as what I mean by optimal: while adhering to the last 4 constraints, I want the weight of each bucket to be the maximum possible value, while having the weights be as far apart as possible (extremely important questions weighted maximally more than very important questions, and very important questions weighted maximally more than important questions).
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 was trying to calculate the AUC using MySQL for the data in table like below:
y p
1 0.872637
0 0.130633
0 0.098054
...
...
1 0.060190
0 0.110938
I came across the following SQL query which is giving the correct AUC score (I verified using sklearn method).
SELECT (sum(y*r) - 0.5*sum(y)*(sum(y)+1)) / (sum(y) * sum(1-y)) AS auc
FROM (
SELECT y, row_number() OVER (ORDER BY p) r
FROM probs
) t
Using pandas this can be done as follows:
temp = df.sort_values(by="p")
temp['r'] = np.arange(1, len(df)+1, 1)
temp['yr'] = temp['y']*temp['r']
print( (sum(temp.yr) - 0.5*sum(temp.y)*(sum(temp.y)+1)) / (sum(temp.y) * sum(1-temp.y)) )
I did not understand how we are able to calculate AUC using this method. Can somebody please give intuition behind this?
I am already familiar with the trapezoidal method which involves summing the area of small trapezoids under the ROC curve.
Short answer: it is Wilcoxon-Mann-Whitney statistic, see
https://en.wikipedia.org/wiki/Receiver_operating_characteristic#Area_under_the_curve
The page has proof as well.
The bottom part of your formula is identical to the formula in the wiki. The top part is trickier. f in wiki corresponds to p in your data and t_0 and t_1 are indexes in the data frame. Note that we first sort by p, which makes our life easier.
Note that the double sum may be decomposed as
Sum_{t_1 such that y(t_1)=1} #{t_0 such that p(t_0) < p(t_1) and y(t_0)=0}
Here # stands for the total number of such indexes.
For each row index t_1 (such that y(t_1) =1), how many t_0 are such that p(t_0) < p(t_1) and y(t_0)=0? We know that there are exactly t_1 values of p that are less or equal than t_1 because values are sorted. We conclude that
#{t_0: p(t_0) < p(t_1) and y(t_0)=1) = t_1 - #{t_0: t_0 <= t_1 and y(t_0)=1}
Now imagine scrolling down the sorted dataframe. For the first time we meet y=1, #{t_0: t_0 <= t_1 and y(t_0)=1}=1, for the second time we meet y=1, the same quantity is 2, for the third time we meet y=1, the quantity is 3, and so on. Therefore, when we sum the equality over all indexes t_1 when y=1, we get
Sum_{t_1: y(t_1)=1}#{t_0: p(t_0) < p(t_1) and y(t_0)=1) = Sum_{t_1: y(t_1)=1} t_1 - (1 + 2 + 3 + ... + n),
where n is the total number of ones in y column. Now we need to do one more simplification. Note that
Sum_{t_1: y(t_1)=1} t_1 = Sum_{t_1: y(t_1)=1} t_1 y(t_1)
If y(t_1) is not one, it is zero. Therefore,
Sum_{t_1: y(t_1)=1} t_1 = Sum_{t_1: y(t_1)=1} t_1 y(t_1) = Sum_{t} t y(t)
Plugging this to our formula and using that
1 + 2+ 3 + ... + n = n(n+1)/2
finished the proof of the formula you found.
P.S. I think that posting this question on math or stats overflow would make more sense.
Chapter 4.5.2 of Elements of Statistical Learning
I don't understand what does it mean：
＂Since for any β and β0 satisfying these inequalities, any positively scaled
multiple satisfies them too, we can arbitrarily set ||β|| = 1/M.＂
Also, how does maximize M becomes minimize 1/2(||β||^2) ?
＂Since for any β and β0 satisfying these inequalities, any positively scaled multiple satisfies them too, we can arbitrarily set ||β|| = 1/M.＂
y_i(x_i' b + b0) >= M ||b||
thus for any c>0
y_i(x_i' [bc] + [b0c]) >= M ||bc||
thus you can always find such c that ||bc|| = 1/M, so we can focus only on b such that they have such norm (we simply limit the space of possible solutions because we know that scaling does not change much)
Also, how does maximize M becomes minimize 1/2(||β||^2) ?
