to get a moving average like filter on ADC data in microcontrollers, i like to use the following code:
average = average + ((new_value - average)>>2);
it works nicely and super fast and I can adjust the filterstrength just by changing the amount of right-shift.
My question is now: has this filter a name?
cheers :)
Somewhere between Cumulative Moving Average and Weighted Moving Average
In general terms, it can be a specific case of Savitzky-Golay method as well. It is a low pass filter with the following explantion:
A Savitzky–Golay filter is a digital filter that can be applied to a set of digital data points for the purpose of smoothing the data, that is, to increase the precision of the data without distorting the signal tendency. This is achieved, in a process known as convolution, by fitting successive sub-sets of adjacent data points with a low-degree polynomial by the method of linear least squares. When the data points are equally spaced, an analytical solution to the least-squares equations can be found, in the form of a single set of "convolution coefficients" that can be applied to all data sub-sets, to give estimates of the smoothed signal, (or derivatives of the smoothed signal) at the central point of each sub-set.
Related
I'm required to calculate median of many parameters received from a kafka stream for 15 min time window.
i couldn't find any built in function for that, but I have found a way using custom WindowFunction.
my questions are:
is it a difficult task for flink? the data can be very large.
if the data gets to giga bytes, will flink store everything in memory until the end of the time window? (one of the arguments of apply WindowFunction implementation is Iterable - a collection of all data which came during the time window )
thanks
Your question contains several aspects, but let me answer the most fundamental one:
Is this a hard task for Flink, why is this not a standard example?
Yes, the median is a hard concept, as the only way to determine it is to keep the full data.
Many statistics don't need the full data to be calculated. For instance:
If you have the total sum, you can take the previous total sum and add the latest observation.
If you have the total count, you add 1 and have the new total count
If you have the average, under the hood you can just keep track of the total sum and count, and at any point calculate the new average based on an observation.
This can even be done with more complicated metrics, like the standard deviation.
However, there is no shortcut for determining the median, the only way to know what the median is after adding a new observation, is by looking at all observations and then figuring out what the middle one is.
As such, it is a challenging metric and the size of the data that comes in will need to be handled. As mentioned there may be estimates in the workings like this: https://issues.apache.org/jira/browse/FLINK-2147
Alternately, you could look at how your data is distributed, and perhaps estimate the median with metrics like Mean, Skew, and Kurtosis.
A final solution I could come up with, is if you need to know approximately what the value should be, is to pick a few 'candidates' and count the fractin of observations below them. The one closest to 50% would then be a reasonable estimate.
I would like to know how noise can be removed from data (say, radio data that is an array of rows and columns with each data point representing intensity of the radiation in the given frequency and time).The array can contain radio bursts. But many fixed frequency radio noise also exists(RFI=radio frequency intereference).How to remove such noise and bring out only the burst.
I don't mean to be rude, but this question isn't clear at all. Please sharpen it up.
The normal way to remove noise is first to define it exactly and then filter it out. Usually this is done in the frequency domain. For example, if you know the normalized power spectrum P(f) of the noise, build a filter with response
e/(e + P(f))
where e<1 is an attenuation factor.
You can implement the filter digitally using FFT or a convolution kernel.
When you don't know the spectrum of the noise or when it's white, then just use the inverse of the signal band.
What is a simple way to see if my low-pass filter is working? I'm in the process of designing a low-pass filter and would like to run tests on it in a relatively straight forward manner.
Presently I open up a WAV file and stick all the samples in a array of ints. I then run the array through the low-pass filter to create a new array. What would an easy way to check if the low-pass filter worked?
All of this is done in C.
You can use a broadband signal such as white noise to measure the frequency response:
generate white noise input signal
pass white noise signal through filter
take FFT of output from filter
compute log magnitude of FFT
plot log magnitude
Rather than coding this all up you can just dump the output from the filter to a text file and then do the analysis in e.g. MATLAB or Octave (hint: use periodogram).
Depends on what you want to test. I'm not a DSP expert, but I know there are different things one could measure about your filter (if that's what you mean by testing).
If the filter is linear then all information of the filter can be found in the impulse response. Read about it here: http://en.wikipedia.org/wiki/Linear_filter
E.g. if you take the Fourier transform of the impulse response, you'll get the frequency response. The frequency response easily tells you if the low-pass filter is worth it's name.
