I have a linear regression equation from school , which gives a value between 1 and -1 indicative of whether or not a set of data points are close enough to a linear function
and the equation given here
http://people.hofstra.edu/stefan_waner/realworld/calctopic1/regression.html
under best fit of a line. I would like to use these to do simple gesture detection based on a point in 3-space (x,y,z) - forward, back, left, right, up, down. First I would see if they fall on a line in 2 of the 3 dimensions, then I would see if that line's slope approached zero or infinity.
Is this fast enough for functional gesture recognition? If not, could someone propose an alternative algorithm?
If I've understood your question correctly then (1) the calculation you describe here would probably be plenty fast enough, (2) it may not actually do what you want, and (3) the stuff that'll be slow in an actual implementation would lie elsewhere.
So, I think you're proposing to do this. (1) Identify the positions of ... something ... (the user's hand, perhaps) in three-dimensional space, at several successive times. (2) For (say) each of {x,y} and {x,z}, look at those two coordinates of each point, compute the correlation coefficient (which is what your formula describes) and see whether it's close to +-1. (3) If both correlation coefficients are close to +-1 then the points lie approximately on a straight line; calculate the gradient of that line (using a formula similar to that of the correlation coefficient). (4) If the gradients are both very close to 0 or +- infinity, then your line is approximately parallel to one axis, which is the case you're trying to recognize.
1: Is it fast enough? You might perhaps be sampling at 50 frames per second or thereabouts, and your gestures might take a second to execute. So you'll have somewhere on the order of 50 positions. So, the total number of arithmetic operations you'll need is maybe a few hundred (including a modest number of square roots). In the worst case, you might be doing this in emulated floating-point on a slow ARM processor or something; in that case, each arithmetic operation might take a couple of hundred cycles, so the whole thing might be 100k cycles, which for a really slow processor running at 100MHz would be about a millisecond. You're not going to have any problem with the time taken to do this calculation.
2: Is it the right thing? It's not clear that it's the right calculation. For instance, suppose your user's hand moves back and forth rapidly several times along the x-axis; that will give you a positive result; is that what you want? Suppose the user attempts the gesture you want but moves at slightly the wrong angle; you may get a negative result. Suppose they move exactly along the x-axis for a bit and then along the y-axis for a bit; then the projections onto the {x,y}, {x,z} and {y,z} planes will all pass your test. These all seem like results you might not want.
3: Is it where the real cost will lie? This all assumes you've already got (x,y,z) coordinates. Getting those is probably going to be more expensive than processing them. For instance, if you have a camera-based system of some kind then there'll be some nontrivial image processing for every frame. Or perhaps you're integrating up data from accelerometers (which, by the way, is likely to give nasty inaccurate position results); the chances are that you're doing some filtering and other calculations to get position data. I bet that the cost of performing a calculation like this one will be substantially less than the cost of getting the coordinates in the first place.
Related
Problem
I want to find
The first root
The first local minimum/maximum
of a black-box function in a given range.
The function has following properties:
It's continuous and differentiable.
It's combination of constant and periodic functions. All periods are known.
(It's better if it can be done with weaker assumptions)
What is the fastest way to get the root and the extremum?
Do I need more assumptions or bounds of the function?
What I've tried
I know I can use root-finding algorithm. What I don't know is how to find the first root efficiently.
It needs to be fast enough so that it can run within a few miliseconds with precision of 1.0 and range of 1.0e+8, which is the problem.
Since the range could be quite large and it should be precise enough, I can't brute-force it by checking all the possible subranges.
I considered bisection method, but it's too slow to find the first root if the function has only one big root in the range, as every subrange should be checked.
It's preferable if the solution is in java, but any similar language is fine.
Background
I want to calculate when arbitrary celestial object reaches certain height.
It's a configuration-defined virtual object, so I can't assume anything about the object.
It's not easy to get either analytical solution or simple approximation because various coordinates are involved.
I decided to find a numerical solution for this.
For a general black box function, this can't really be done. Any root finding algorithm on a black box function can't guarantee that it has found all the roots or any particular root, even if the function is continuous and differentiable.
The property of being periodic gives a bit more hope, but you can still have periodic functions with infinitely many roots in a bounded domain. Given that your function relates to celestial objects, this isn't likely to happen. Assuming your periodic functions are sinusoidal, I believe you can get away with checking subranges on the order of one-quarter of the shortest period (out of all the periodic components).