We put ||b|| = 1/M, thus M=1/||b||
max_b M = max_b 1 / ||b||
now maximization of positive f(b) is equivalent of minimization of 1/f(b), so
min ||b||
and since ||b|| is positive, its minimization is equivalent to minimization of the square, as well as multiplied by 1/2 (this does not change the optimal b)
min 1/2 ||b||^2
I have been given this question to work on a solution. I'm struggling to get my head around the recursion. Some break down of the question would be very helpful.
Given that Pi can be estimated using the function 4 * (1 – 1/3 + 1/5 – 1/7 + …) with more terms giving greater accuracy, write a function that calculates Pi to an accuracy of 5 decimal places.
I have got some example code however I really don't understand where/why the variables are entered like this. Possible breakdown of this code and why it is not accurate would be appreciated.
-module (pi).
-export ([pi/0]).
pi() -> 4 * pi(0,1,1).
pi(T,M,D) ->
A = 1 / D,
if
A > 0.00001 -> pi(T+(M*A), M*-1, D+2);
true -> T
end.
The formula comes from the evaluation of tg(pi/4) which is equal to 1. The inverse:
pi/4 = arctg(1)
so
pi = 4* arctg(1).
using the technique of the Taylor series:
arctg (x) = x - x^3/3 + ... + (-1)^n x^(2n+1)/(2n+1) + o(x^(2n+1))
so when x = 1 you get your formula:
pi = 4 * (1 – 1/3 + 1/5 – 1/7 + …)
the problem is to find an approximation of pi with an accuracy of 0.00001 (5 decimal). Lookinq at the formula, you can notice that
at each step (1/3, 1/5,...) the new term to add:
is smaller than the previous one,
has the opposite sign.
This means that each term is an upper estimation of the error (the term o(x^(2n+1))) between the real value of pi and the evaluation up to this term.
So it can be use to stop the recursion at a level where it is guaranty that the approximation is better than this term. To be correct, the program
you propose multiply the final result of the recursion by 4, so the error is no more guaranteed to be smaller than term.
looking at the code:
pi() -> 4 * pi(0,1,1).
% T = 0 is the initial estimation
% M = 1 is the sign
% D = 1 initial value of the term's index in the Taylor serie
pi(T,M,D) ->
A = 1 / D,
% evaluate the term value
if
A > 0.00001 -> pi(T+(M*A), M*-1, D+2);
% if the precision is not reach call the pi function with,
% new serie's evaluation (the previous one + sign * term): T+(M*A)
% new inverted sign: M*-1
% new index: D+2
true -> T
% if the precision is reached, give the result T
end.
To be sure that you have reached the right accuracy, I propose to replace A > 0.00001 by A > 0.0000025 (= 0.00001/4)
I can't find any error in this code, but I can't test it right now, anyway:
T is probably "total", M is "multiplicator", and D is "divisor".
By every step you:
check (the 'if' is in some way similar to a switch/case in c/c++/java) if the next term (A = 1/D) is bigger than 0.00001. If not, you can stop the recursion, you've got the 5 decimal places you were looking for. So "if true (default case) -> return T"
if it's bigger, you multiply A by M, add to the total, then multiply M by -1, add 2 to D, and repeat (so you get the next term, add again, and so on).
pi(T,M,D) ->
A = 1 / D,
if
A > 0.00001 -> pi(T+(M*A), M*-1, D+2);
true -> T
end.
I don't know Erlang myself but from the looks of it you are checking if 1/D is < 0.00001 when in reality you should be checking 4 * 1/D because that 4 is going to be multiplied through. For example in your case if 1/D was 0.000003 you would stop four function, but your total would actually have changed by 0.000012. Hope this helps.
I want to find the standard deviation:
Minimum = 5
Mean = 24
Maximum = 84
Overall score = 90
I just want to find out my grade by using the standard deviation
Thanks,
A standard deviation cannot in general be computed from just the min, max, and mean. This can be demonstrated with two sets of scores that have the same min, and max, and mean but different standard deviations:
1 2 4 5 : min=1 max=5 mean=3 stdev≈1.5811
1 3 3 5 : min=1 max=5 mean=3 stdev≈0.7071
Also, what does an 'overall score' of 90 mean if the maximum is 84?