Maybe I underestimate your knowledge about DSP, but I recommend you to read the book on this website: http://www.dspguide.com. It's a very accessible book without difficult math. It's available as a real book, but you can also read it online for free.
EDIT: After reading it I'm convinced that every programmer that ever touches an ADC should definitely have read this book first. I discovered that I did a lot of things the difficult way in past projects that I could have done a thousand times better when I had a little bit more knowledge about DSP. Most of the times an unexperienced programmer is doing DSP without knowing it.
Create two monotone signals, one of a low frequency and one of a high frequency. Then run your filter on the two. If it works, then the low frequency signal should be unmodified whereas the high frequency signal will be filtered out.
Like Bart above mentioned.
If it's LTI system, I would insert impulse and record the samples and perform FFT using matlab and plot magnitude.
You ask why?
In time domain, you have to convolute the input x(t) with the impulse response d(t) to get the transfer function which is tedious.
y(t) = x(t) * d(t)
In frequency domain, convolution becomes simple multiplication.
y(s) = x(s) x d(s)
So, transfer function is y(s)/x(s) = d(s).
That's the reason you take FFT of impulse response to see the behavior of the filter.
You should be able to programmatically generate tones (sine waves) of various frequencies, stuff them into the input array, and then compare the signal energy by summing the squared values of the arrays (and dividing by the length, though that's mathematically not necessary here because the signals should be the same length). The ratio of the output energy to the input energy gives you the filter gain. If your LPF is working correctly, the gain should be close to 1 for low frequencies, close to 0.5 at the bandwidth frequency, and close to zero for high frequencies.
A note: There are various (but essentially the same in spirit) definitions of "bandwidth" and "gain". The method I've suggested should be relatively insensitive to the transient response of the filter because it's essentially averaging the intensity of the signal, though you could improve it by ignoring the first T samples of the input, where T is related to the filter bandwidth. Either way, make sure that the signals are long compared to the inverse of the filter bandwidth.
When I check a digital filter, I compute the magnitude response graph for the filter and plot it. I then generate a linear sweeping sine wave in code or using Audacity, and pass the sweeping sine wave through the filter (taking into account that things might get louder, so the sine wave is quiet enough not to clip) . A visual check is usually enough to assert that the filter is doing what I think it should. If you don't know how to compute the magnitude response I suspect there are tools out there that will compute it for you.
Depending on how certain you want to be, you don't even have to do that. You can just process the linear sweep and see that it attenuated the higher frequencies.
Given is an array of 320 elements (int16), which represent an audio signal (16-bit LPCM) of 20 ms duration. I am looking for a most simple and very fast method which should decide whether this array contains active audio (like speech or music), but not noise or silence. I don't need a very high quality of the decision, but it must be very fast.
It occurred to me first to add all squares or absolute values of the elements and compare their sum with a threshold, but such a method is very slow on my system, even if it is O(n).
You're not going to get much faster than a sum-of-squares approach.
One optimization that you may not be doing so far is to use a running total. That is, in each time step, instead of summing the squares of the last n samples, keep a running total and update that with the square of the most recent sample. To avoid your running total from growing and growing over time, add an exponential decay. In pseudocode:
decay_constant=0.999; // Some suitable value smaller than 1
total=0;
for t=1,...
// Exponential decay
total=total*decay_constant;
// Add in latest sample
total+=current_sample;
if total>threshold
// do something
end
end
Of course, you'll have to tune the decay constant and threshold to suit your application. If this isn't fast enough to run in real time, you have a seriously underpowered DSP...
You might try calculating two simple "statistics" - first would be spread (max-min). Silence will have very low spread. Second would be variety - divide the range of possible values into say 16 brackets (= value range) and as you go through the elements, determine in which bracket that element goes. Noise will have similar numbers for all brackets, whereas music or speech should prefer some of them while neglecting others.
This should be possible to do in just one pass through the array and you do not need complicated arithmetics, just some addition and comparison of values.
Also consider some approximation, for example take only each fourth value, thus reducing the number of checked elements to 80. For audio signal, this should be okay.
I did something like this a while back. After some experimentation I arrived at a solution that worked sufficiently well in my case.