Maybe try Brent's Method on the shortest quarter period subranges?
Another approach would be to apply your root finding algorithm iteratively. If your range is (a, b), then apply your algorithm to that range to find a root at say c < b. Then apply your algorithm to the range (a, c) to find a root in that range. Continue until no more roots are found. The last root you found is a good candidate for your minimum root.
Black box function for any range? You cannot even be sure it has the continuous domain over that range. What kind of solutions are you looking for? Natural numbers, integers, real numbers, complex? These are all the question that greatly impact the answer.
So 1st thing should be determining what kind of number you accept as the result.
Second is having some kind of protection against limes of function that will try to explode your calculations as it goes for plus or minus infinity.
Since we are touching the limes topics you could have your solution edge towards zero and look like a solution but never touch 0 and become a solution. This depends on your margin of error, how close something has to be to be considered ok, it's good enough.
I think for this your SIMPLEST TO IMPLEMENT bet for real number solutions (I assume those) is to take an interval and this divide and conquer algorithm:
Take lower and upper border and middle value (or approx middle value for infinity decimals border/borders)
Try to calculate solution with all 3 and have some kind of protection against infinities
remember all 3 values in an array with results from them (3 pair of values)
remember the current best value (one its closest to solution) in seperate variable (a pair of value and result for that value)
STEP FORWARD - repeat above with 1st -2nd value range and 2nd -3rd value range
have a new pair of value and result to be closest to solution.
clear the old value-result pairs, replace them with new ones gotten from this iteration while remembering the best value solution pair (total)
Repeat above for how precise you wish to get and look at that memory explode with each iteration, keep in mind you are gonna to have exponential growth of values there. It can be further improved if you lets say take one interval and go as deep as you wanna, remember best value-result pair and then delete all other memory and go for next interval and dig deep.
I've been doing BigO recently, and I get the formula ok, but I've written a piece of code that takes and input and returns a time taken to complete a sort. So I have the input and time, how do I use this to classify what sort of BigO it is? I've made graphs and can see which sort they are but I can't do it using the formula? I'm not strong on maths which I think is my problem here!
For instance I get:
Size Time Operations
200 2 163648
400 1 162240
800 15 2489456
1600 6 10247376
3200 19 40858160
6400 79 165383984
12800 318 656588080
25600 1274 2624318128
51200 5059 10476803408
102400 20333 41969291968
I know that this is O(n^2) by looking at the graph and comparing, but how do I prove it?
Yes, you can sample a thousand different input sizes, and then try to derive a Big-O value from that, but you shouldn't - not only because it doesn't actually prove anything, but because that isn't the point.
The way to prove O(n^2) is to prove it on the code itself, not through experiments. The actual running time isn't important, because Big-O notation doesn't say anything about that - in simple terms, it only specifies the dominant term of whatever formula you would use to calculate the exact running time, in the sense of the number of operations executed for that function. Constants are thrown away, and so are smaller terms - the actual running time of a function might be 1000n^2+1000000n, but that's still O(n^2).
You can't mathematically prove anything from this table; the complexity might be O(1) if Time remains at 20333 for all larger values.
The best you can do is try fitting several curves to this table and selecting the best fit according to Occam's razor.
You can't prove it by looking at the timings, you can only prove it by analysing the code to see how many steps are performed. The reason for this is that the time taken is a function not only of your program but many other things outside of your control as well.
For example, who can say whether your machine didn't spend an inordinate amount of time in other processes during one particular test run of your program? This sort of thing can be minimsed to a point using statistical methods but the proof requires solid data.
What you can do is to look at some of your data points to get support for the contention that it's O(n2). Have a look at the last four entries:
Input Time
128 318
256 1274 1274 / 318 = 4.006
512 5059 5059 / 1274 = 3.971
1024 20333 20333 / 5059 = 4.019
You can see that each doubling of the input size has a multiplier effect of the time of about 4 which would tend to indicate an O(n2) property.
But this is support only. It applies only to that particular range of input values and, as stated, is subject to factors outside your control. Note also that the support would be harder to see if the time taken was not a simple one. For example, if the time function was t = n2/10 + 123n + 123456789, it would be a little harder to figure out.