I actually did a quick-and-dirty calculation of the type M Rad mentions. It involves assuming that the distribution is Gaussian or "normal." This does not apply to your situation but might help others asking the same question. (You can tell your distribution is not normal because the distance from mean to max and mean to min is not close). Even if it were normal, you would need something you don't mention: the number of samples (number of tests taken in your case).
Those readers who DO have a normal population can use the table below to give a rough estimate by dividing the difference of your measured minimum and your calculated mean by the expected value for your sample size. On average, it will be off by the given number of standard deviations. (I have no idea whether it is biased - change the code below and calculate the error without the abs to get a guess.)
Num Samples Expected distance Expected error
10 1.55 0.25
20 1.88 0.20
30 2.05 0.18
40 2.16 0.17
50 2.26 0.15
60 2.33 0.15
70 2.38 0.14
80 2.43 0.14
90 2.47 0.13
100 2.52 0.13
This experiment shows that the "rule of thumb" of dividing the range by 4 to get the standard deviation is in general incorrect -- even for normal populations. In my experiment it only holds for sample sizes between 20 and 40 (and then loosely). This rule may have been what the OP was thinking about.
You can modify the following python code to generate the table for different values (change max_sample_size) or more accuracy (change num_simulations) or get rid of the limitation to multiples of 10 (change the parameters to xrange in the for loop for idx)
#!/usr/bin/python
import random
# Return the distance of the minimum of samples from its mean
#
# Samples must have at least one entry
def min_dist_from_estd_mean(samples):
total = 0
sample_min = samples[0]
for sample in samples:
total += sample
sample_min = min(sample, sample_min)
estd_mean = total / len(samples)
return estd_mean - sample_min # Pos bec min cannot be greater than mean
num_simulations = 4095
max_sample_size = 100
# Calculate expected distances
sum_of_dists=[0]*(max_sample_size+1) # +1 so can index by sample size
for iternum in xrange(num_simulations):
samples=[random.normalvariate(0,1)]
while len(samples) <= max_sample_size:
sum_of_dists[len(samples)] += min_dist_from_estd_mean(samples)
samples.append(random.normalvariate(0,1))
expected_dist = [total/num_simulations for total in sum_of_dists]
# Calculate average error using that distance
sum_of_errors=[0]*len(sum_of_dists)
for iternum in xrange(num_simulations):
samples=[random.normalvariate(0,1)]
while len(samples) <= max_sample_size:
ave_dist = expected_dist[len(samples)]
if ave_dist > 0:
sum_of_errors[len(samples)] += \
abs(1 - (min_dist_from_estd_mean(samples)/ave_dist))
samples.append(random.normalvariate(0,1))
expected_error = [total/num_simulations for total in sum_of_errors]
cols=" {0:>15}{1:>20}{2:>20}"
print(cols.format("Num Samples","Expected distance","Expected error"))
cols=" {0:>15}{1:>20.2f}{2:>20.2f}"
for idx in xrange(10,len(expected_dist),10):
print(cols.format(idx, expected_dist[idx], expected_error[idx]))
Yo can obtain an estimate of the geometric mean, sometimes called the geometric mean of the extremes or GME, using the Min and the Max by calculating the GME= $\sqrt{ Min*Max }$. The SD can be then calculated using your arithmetic mean (AM) and the GME as:
SD= $$\frac{AM}{GME} * \sqrt{(AM)^2-(GME)^2 }$$
This approach works well for log-normal distributions or as long as the GME, GM or Median is smaller than the AM.
In principle you can make an estimate of standard deviation from the mean/min/max and the number of elements in the sample. The min and max of a sample are, if you assume normality, random variables whose statistics follow from mean/stddev/number of samples. So given the latter, one can compute (after slogging through the math or running a bunch of monte carlo scripts) a confidence interval for the former (like it is 80% probable that the stddev is between 20 and 40 or something like that).
That said, it probably isn't worth doing except in extreme situations.