I'm having sample data of tri-axial accelerometer as listed below:
Timestamp, AcceX, AcceY, AcceZ
0.0, -0.96, -0.69, -1.24
0.1, ............
I want to determine velocity and distance traveled by the object with accelerometer.
Determining the velocity and position from an accelerometer is much harder than it might at first seem.
Depending on your specific needs, there might be a way to form an estimate that would work, but it is generally a very inaccurate approach, and almost worthless given what's usually available on a phone. With specialized equipment one can do much better.
First problem: gravity will be recorded as an acceleration and it's hard to remove (so you at least need a gyro too).
Second problem: to get distance and velocity you need to integrate your acceleration so small errors will accumulate.
To find out more, search for "dead reckoning". See, for example, the first google hit i got, which seems to explain the issues well enough.
Related
First of all a disclaimer: I'm posting this question here even if I realize it is quite maths heavy, because I have trouble figuring out on what other site it could belong.
I'm writing a 2d spaceships game where the player will have to select the ship's destination and a course will be automatically plotted.
Along with this, I'm offering various options to control the ship's acceleration while it gets there. All these options have to do with the target velocity at the destination.
One option is to select the desired destination and velocity vector, in which case the program will use cubic interpolation, since starting and target coordinates and velocity are available.
Another option is to just select the destination point, but let the game calculate the final velocity vector. This is done through quadratic interpolation (ie. acceleration is constant).
I would like to introduce another option: let the player select the destination and the maximum absolute value of the velocity vector, as in
sqrt( vf_x^2 + vf_y^2 ) <= Vf_max
I take I shall use a 3rd order polynomial to model the course in this case, but I'm having a pretty hard time figuring out the coefficients, since I miss one equation for each coordinate (the one given by velocity at destination). Furthermore, I'm confused as to how I should use the Vf_max constraint to help me figure out the missing coefficients.
I suspect it could be an optimization problem, but I'm quite ignorant about this topic.
Can anybody help me find a solution or point me in the right direction, please?
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.
I'm designing a real time Audio Analyser to be embedded on a FPGA chip. The finished system will read in a live audio stream and output frequency and amplitude pairs for the X most prevalent frequencies.
I've managed to implement the FFT so far, but it's current output is just the real and imaginary parts for each window, and what I want to know is, how do I convert this into the frequency and amplitude pairs?
I've been doing some reading on the FFT, and I see how they can be turned into a magnitude and phase relationship but I need a format that someone without a knowledge of complex mathematics could read!
Thanks
Thanks for these quick responses!
The output from the FFT I'm getting at the moment is a continuous stream of real and imaginary pairs. I'm not sure whether to break these up into packets of the same size as my input packets (64 values), and treat them as an array, or deal with them individually.
The sample rate, I have no problem with. As I configured the FFT myself, I know that it's running off the global clock of 50MHz. As for the Array Index (if the output is an array of course...), I have no idea.
If we say that the output is a series of One-Dimensional arrays of 64 complex values:
1) How do I find the array index [i]?
2) Will each array return a single frequency part, or a number of them?
Thankyou so much for all your help! I'd be lost without it.
Well, the bad news is, there's no way around needing to understand complex numbers. The good news is, just because they're called complex numbers doesn't mean they're, y'know, complicated. So first, check out the wikipedia page, and for an audio application I'd say, read down to about section 3.2, maybe skipping the section on square roots: http://en.wikipedia.org/wiki/Complex_number
What that's telling you is that if you have a complex number, a + bi, you can picture it as living in the x,y plane at location (a,b). To get the magnitude and phase, all you have to do is find two quantities:
The distance from the origin of the plane, which is the magnitude, and
The angle from the x-axis, which is the phase.
The magnitude is simple enough: sqrt(a^2 + b^2).
The phase is equally simple: atan2(b,a).
The FFT result will give you an array of complex values. The twice the magnitude (square root of sum of the complex components squared) of each array element is an amplitude. Or do a log magnitude if you want a dB scale. The array index will give you the center of the frequency bin with that amplitude. You need to know the sample rate and length to get the frequency of each array element or bin.
f[i] = i * sampleRate / fftLength
for the first half of the array (the other half is just duplicate information in the form of complex conjugates for real audio input).
The frequency of each FFT result bin may be different from any actual spectral frequencies present in the audio signal, due to windowing or so-called spectral leakage. Look up frequency estimation methods for the details.
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..