I used the rate of change in the cube of the running average over about 120ms. When there is silence (only noise that is) the expression should be hovering around zero. As soon as the rate starts increasing over a couple of runs, you probably have some action going on.
rate = cur_avg^3 - prev_avg^3
I used a cube because the square just wasn't agressive enough. If the cube is to slow for you, try using the square and a bitshift instead. Hope this helps.
It is clearly that the complexity should be at least O(n). Probably some simple algorithms that calculate some value range are good for the moment but I would look for Voice Activity Detection on web and for related code samples.
I have a number of tracks recorded by a GPS, which more formally can be described as a number of line strings.
Now, some of the recorded tracks might be recordings of the same route, but because of inaccurasies in the GPS system, the fact that the recordings were made on separate occasions and that they might have been recorded travelling at different speeds, they won't match up perfectly, but still look close enough when viewed on a map by a human to determine that it's actually the same route that has been recorded.
I want to find an algorithm that calculates the similarity between two line strings. I have come up with some home grown methods to do this, but would like to know if this is a problem that's already has good algorithms to solve it.
How would you calculate the similarity, given that similar means represents the same path on a map?
Edit: For those unsure of what I'm talking about, please look at this link for a definition of what a line string is: http://msdn.microsoft.com/en-us/library/bb895372.aspx - I'm not asking about character strings.
Compute the Fréchet distance on each pair of tracks. The distance can be used to gauge the similarity of your tracks.
Math alert: Fréchet was a pioneer in the field of metric space which is relevant to your problem.
I would add a buffer around the first line based on the estimated probable error, and then determine if the second line fits entirely within the buffer.
To determine "same route," create the minimal set of normalized path vectors, calculate the total power differences and compare the total to a quality measure.
Normalize the GPS waypoints on total path length,
walk the vectors of the paths together, creating a new set of path vectors for each path based upon the shortest vector at each waypoint,
calculate the total power differences between endpoints of each vector in the normalized paths weighting for vector length, and
compare against a quality measure.
Tune the power of the differences (start with, say, squared differences) and the quality measure (say as a percent of the total power differences) visually. This algorithm produces a continuous quality measure of the path match as well as a binary result (Are the paths the same?)
Paul Tomblin said: I would add a buffer
around the first line based on the
estimated probable error, and then
determine if the second line fits
entirely within the buffer.
You could modify the algorithm as the normalized vector endpoints are compared. You could determine if any endpoint difference was above a certain size (implementing Paul's buffer idea) or perhaps, if the endpoints were outside the "buffer," use that fact to ignore that endpoint difference, allowing a comparison ignoring side trips.
You could walk along each point (Pa) of LineString A and measure the distance from Pa to the nearest line-segment of LineString B, averaging each of these distances.
This is not a quick or perfect method, but should be able to give use a useful number and is pretty quick to implement.
Do the line strings start and finish at similar points, or are they of very different extents?
If you consider a single line string to be a sequence of [x,y] points (or [x,y,z] points), then you could compute the similarity between each pair of line strings using the Needleman-Wunsch algorithm. As described in the referenced Wikipedia article, the Needleman-Wunsch algorithm requires a "similarity matrix" which defines the distance between a pair of points. However, it would be easy to use a function instead of a matrix. In your case you could simply use the 2D Euclidean distance function (or a 3D Euclidean function if your points have elevation) to provide the distance between each pair of points.
I actually side with the person (Aaron F) who said that you might be interested in the Levenshtein distance problem (and cited this). His answer seems to me to be the best so far.
More specifically, Levenshtein distance (also called edit distance), does not measure strictly the character-by-character distance, but also allows you to perform insertions and deletions. The best algorithm for this distance measure can be computed in quadratic time (pretty slow if your strings are long), but the computational biologists have pretty good heuristics for this, that might be of interest to you on their own. Check out BLAST and FASTA.
In your problem, it seems that you are dealing with differences between strings of numbers, and you care about the numbers. If you give more information, I might be able to direct you to the right variant of BLAST/FASTA/etc for your purposes. In any case, you might consider adapting BLAST and FASTA for your needs. They're quite simple.
1: http://en.wikipedia.org/wiki/Levenshtein_distance, http://www.nist.gov/dads/HTML/Levenshtein.html