Hopefully someone can help explain this to me as I am getting very confused trawling through google.
I would like to know the exact steps to perform the FFT and IFFT of a set of real data samples, all the examples I found are either start with complex values or don't explain the exact steps without bringing huge amounts of mathematics into the process.
I am after a simple bullet point list of what to do, is that too much to ask for.
Cheers
A Fast Fourier Transform(FFT) is an algorithm that efficiently computes the Discrete Fourier Transform.
There are several FFT algorithms so the exact steps to be followed will depend on the particular FFT algorithm you are looking to implement.
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This is essentially a question on fundamentals, and whether or not there is a more efficient way to achieve what I am looking for. I have built a working fluid dynamics calculator in Excel to find the flow rates required for a target pressure loss, the optimisation is handled using Solver but it's very clunky and not user friendly.
I'm trying to replicate the function in Octave since it's widely used here, but I am a complete beginner; I'm probably missing something obvious. I can easily enter all of the math for a single iteration via a series of functions, but my excel file required using the 'Solver' macro, and I'm unsure how to efficiently replicate this in Octave.
I am aware that linprog (in matlab) and glpk (octave) can be used to solve systems of linear equations.
I have a series of nested equations which are all dependant on a single matrix, Q (flow rates at various locations). Many other inputs are required, but they either remain constant throughout calculation (e.g. system geometry) or are dictated by Q (e.g. Reynolds number and loss coefficients). In trying to simplify my problem I have settled on two steps:
Write code to solve my problem, input: Q matrix, output: pressure loss matrix
Create a loop that iterates different Q matrices until some conditions for the pressure loss matrix are met.
I don't think it will be practical to get my expressions into the form of A*x = B (in order to use glpk) given the complexity. In excel, I can point solver at a Q value that drives a multitude of equations that impact pressure loss, and it will find the value I need to achieve a target. How can I most efficiently replicate this functionality in Octave?
First off all Solver is not a macro. Pretty far from.
So, you're going to replicate a comprehensive "What-If" Analysis Plug-in -- so complex in fact, that Microsoft chose to contract a 3rd Party company of experts to develop the tool and provide support for it (successfully based on the 1.2 Billion copies they've distributed).
And you're going to this an inferior coding language that you're a complete beginner with? Cool. I'd like to see this!
Cool. Here's a checklist of Solver's features, so you don't miss anything:
Good Luck!
More Information:
Wikipedia : Solver
Office.com : Define and Solve a Problem by using Solver
Frontline: Official Solver Page: http://solver.com
AppSource.Microsoft.com : Solver (with Video)
Frontline:L Solver International Manazine
My understanding is that filters in convolutional neural networks are going to extract features in raw data (or previous layers), so designing them by supervised learning through backpropagation makes complete sense. But I have seen some papers in which the filters are found by unsupervised clustering of input data samples. That looks strange to me how cluster centers can be regarded as good filters for feature extraction. Does anybody have a good explanation for that?
Certain popular clustering algorithms such as k-means are vector quantization methods.
They try to find a good least-squares quantization of the data, such that every data point can be represented by a similar vector with least-squares difference.
So from a least-squares approximation point of view, the cluster centers are good approximations (we can't afford to find the optimal centers, but we have a good chance at finding reasonably good centers). Whether or not least squares is appropriate depends a lot on the data, for example all attributes should be of the same kind. For a typical image processing task, where each pixel is represented the same way, this will be a good starting point for later supervised optimization. But I believe soft factorizations will usually be better that do not assume every patch is of exactly one kind.
So I have been making a simple HTML5 tuner using the Web Audio API. I have it all set up to respond to the correct frequencies, the problem seems to be with getting the actual frequencies. Using the input, I create an array of the spectrum where I look for the highest value and use that frequency as the one to feed into the tuner. The problem is that when creating an analyser in Web Audio it can not become more specific than an FFT value of 2048. When using this if i play a 440hz note, the closest note in the array is something like 430hz and the next value seems to be higher than 440. Therefor the tuner will think I am playing these notes when infact the loudest frequency should be 440hz and not 430hz. Since this frequency does not exist in the analyser array I am trying to figure out a way around this or if I am missing something very obvious.
I am very new at this so any help would be very appreciated.
Thanks
There are a number of approaches to implementing pitch detection. This paper provides a review of them. Their conclusion is that using FFTs may not be the best way to go - however, it's unclear quite what their FFT-based algorithm actually did.
If you're simply tuning guitar strings to fixed frequencies, much simpler approaches exist. Building a fully chromatic tuner that does not know a-priori the frequency to expect is hard.
The FFT approach you're using is entirely possible (I've built a robust musical instrument tuner using this approach that is being used white-label by a number of 3rd parties). However you need a significant amount of post-processing of the FFT data.
To start, you solve the resolution problem using the Short Timer FFT (STFT) - or more precisely - a succession of them. The process is described nicely in this article.
If you intend building a tuner for guitar and bass guitar (and let's face it, everyone who asks the question here is), you'll need t least a 4092-point DFT with overlapping windows in order not to violate the nyquist rate on the bottom E1 string at ~41Hz.
You have a bunch of other algorithmic and usability hurdles to overcome. Not least, perceived pitch and the spectral peak aren't always the same. Taking the spectral peak from the STFT doesn't work reliably (this is also why the basic auto-correlation approach is also broken).