Just by making a comparison between the values may not make any sense.However,if you plot a graph using this values( x-axis : input , y-axis:time),you will get a curve or a linear shape or whatever.Using this information,you can predict the BigO value of that function.Of course there may be(not always) some interrupts that affects the running of that process,but that does not last during the whole period.It is slight overhead that cannot affect the result.
In order to predict the BigO value , you will need some Calculus knowledge in order to make the analogy between the shape and BigO result.
For example,let's say that you got a linear shape and you know that it means O(n).In that point,you reached that result because you know the shape of a linear function graph and your graph looks like it.In order to reach the true proof , you have to draw both your functions curve and the graph of the mathematical function that has the closest shape to your graph.
There are some other functions like Big-Theta , Small-Omega that binds your function from upper or from lower.The mathematical function could be both of them,but as a result,your Big-O function is the closest one to that shape.
Basically I have a large (could get as large as 100,000-150,000 values) data set of 4-byte inputs and their corresponding 4-byte outputs. The inputs aren't guaranteed to be unique (which isn't really a problem because I figure I can generate pseudo-random numbers to add or xor the inputs with so that they do become unique), but the outputs aren't guaranteed to be unique either (so two different sets of inputs might have the same output).
I'm trying to create a function that effectively models the values in my data-set. I don't need it to interpolate efficiently, or even at all (by this I mean that I'm never going to feed it an input that isn't contained in this static data-set). However it does need to be as efficient as possible. I've looked into interpolation and found that it doesn't really fit what I'm looking for. For example, the large number of values means that spline interpolation won't do since it creates a polynomial per interval.
Also, from my understanding polynomial interpolation would be way too computationally expensive (n values means that the polynomial could include terms as high as pow(x,n-1). For x= a 4-byte number and n=100,000 it's just not feasible). I've tried looking online for a while now, but I'm not very strong with math and must not know the right terms to search with because I haven't come across anything similar so far.
I can see that this is not completely (to put it mildly) a programming question and I apologize in advance. I'm not looking for the exact solution or even a complete answer. I just need pointers on the topics that I would need to read up on so I can solve this problem on my own. Thanks!
TL;DR - I need a variant of interpolation that only needs to fit the initially given data-points, but which is computationally efficient.
Edit:
Some clarification - I do need the output to be exact and not an approximation. This is sort of an optimization of some research work I'm currently doing and I need to have this look-up implemented without the actual bytes of the outputs being present in my program. I can't really say a whole lot about it at the moment, but I will say that for the purposes of my work, encryption (or compression or any other other form of obfuscation) is not an option to hide the table. I need a mathematical function that can recreate the output so long as it has access to the input. I hope that clears things up a bit.
Here is one idea. Make your function be the sum (mod 232) of a linear function over all 4-byte integers, a piecewise linear function whose pieces depend on the value of the first bit, another piecewise linear function whose pieces depend on the value of the first two bits, and so on.
The actual output values appear nowhere, you have to add together linear terms to get them. There is also no direct record of which input values you have. (Someone could conclude something about those input values, but not their actual values.)
The various coefficients you need can be stored in a hash. Any lookups you do which are not found in the hash are assumed to be 0.
If you add a certain amount of random "noise" to your dataset before starting to encode it fairly efficiently, it would be hard to tell what your input values are, and very hard to tell what the outputs are even approximately without knowing the inputs.
Since you didn't impose any restriction on the function (continuous, smooth, etc), you could simply do a piece-wise constant interpolation:
or a linear interpolation:
I assume you can figure out how to construct such a function without too much trouble.
EDIT: In light of your additional requirement that such a function should "hide" the data points...
For a piece-wise constant interpolation, the constant intervals should be randomized so as to not reveal where the data point is. So for example in the picture, the intervals are centered about the data point it's interpolating. Instead, you might want to do something like:
[0 , 0.3) -> 0
[0.3 , 1.9) -> 0.8
[1.9 , 2.1) -> 0.9
[2.1 , 3.5) -> 0.2
etc
Of course, this only hides the x-coordinate. To hide the y-coordinate as well, you can use a linear interpolation.
Simply make it so that the "pointy" part isn't where the data point is. Pick random x-values such that every adjacent data point has one of these x-values in between. Then interpolate such that the "pointy" part is at these x-values.
I suggest a huge Lookup Table full of unused entries. It's the brute-force approach, having an ordered table of outputs, ordered by every possible value of the input (not just the data set, but also all other possible 4-byte value).
