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Using continued fractions, I'm generating integer ratios between frequencies to a certain precision, along with the error (difference from integer ratio to the real ratio). So I end up with things like:
101 Hz with 200 Hz = 1:2 + 0.0005
61 Hz with 92 Hz = 2:3 - 0.0036
However, I've run into a snag on actually deciding which of these will be more dissonant than others. At first I thought low numbers = better, but something like 1:51 would likely be not very dissonant since it's a frequency up 51 octaves from the other. It might be a screaming high, ear bleeding pitch, but I don't think it would have dissonance.
It seems to me that it must be related to the product of the two sides of the ratio compared to the constituents somehow. 1 * 51 = 51, which doesn't "go up much" from one side. 2 * 3 = 6, which I would think would indicate higher dissonance than 1:51. But I need to turn this feeling into an actual number, so I can compare 5:7 vs 3:8, or any other combinations.
And how could I work error into this? Certainly 1:2 + 0 would be less dissonant than 1:2 + 1. Would it be easier to apply an algorithm that works for the above integer ratios directly to the frequencies themselves? Or does having the integer ratio with an error allow for a simpler calculation?
edit: Thinking on it, an algorithm that could extend to any set of N frequencies in a chord would be awesome, but I get the feeling that would be much more difficult...
edit 2: Clarification:
Let's consider that I am dealing with pure sine waves, and either ignoring the specific thresholds of the human ear or abstracting them into variables. If there are severe complications, then they are ignored. My question is how it could be represented in an algorithm, in that case.
Have a look at Chapter 4 of http://homepages.abdn.ac.uk/mth192/pages/html/maths-music.html. From memory:
1) If two sine waves are just close enough for the human ear to be confused, but not so close that the human ear cannot tell they are different, there will be dissonance.
2) Pure sine waves are extremely rare - most tones have all sorts of harmonics. Dissonance is very likely to occur from colliding harmonics, rather than colliding main tones - to sort of follow your example, two tones many octaves apart are unlikely to be dissonant because their harmonics may not meet, whereas with just a couple of octaves different and loads of harmonics a flute could sound out of tune with a double bass. Therefore dissonance or not depends not only on the frequencies of the main tones, but on the harmonics present, and this has been experimentally demonstrated by constructing sounds with peculiar pseudo-harmonics.
The answer is in Chapter 4 of Music: a Mathematical Offering. In particular, see the following two figures:
consonance / dissonance plotted against the x critical bandwidth in 4.3. History of consonance and dissonance
dissonance vs. frequency in 4.5. Complex tones
Of course you still have to find a nice way to turn these data into a formula / program that gives you a measure of dissonance but I believe this gives you a good start. Good luck!
This will help:
http://www.acs.psu.edu/drussell/demos/superposition/superposition.html
You want to look at superposition.
Discrete or Fast Fourier Transform is the most generic means to get what you're asking for.
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Recently,I have been trying to understand how the Binary Extended Euclidean Algorithm works at the processor level. This question is all about finding an Inverse element in GF(2^m) with polynomial basis.
Generally I came across the Extended Euclidean Algorithm for evaluating an inverse element but the fact is that it involves too many addition and multiplication operations. The Binary EEA algorithm requires just bit shifting operations (equivalent to division by 2--logical shift right). The algorithm is in this link, page number 8.
In step 3 and 5 of this algorithm, every iteration shifts the parameters u and b by 1 bit to the right adding zero to the MSB at the same time. The loop ends when u == 1 and returns b. My question is how many primitive operations does a processor (say a 32 bit processor for example) perform in step 3 or step 5 of every iteration?
I came across barrel shifter and I am quite confused about how fast the shifting takes place. Should I really consider these primitive operations or should I ignore them if because the shifting may be faster?
It would really help me a lot if someone would show the primitive operations for the case where the size of u is 194 bits.
In case you might be wondering about the denominator x in step 3 and 5 of the algorithm, its the polynomial representation and x means nothing but 10 in binary and parameter u is an N-bit binary number.
There is no generic answer to this question: you can use portable code that will be tedious to optimize or highly machine specific code that will be even more complicated to optimize without breaking.
If you want real performance, you have to use MMX/AVX registers on the maximum width you can get your hands on. Intel provides lightweight wrappers on low-level instructions as macros and inline functions.