I am trying to implement the cosine and sine functions in floating point (but I have no floating point hardware).
Since my processor has no floating-point hardware, nor instructions, I have already implemented algorithms for floating point multiplication, division, addition, subtraction, and square root. So those are the tools I have available to me to implement cosine and sine.
I was considering using the CORDIC method, at this site
However, I implemented division and square root using newton's method, so I was hoping to use the most efficient method.
Please don't tell me to just go look in a book or that "paper's exist", no kidding they exist. I am looking for names of well known algorithms that are known to be fast and efficient.
First off, depending on your accuracy requirements, this can be considerably fussier than your earlier questions.
Now that you've been warned: you'll first want to reduce the argument modulo pi/2 (or 2pi, or pi, or pi/4) to get the input into a manageable range. This is the subtle part. For a nice discussion of the issues involved, download a copy of K.C. Ng's ARGUMENT REDUCTION FOR HUGE ARGUMENTS: Good to the Last Bit. (simple google search on the title will get you a pdf). It's very readable, and does a great job of describing why this is tricky.
After doing that, you only need to approximate the functions on a small range around zero, which is easily done via a polynomial approximation. A taylor series will work, though it is inefficient. A truncated chebyshev series is easy to compute and reasonably efficient; computing the minimax approximation is better still. This is the easy part.
I have implemented sine and cosine exactly as described, entirely in integer, in the past (sorry, no public sources). Using hand-tuned assembly, results in the neighborhood of 100 cycles are entirely reasonable on "typical" processors. I don't know what hardware you're dealing with (the performance will mostly be gated on how quickly your hardware can produce the high part of an integer multiply).
For various levels of precision, you can find some good approximations here:
http://www.ganssle.com/approx.htm
With the added advantage that they are deterministic in runtime unlike the various "converging series" options which can vary wildly depending on the input value. This matters if you are doing anything real-time (games, motion control etc.)
Since you have the basic arithmetic operations implemented, you may as well implement sine and cosine using their taylor series expansions.
I don't have much experience in machine learning, pattern recognition, data mining, etc. and in their underlying theory and systems.
I would like to develop an artificial model of the time it takes a human to make a move in a given Sudoku puzzle.
So what I'm looking for as an output from the machine learning process is a model that can give predictions on how long does it take for a target human to make a move in a given Sudoku situation.
Same input doesn't always map to same outcome. It takes different times for the human to make a move with the same situation, but my hypothesis is that there's a tendency in the resulting probability distribution. (My educated guess is that it is ~normal.)
I have ideas about the factors that influence the distribution (like #empty slots) but would preferably leave it to the system to figure these patterns out. Please notice, that I'm not interested in the patterns, just the model.
I can generate sample and test data easily by running sudoku puzzles and measuring the times it takes to make the moves.
What kind of learning algorithm would you suggest to use for this?
I was thinking NNs, but I'm not sure if they can have the desired property of giving weighted random outcomes for the same input.
If I understand this correctly you have an input vector of length 81, which contains 1 if the square is filled in and 0 otherwise. You want to learn a function which returns a probability distribution which models the response time of a human to that board position.
My first response would be that this is a regression problem and you should try straightforward linear regression. This will not provide you with a distribution of response times, but a single 'best-guess' response time.
I'm not clear on why you want to model a distribution of response times. However, if you really want to do want to output a distribution then it sounds like you want to look at Bayesian methods. I'm not really an expert on Bayesian inference, so I can't help you much further here.
However, I don't really think your approach is going to work because I agree with your intuition about features such as the number of empty slots being important. There are also other obvious features, such as the number of empty slots per row/column that are likely to be important. Explicitly putting these features in your representation will probably be much more successful than expecting that the learning algorithm will infer something similar on its own.
The monte carlo method seems like it would work well here but would require a stack of solutions the size of the moon to really do it. And it wouldn't give you the time per person, just the time on average.
My understanding of it, tenuous as it is, is that you have a database with a board position and the time it took a human to make the next move. At the very least you have a starting point for most moves. Even if it's not in the database you could start to calculate how long it would take to make a move based on some algorithm. Though I know you had specified you wanted machine learning to do this it might be worth segmenting the problem into something a little smaller then building on it.
If you have some guesstimate as to what influences the function (# of empty cell, etc), try to train a classifier on a vector of features, and not on the 81 cells vector (0/1 or 0..9, doesn't really matter for my argument).
I think that your claim:
we wouldn't have to necessary know the underlying patterns, the "trained patterns" in a learning system automatically encodes these sometimes quite delicate and subtle patterns inside them -- that's one of their great power
is wrong. you do have to give the network the right domain. for example, when trying to detect object in an image, working in the pixel domain is pointless. you'll only get results if you first run some feature detection to detect edges, corners, etc.
Theoretically, with enough non-linearity (in NN - enough layers in the network) it can detect such things, but in practice, I have never seen that work, without giving the classifier the right features to work with.
I was thinking NNs, but I'm not sure if they can have the desired property of giving weighted random outcomes for the same input.
You're just trying to learn a function from 2^81 or 10^81 (or a much smaller feature space as I suggest) to R (response time between 0 and Inf) or some discretization of that. So NN and other classifiers can do that.