Though all of your data would be there, you could fill the non-used inputs with random, arbitrary, or stochastic (random whithin potentially complex constraints) data. If you make it convincing, no one could pick your real data out of it. If a "real" function interpolated all your data, it would also "contain" all the information of your real data, and anyone with access to it could use it to generate an LUT as described above.
LUTs are lightning-fast, but very memory hungry. Your case is on the edge of feasibility, requiring (2^32)*32= 16 Gigabytes of RAM, which requires a 64-bit machine to run. That is just for the data, not the program, the Operating System, or other data. It's better to have 24, just to be sure. If you can afford it, they are the way to go.
Okay, so this is a straight math question and I read up on meta that those need to be written to sound like programming questions. I'll do my best...
So I have graph made in flot that shows the network usage (in bytes/sec) for the user. The data is 4 minutes apart when there is activity, and otherwise set at the start of the usage range (let's say day 1) and the end of the range (day 7). The data is coming from a CGI script I have no control over, so I'm fairly limited in what I can provide the user.
I never took trig or calculus, so I'm pretty much in over my head. What I want is for the user to have the option to click any point on the graph and see their bandwidth usage for that moment. Since the lines between real data points are drawn straight, this can be done by getting the points before and after where the user has clicked and finding the y-interval.
It took me weeks to finally get a helpful math person to explain this to me. Everyone else has insisted on trying to teach me Riemann sum techniques and all sorts of other heavy stuff that not only is confusing to me, doesn't seem necessary for the problem.
But I also want the user to be able to highlight the graph from two arbitrary points on the y-axis (time) to get the amount of network usage total during that range. I know this would be inaccurate, but I need it to be the right inaccurate using a solid equation.
I thought this was the area under the line, but experiments with much simpler graphs makes this seem just far too high. I figured out I could take the distance from y2 - y1 and multiply it by x2 - x1 and then divide by two to get the area of the graph below the line like a triangle, but again, the numbers seemed to high. (maybe they are just big numbers and I don't get this math stuff at all).
So what I need, if anyone would be really awesome enough to provide it before this question is closed down for being too pure-math, is either the name of the concept I should be researching or the equation itself. Or the bad news that I do need advanced math to get an accurate result.
I am not bad at math, just as a last note, I just am not familiar with math beyond 10th grade and so I need some place to start. All the math sites seem to keep it too simple or way over my paygrade.
If I understood correctly what you're asking (and that is somewhat doubtful), you should find what you seek in these links:
Linear interpolation
(calculating the value of the point in between)
Trapezoidal rule
(calculating the area below the "curve")
*****Edit, so we can get this over :) without much ado:*****
So I have graph made in flot that shows the network usage (in bytes/sec) for the user. The data is 4 minutes apart when there is activity, and otherwise set at the start of the usage range (let's say day 1) and the end of the range (day 7). The data is coming from a CGI script I have no control over, so I'm fairly limited in what I can provide the user.
What is a "flot" ?
Okey, so you have speed on y axis [in bytes/sec]; and time on x axis in [sec], right?
That means, that if you're flotting (I'm bored, yes :) speed over time, in linear segments, interpolating at some particular point in time you'll get speed at that particular point in time.
If you wish to calculate how much bandwidth you've spend, you need to determine the area beneath that curve. The area from point "a" to point "b" will determine the spended bandwidth in [bytes] in that time period.
It took me weeks to finally get a helpful math person to explain this to me. Everyone else has insisted on trying to teach me Riemann sum techniques and all sorts of other heavy stuff that not only is confusing to me, doesn't seem necessary for the problem.
In the immortal words of Snoopy: "Good grief !"
But I also want the user to be able to highlight the graph from two arbitrary points on the y-axis (time) to get the amount of network usage total during that range. I know this would be inaccurate, but I need it to be the right inaccurate using a solid equation.
It would not be inaccurate.
It would be actually perfectly accurate (well, apart from roundoff error in bytes :), since you're using linear interpolation on linear segments.
I thought this was the area under the line, but experiments with much simpler graphs makes this seem just far too high. I figured out I could take the distance from y2 - y1 and multiply it by x2 - x1 and then divide by two to get the area of the graph below the line like a triangle, but again, the numbers seemed to high. (maybe they are just big numbers and I don't get this math stuff at all).
"like a triangle" --> should be "like a trapezoid"
If you do deltax*(y2-y1)/2 you will get the area, yes (this works only for linear segments). This is the basis principle of trapezoidal rule.