Always use unsigned types for your shifting operations to avoid unnecessary steps.
Usually ther is a "right shift" assembly OP code which is able to right shift a register a given number of bits. Such an operation takes one cycle.
This assumes thet your value is already loaded to the register however.
The best answer anyway: Implement this algorithm in a low level language (C, C++) and look at the assembly code produced by the compiler.
I found a web page describing all the existing windows for FFT. it's here:
http://en.wikipedia.org/wiki/Window_function
it's very interesting as it shows the frequency response depending on the window used.
So when i look at the freq responses, i found that Blackman-nuttall and Dolph–Chebyshev windows seems the best
but what is the best of the best ?
and are they really better for audio processing than Hamming or
Hanning?
many thanks
Jeff
Blow your mind here:
http://www.rssd.esa.int/SP/LISAPATHFINDER/docs/Data_Analysis/GH_FFT.pdf
I can tell you a couple of things on the matter.
There is no "best" window function because it depends on what your application is about. The common parameters on which you should focus your choice are:
Scalloping loss
Main lobe width (of a sine wave)
Sidelobes max level/decrease
Computational cost
For example, the simple rectangular window does not require any computational cost and it provides the thinnest possible lobe, but at the expense of a big scalloping and very noisy sidelobes.
Blackman-style windows are usually built to minimize sidelobe levels, but they tend to have a heavy scalloping. You might instead choose one of the so-called "flat-top" windows if you need more precise peak measurements since scalloping is usually less than 1% even with the simplest ones, but their lobes are very fat (6-10 bins in width perhaps).
Example Nuttall window in [0, 1]:
1 - 1.369982685*cos(z) + 0.4054102674*cos(2*z) - 0.03542758202*cos(3*z)
Example flat-top window (SFT3M) in [0, 1]:
1 - 1.84540464*cos(z) + 0.6962635*cos(2*z)
If there is a window function that has no scalloping loss, that is very narrow and with no sidelobes, then it would be extremely expensive to calculate.
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I have a math problem that I can't solve: I don't know how to find the value of n so that
365! / ((365-n)! * 365^n) = 50%.
I am using the Casio 500ms scientific calculator but I don't know how.
Sorry because my question is too easy, I am changing my career so I have to review and upgrade my math, the subject that I have neglected for years.
One COULD in theory use a root-finding scheme like Newton's method, IF you could take derivatives. But this function is defined only on the integers, since it uses factorials.
One way out is to recognize the identity
n! = gamma(n+1)
which will effectively allow you to extend the function onto the real line. The gamma function is defined on the positive real line, though it does have singularities at the negative integers. And of course, you still need the derivative of this expression, which can be done since gamma is differentiable.
By the way, a danger with methods like Newton's method on problems like this is it may still diverge into the negative real line. Choose poor starting values, and you may get garbage out. (I've not looked carefully at the shape of this function, so I won't claim for what set of starting values it will diverge on you.)
Is it worth jumping through the above set of hoops? Of course not. A better choice than Newton's method might be something like Brent's algorithm, or a secant method, which here will not require you to compute the derivative. But even that is a waste of effort.
Recognizing that this is indeed a problem on the integers, one could use a tool like bisection to resolve the solution extremely efficiently. It never requires derivatives, and it will work nicely enough on the integers. Once you have resolved the interval to be as short as possible, the algorithm will terminate, and take vary few function evaluations in the process.
Finally, be careful with this function, as it does involve some rather large factorials, which could easily overflow many tools to evaluate the factorial. For example, in MATLAB, if I did try to evaluate factorial(365):
factorial(365)
ans =
Inf
I get an overflow. I would need to move into a tool like the symbolic toolbox, or my own suite of variable precision integer tools. Alternatively, one could recognize that many of the terms in these factorials will cancel out, so that
365! / (365 - n)! = 365*(365-1)*(365-2)*...*(365-n+1)
The point is, we get an overflow for such a large value if we are not careful. If you have a tool that will not overflow, then use it, and use bisection as I suggested. Here, using the symbolic toolbox in MATLAB, I get a solution using only 7 function evaluations.
f = #(n) vpa(factorial(sym(365))/(factorial(sym(365 - n))*365^sym(n)));
f(0)
ans =
1.0
f(365)
ans =
1.4549552156187034033714015903853e-157
f(182)
ans =
0.00000000000000000000000095339164972764493041114884521295
f(91)
ans =
0.000004634800180846641815683109605743
f(45)
ans =
0.059024100534225072005461014516788
f(22)
ans =
0.52430469233744993108665513602619
f(23)
ans =
0.49270276567601459277458277166297
Or, if you can't take an option like that, but do have a tool that can evaluate the log of the gamma function, AND you have a rootfinder available as MATLAB does...
f = #(n) exp(gammaln(365+1) - gammaln(365-n + 1) - n*log(365));
fzero(#(n) f(n) - .5,10)
ans =
22.7677
As you can see here, I used the identity relating gamma and the factorial function, then used the log of the gamma function, in MATLAB, gammaln. Once all the dirty work was done, then I exponentiated the entire mess, which will be a reasonable number. Fzero tells us that the cross-over occurs between 22 and 23.
If a numerical approximation is ok, ask Wolfram Alpha:
n ~= -22.2298272...
n ~= 22.7676903...
I'm going to assume you have some special reason for wanting an actual algorithm, even though you only have one specific problem to solve.
You're looking for a value n where...
365! / ((365-n)! * 365^n) = 0.5
And therefore...
(365! / ((365-n)! * 365^n)) - 0.5 = 0.0
The general form of the problem is to find a value x such that f(x)=0. One classic algorithm for this kind of thing is the Newton-Raphson method.
[EDIT - as woodchips points out in the comment, the factorial is an integer-only function. My defence - for some problems (the birthday problem among them) it's common to generalise using approximation functions. I remember the Stirling approximation of factorials being used for the birthday problem - according to this, Knuth uses it. The Wikipedia page for the Birthday problem mentions several approximations that generalise to non-integer values.
It's certainly bad that I didn't think to mention this when I first wrote this answer.]
One problem with that is that you need the derivative of that function. That's more a mathematics issue, though you can estimate the derivative at any point by taking values a short distance either side.
You can also look at this as an optimisation problem. The general form of optimisation problems is to find a value x such that f(x) is maximised/minimised. In your case, you could define your function as...
f(x)=((365! / ((365-n)! * 365^n)) - 0.5)^2
Because of the squaring, the result can never be negative, so try to minimise. Whatever value of x gets you the smallest f(x) will also give you the result you want.
There isn't so much an algorithm for optimisation problems as a whole field - the method you use depends on the complexity of your function. However, this case should be simple so long as your language can cope with big numbers. Probably the simplest optimisation algorithm is called hill-climbing, though in this case it should probably be called rolling-down-the-hill. And as luck would have it, Newton-Raphson is a hill-climbing method (or very close to being one - there may be some small technicality that I don't remember).
[EDIT as mentioned above, this won't work if you need an integer solution for the problem as actually stated (rather than a real-valued approximation). Optimisation in the integer domain is one of those awkward issues that helps make optimisation a field in itself. The branch and bound is common for complex functions. However, in this case hill-climbing still works. In principle, you can even still use a tweaked version of Newton-Raphson - you just have to do some rounding and check that you don't keep rounding back to the same place you started if your moves are small.]
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Please recommend an error correcting algorithm for using very strange data channel.
The channel consists of two parts: Corrupter and Eraser.
Corrupter receives a word consisting of 10000 symbols in 3-symbol alphabet, say, {'a','b','c'}.
Corrupter changes each symbol with probability 10%.
Example:
Corrupter input: abcaccbacbbaaacbcacbcababacb...
Corrupter output: abcaacbacbbaabcbcacbcababccb...
Eraser receives corrupter output and erases each symbol with probability 94%.
Eraser produces word of the same length in 4-symbol alphabet {'a','b','c','*'}.
Example:
Eraser input: abcaacbacbbaabcbcacbcababccb...
Eraser output: *******a*****************c**...
So, on eraser output, approximately 6%*10000=600 symbols would not be erased, approximately 90%*600=540 of them would preserve their original values and approximately 60 would be corrupted.
What encoding-decoding algorithm with error correction is best suited for this channel?
What amount of useful data could be transmitted providing > 99.99% probability of successful decoding?
Is it possible to transmit 40 bytes of data through this channel? (256^40 ~ 3^200)
Here's something you can at least analyze:
Break your 40 bytes up into 13 25-bit chunks (with some wastage so this bit can obviously be improved)
2^25 < 3^16 so you can encode the 25 bits into 16 a/b/c "trits" - again wastage means scope for improvement.