If you're uncertain about what you're calculating use dimensional analysis: speed is in bytes/sec, time is in sec, bandwidth is in bytes. Multiplying speed*time=bandwidth, and so on.
What I want is for the user to have
the option to click any point on the
graph and see their bandwidth usage
for that moment. Since the lines
between real data points are drawn
straight, this can be done by getting
the points before and after where the
user has clicked and finding the
y-interval.
Yes, that's a good way to find that instantaneous value. When you report that value back, it's in the same units as the y-axis, so that means bytes/sec, right?
I don't know how rapidly the rate changes between points, but it's even simpler if you simply pick the closest point and report its value. You simplify your problem without sacrificing too much accuracy.
I thought this was the area under the
line, but experiments with much
simpler graphs makes this seem just
far too high. I figured out I could
take the distance from y2 - y1 and
multiply it by x2 - x1 and then divide
by two to get the area of the graph
below the line like a triangle, but
again, the numbers seemed to high.
(maybe they are just big numbers and I
don't get this math stuff at all).
To calculate the total bytes over a given time interval, you should find the index closest to the starting and ending point and multiply the value of y by the spacing of your x-points and add them all together. That will give you the total # of bytes consumed during that time interval, but there's one more wrinkle you might have forgotten.
You said that the points come in "4 minutes apart", and your y-axis is in bytes/second. Remember that units matter. Your area is the sum of bytes/second times a spacing in minutes. To make the units come out right you have to multiply by 60 seconds/minute to get the final value of bytes that you want.
If that "too high" value is still off, consider units again. It's 1024 bytes per kbyte, and 1024*1024 bytes per MB. Check the units of the values you're checking the calculation against.
UPDATE:
No wonder you're having problems. Your original question CLEARLY stated bytes/sec. Even this question is imprecise and confusing. How did you arrive at "amount of data" at a given time stamp? Are those the total bits transferred since the last time stamp? If yes, simply add the values between the start and end of the interval you want and convert to the units convenient for you.
The network usage total is not in bytes (kilo-, mega-, whatever) per second. It would be in just straight bytes (or kilo-, or whatever).
For example, 2 megabytes per second over an interval of 10 seconds would be 20 megabytes total. It would not be 20 megabytes per second.
Or do you perhaps want average bytes per second over an interval?
This would be a lot easier for you if you would accept that there is well-established terminology for the concepts that you are having trouble expressing concisely or accurately, and that these mathematical terms have been around far longer than you. Since you've clearly gone through most of the trouble of understanding the concepts, you might as well break down and start calling them by their proper names.
That said:
There are 2 obvious ways to graph bandwidth, and two ways you might be getting the bandwidth data from the server. First, there's the cumulative usage function, which for any time is simply the total amount of data transferred since the start of the measurement. If you plot this function, you get a graph that never decreases (since you can't un-download something). The units of the values of this function will be bytes or kB or something like that.
What users are typically interested is in the instantaneous usage function, which is an indicator of how much bandwidth you are using right now. This is what users typically want to see. In mathematical terms, this is the derivative of the cumulative function. This derivative can take on any value from 0 (you aren't downloading) to the rated speed of your network link (indicating that you're pushing as much data as possible through your connection). The units of this function are bytes per second, or something related like Mbps (megabits per second).
You can approximate the instantaneous bandwidth with the average data usage over the past few seconds. This is computed as
(number of bytes transferred)
-----------------------------------------------------------------
(number of seconds that elapsed while transferring those bytes)
Generally speaking, the smaller the time interval, the more accurate the approximation. For simplicity's sake, you usually want to compute this as "number of bytes transferred since last report" divided by "number of seconds since last report".
As an example, if the server is giving you a report every 4 minutes of "total number of bytes transferred today", then it is giving you the cumulative function and you need to approximate the derivative. The instantaneous bandwidth usage rate you can report to users is:
(total transferred as of now) - (total as of 4 minutes ago) bytes
-----------------------------------------------------------
4*60 seconds
If the server is giving you reports of the form "number of bytes transferred since last report", then you can directly report this to users and plot that data relative to time. On the other hand, if the user (or you) is concerned about a quota on total bytes transferred per day, then you will need to transform the (approximately) instantaneous data you have into the cumulative data. This process, known as computing the integral, is the opposite of computing the derivative, and is in some ways conceptually simpler. If you've kept track of each of the reports from the server and the timestamp, then for each time, the value you plot is the total of all the reports that came in before that time. If you're doing this in realtime, then every time you get a new report, the graph jumps up by the amount in that report.