With 10,000 trits available you can give each of your 13 encoded byte triples 769 output trits. Pick (probably at random) 769 different linear (mod 3) functions on 16 trits - each function is specified by 16 trits and you take a vector dot product between those trits and the 16 input trits. This gives you your 769 output trits.
Decode by considering all possible (2^25) chunks and pick the one which matches most of the surviving trits. You have some hope of getting the right answer as long as there are at least 16 surviving trits, which I think excel is telling me via BINOMDIST() happens often enough that there is a pretty good chance that it will happen for all of the 13 25-bit chunks.
I have no idea what error rate you get from garbling but random linear codes have a pretty good reputation, even if this one has a short blocksize because of my brain-dead decoding technique. At worst you could try simulating the encoding transmission and decoding of 25-bit chunks and work it out from there. You can get a slightly more accurate lower bound on error rate than above if you pretend that the garbling stage erases as well and so recalculate with a slightly higher probability of erasure.
I think this might actually work in practice if you can afford the 2^25 guesses per 25-bit block to decode. OTOH if this is a question in a class my guess is you need to demonstrate your knowledge of some less ad-hoc techniques already discussed in your class.
I apologize for being a bit verbose in advance: if you want to skip all the background mumbo jumbo you can see my question down below.
This is pretty much a follow up to a question I previously posted on how to compare two 1D (time dependent) signals. One of the answers I got was to use the cross-correlation function (xcorr in MATLAB), which I did.
Background information
Perhaps a little background information will be useful: I'm trying to implement an Independent Component Analysis algorithm. One of my informal tests is to (1) create the test case by (a) generate 2 random vectors (1x1000), (b) combine the vectors into a 2x1000 matrix (called "S"), and multiply this by a 2x2 mixing matrix (called "A"), to give me a new matrix (let's call it "T").
In summary: T = A * S
(2) I then run the ICA algorithm to generate the inverse of the mixing matrix (called "W"), (3) multiply "T" by "W" to (hopefully) give me a reconstruction of the original signal matrix (called "X")
In summary: X = W * T
(4) I now want to compare "S" and "X". Although "S" and "X" are 2x1000, I simply compare S(1,:) to X(1,:) and S(2,:) to X(2,:), each which is 1x1000, making them 1D signals. (I have another step which makes sure that these vectors are the proper vectors to compare to each other and I also normalize the signals).
So my current quandary is how to 'grade' how close S(1,:) matches to X(1,:), and likewise with S(2,:) to X(2,:).
So far I have used something like: r1 = max(abs(xcorr(S(1,:), X(1,:)))
My question
Assuming that using the cross correlation function is a valid way to go about comparing the similarity of two signals, what would be considered a good R value to grade the similarity of the signals? Wikipedia states that this is a very subjective area, and so I defer to the better judgment of those who might have experience in this field.
As you might realize, I'm not coming from a EE/DSP/statistical background at all (I'm a medical student) so I'm going through a sort of "baptism through fire" right now, and I appreciate all the help I can get. Thanks!
(edit: as far as directly answering your question about R values, see below)
One way to approach this would be to use cross-correlation. Bear in mind that you have to normalize amplitudes and correct for delays: if you have signal S1, and signal S2 is identical in shape, but half the amplitude and delayed by 3 samples, they're still perfectly correlated.
For example:
>> t = 0:0.001:1;
>> y = #(t) sin(10*t).*exp(-10*t).*(t > 0);
>> S1 = y(t);
>> S2 = 0.4*y(t-0.1);
>> plot(t,S1,t,S2);
These should have a perfect correlation coefficient. A way to compute this is to use maximum cross-correlation:
>> f = #(S1,S2) max(xcorr(S1,S2));
f =
#(S1,S2) max(xcorr(S1,S2))
>> disp(f(S1,S1)); disp(f(S2,S2)); disp(f(S1,S2));
12.5000
2.0000
5.0000
The maximum value of xcorr() takes care of the time-delay between signals. As far as correcting for amplitude goes, you can normalize the signals so that their self-cross-correlation is 1.0, or you can fold that equivalent step into the following:
ρ2 = f(S1,S2)2 / (f(S1,S1)*f(S2,S2);
In this case ρ2 = 5 * 5 / (12.5 * 2) = 1.0
You can solve for ρ itself, i.e. ρ = f(S1,S2)/sqrt(f(S1,S1)*f(S2,S2)), just bear in mind that both 1.0 and -1.0 are perfectly correlated (-1.0 has opposite sign)
Try it on your signals!
with respect to what threshold to use for acceptance/rejection, that really depends on what kind of signals you have. 0.9 and above is fairly good but can be misleading. I would consider looking at the residual signal you get after you subtract out the correlated version. You could do this by looking at the time index of the maximum value of xcorr():
>> t = 0:0.001:1;
>> y = #(a,t) sin(a*t).*exp(-a*t).*(t > 0);
>> S1=y(10,t);
>> S2=0.4*y(9,t-0.1);
>> f(S1,S2)/sqrt(f(S1,S1)*f(S2,S2))
ans =
0.9959
This looks pretty darn good for a correlation. But let's try fitting S2 with a scaled/shifted multiple of S1:
>> [A,i]=max(xcorr(S1,S2)); tshift = i-length(S1);
>> S2fit = zeros(size(S2)); S2fit(1-tshift:end) = A/f(S1,S1)*S1(1:end+tshift);
>> plot(t,[S2; S2fit]); % fit S2 using S1 as a basis
>> plot(t,[S2-S2fit]); % residual
Residual has some energy in it; to get a feel for how much, you can use this:
>> S2res=S2-S2fit;
>> dot(S2res,S2res)/dot(S2,S2)
ans =
0.0081
>> sqrt(dot(S2res,S2res)/dot(S2,S2))
ans =
0.0900
This says that the residual has about 0.81% of the energy (9% of the root-mean-square amplitude) of the original signal S2. (the dot product of a 1D signal with itself will always be equal to the maximum value of cross-correlation of that signal with itself.)
I don't think there's a silver bullet for answering how similar two signals are with each other, but hopefully I've given you some ideas that might be applicable to your circumstances.
A good starting point is to get a sense of what a perfect match will look like by calculating the auto-correlations for each signal (i.e. do the "cross-correlation" of each signal with itself).
THIS IS A COMPLETE GUESS - but I'm guessing max(abs(xcorr(S(1,:),X(1,:)))) > 0.8 implies success. Just out of curiosity, what kind of values do you get for max(abs(xcorr(S(1,:),X(2,:))))?
Another approach to validate your algorithm might be to compare A and W. If W is calculated correctly, it should be A^-1, so can you calculate a measure like |A*W - I|? Maybe you have to normalize by the trace of A*W.
Getting back to your original question, I come from a DSP background, so I get to deal with fairly noise-free signals. I understand that's not a luxury you get in biology :) so my 0.8 guess might be very optimistic. Perhaps looking at some literature in your field, even if they aren't using cross-correlation exactly, might be useful.
Usually in such cases people talk about "false acceptance rate" and "false rejection rate".
The first one describes how many times algorithm says "similar" for non-similar signals, the second one is the opposite.
Selecting a threshold thus becomes a trade-off between these criteria. To make FAR=0, threshold should be 1, to make FRR=0 threshold should be -1.
So probably, you will need to decide which trade-off between FAR and FRR is acceptable in your situation and this will give the right value for threshold.
Mathematically this can be expressed in different ways. Just a couple of examples:
1. fix some of rates at acceptable value and minimize other one
2. minimize max(FRR,FAR)
3. minimize aFRR+bFAR
Since they should be equal, the correlation coefficient should be high, between .99 and 1. I would take the max and abs functions out of your calculation, too.
EDIT:
I spoke too soon. I confused cross-correlation with correlation coefficient, which is completely different. My answer might not be worth much.
I would agree that the result would be subjective. Something that would involve the sum of the squares of the differences, element by element, would have some value. Two identical arrays would give a value of 0 in that form. You would have to decide what value then becomes "bad". Make up 2 different vectors that "aren't too bad" and find their cross-correlation coefficient to be used as a guide.
(parenthetically: if you were doing a correlation coefficient where 1 or -1 would be great and 0 would be awful, I've been told by bio-statisticians that a real-life value of 0.7 is extremely good. I understand that this is not exactly what you are doing but the comment on correlation coefficient came up earlier.)