I am not bad at math, ... I just am not familiar with math beyond 10th grade
This is like saying "I'm not bad at programming, I have no trouble with ifs and loops but I never got around to writing more than one function."
I would suggest you enrol in a maths class of some kind. An understanding of matrices and the basics of calculus gives you an appreciation of many things, and can be useful in all sorts of areas. You'll be able to understand more of Wikipedia articles and SO answers - and questions!
If you can't afford that, try to find some lecture videos or something.
Everyone else has insisted on trying to teach me Riemann sum techniques
I can't see why. You don't need them for this - though if you had learned them, I expect you would find it easier to come up with a solution. You see, Riemann sums attempt to give you a "familiar" notion of area. The sort of area you (hopefully) learned years ago.
Getting the area below your usage graph between two points will tell you (approximately) how much was used over that period.
How do you find the area of a floor plan? You break it up into rectangles and triangles, find the area of each, and add them together. You can do the same thing with your graph, basically. Someone has worked out a simple way of doing this called the trapezoidal rule. It's just a matter of choosing how to divide your graph into strips, and in your case this is easy: just use the data points themselves as dividers. (You'll also need to work out the value of the graph at the left and right ends of the region selected by the user, using linear interpolation.)
If there's anything I've said that isn't clear to you (as there may well be), please leave a comment.
The typical FFT for audio looks pretty similar to this, with most of the action happening on the far left side
http://www.flight404.com/blog/images/fft.jpg
He multiplied it by a partial sine wave to get it to the bottom, but the article isn't too specific on this part of it. It also seems like a "good enough" modification of the dataset, rather than one based on some property. I understand that human hearing is better suited to the higher frequencies, thus, most music will have amplified bass and attenuated treble so that both sound to us as being of relatively equal strength.
My question is what modification needs to be done to the FFT to compensate for this standard falloff?
for(i = 0; i < fft.length; i++){
fft[i] = fft[i] * Math.log(i + 1); // does, eh, ok but the high
// end is still not really "loud"
// enough
}
EDIT ::
http://en.wikipedia.org/wiki/Equal-loudness_contour
I came across this article, I think it might be the direction to head in, but there still might be some property of an FFT that needs to be counteracte.
First, are you sure you want to do this? It makes sense to compensate for some things, like the microphone response not being flat, but not human perception. People are used to hearing sounds with the spectral content that the sounds have in the real world, not along perceptual equal loudness curves. If you play a sound that you've modified in the way you suggest it would sound strange. Maybe some people like the music to have enhanced low frequencies, but this is a matter of taste, not psychophysics.
Or maybe you are compensating for some other reason, for example, taking into account the poorer sensitivity to lower frequencies might enhance a compression algorithm. Is this the idea?
If you do want to normalize by the equal loudness curves, one should note that most of the curves and equations are in terms of sound pressure level (SPL). SPL is the log of the square of the waveform amplitude, so when you work with the FFTs, it's probably easiest to work with their square (the power specta). (Or, of course, you could compensate in other ways by, say, multiplying by sqrt(log(i+1)) in your equation above -- assuming that the log was an approximation of the inverse equal-loudness curve.)
I think the equal loudness contour is exactly the right direction.
However, its shape depends on the absolute pressure level.
In other words the sensitivity curve of our hearing changes with sound pressure.
There is no "correct normalization" if you have no information about absolute levels.
If this is a problem depends on what you want to do with the data.
The loudness contour is standardized in ISO 226 but this document is not freely available for download. It should be in a decent university library though.
Here is another source for
loudness contours
So you are trying to raise the level of the high end frequencies? Sounds like a high pass filter with a minimum multiplier might work, so that you don't attenuate the low frequency signals too much. Pick up a good book on filter design, maybe monkey around with this applet
In the old days of first samplers, this is before MOTU Boost people :) it wasn't FFT but simple (Fairlight or Roland it first I think) Normalisation done on the original or resulting time-domain signal (if you are doing beat slicing, recycle-style); can't you do that? Or only go for the FFT after you compensate to counteract for it?
Seems like a two phase procedure otherwise, I'd personally leave FFT as is for the task..