I'm trying to understand how MINPEAKDISTANCE works. I returned to the documentation, here, but it wasn't very clear how this parameter works.
Can you kindly clarify it a bit?
Thanks.
Minimum peak separation Specify the minimum peak distance, or minimum
separation between peaks as a positive integer. You can use the
'MINPEAKDISTANCE' option to specify that the algorithm ignore small
peaks that occur in the neighborhood of a larger peak. When you
specify a value for 'MINPEAKDISTANCE', the algorithm initially
identifies all the peaks in the input data and sorts those peaks in
descending order. Beginning with the largest peak, the algorithm
ignores all identified peaks not separated by more than the value of
'MINPEAKDISTANCE'. Default: 1
So if you consider your peak heights as values in the "y" direction, then the separation that this is talking about is in the "x" direction. So for example look at this image (from Matlab docs and if you have the image processing toolbox you can get the data too load noisyecg.mat):
lets say you just want to identify thos 4 big distinct peaks, but not the hundreds of little peaks caused by noise, setting MINPEAKDISTANCE is a feasible way accomplish this because the noisy peaks are at a much higher frequency, i.e. they are closer to each other in the "x" direction, or have a smaller distance separating them than the big peaks do. So choosing a large enough MINPEAKDISTANCE, say 100 or 350 for example depending on what peaks you're interested in, would help you to not detect these undesired noise peaks.
Try findpeaks on this data with different MINPEAKDISTANCE values and see what you get!
If you've got noisy data, you may find that instead of one solid peak, you get lots of small ones (see the folowing image).
The important data here is when the signal is high and when it is low - you don't care about small variations in value, you only want to use one of those peaks and not look at all the smaller local ones around it. If you know the frequency of your signal (i.e. how often the peaks should occur), you can tell the function to ensure that the peaks are separated by a certain amount.
In the above example, the peak is every 15 milliseconds and lasts for 5 milliseconds, so you might set your MINPEAKDISTANCE parameter to 15 or so.
Related
I'm dealing with CWT, and I have a big problem converting scales to frequencies. In the MAtlab Wavelet Tutorial they use this expression to convert scales to frequencies
But if i use the default function scal2freq I obtain different result.
I don't understand the role of the Morlet Fourier Factor
Thanks in advance
It is a pretty complicated concept, which I somewhat understand it. I'll write some points here so that you might figure it out yourself, rather easier.
A simple fact is that:
Scale is inversely proportional to frequency.
For example, imagine we have a 1-100 Hz range of frequencies in some time series data such as stock markets data or earthquake data. Scale is "supposed to be" the inverse of that. For instance, if scale would be in range of 1 to 100, we'd have had:
Scale(1/Hz) Frequency (Hz)
1 100
50 50
100 1
Therefore,
The frequency is not the real frequency of those time series data (e.g., stock market, earthquake) that we know of. They are only related, inversely.
And we can safely say that here we are calculating some "pseudo-frequencies", which MATLAB does that (by approximating that). You can read about the approximation process in the documentation in the section pseudo-frequencies:
MATlAB does calculate those pseudo-frequencies based on:
In wavelet analysis, the way to relate scales to frequencies is to determine the center frequency of the wavelet function:
which you can visually see in this image and of-course it would differ, when we would change the types of our function in the calculation. Thus, that center frequency will change everytime in our approximation process:
That "MorletFourierFactor" is a variable to approximate a constant so that when you would do the 1/scale, it would closely approximate those "pseudo-frequencies".
I thought this image about shifting (time axis) and scaling (frequency axis) might be a little helpful to look into as well:
The bottom line is that don't worry about pseudo-frequencies, you wouldn't probably need those. If you would want any frequency spectrum, you can likely go towards applying some of those frequency methods (such as Fast Fourier Transform) on whatever time series data that you have.
If you really really want to map that, you can also try to design some methods to approximate it yourself.
Source
Harvard Seismology
I'm trying to get all large peaks values of this signal :
As you can see there is one large peak followed by one smaller peak, and I want to get each value of the largest peak. I already tried this [pks1,locs1] = findpeaks(y1,'MinPeakHeight',??); but I can't find what I can write instead of the ?? knowing that the signal will not be the same every time (of course there will ever be a large+smaller peak schema but time intervals and amplitudes can change). I tried a lot of things using std(), mean(),max() but none of the combination works properly.
Any ideas on how can I solve the problem ?
You could try using the 'MinPeakDistance' keyword and enter a minimum distance between the two peaks slightly higher than the distance between the large peak and the following small peak. So for example:
[pks1,locs1] = findpeaks(y1,'MinPeakDistance',0.3);
Edit:
If the time between peaks (and the following smaller one) varies a lot you'll probably have to do some post-processing. First find all the peaks including the smaller second ones. Then in your array of peaks remove every peak which is significantly lower than its two neighbours.
You could also try fiddling with 'MinPeakProminence'.
Generally these problems require a lot of calibration for the final few percent of the algorithms accuracy, and there's no universal cure.
I also recommend having a look at all the other options in the documentation.
I'm a neuroscientist, and not a very good one. My colleague has kindly provided me with a noisy voltage measurements of the PY neuron of the Stomatogastric Ganglion of the lobster.
The activity of this neuron is characterised by a slow depolarised plateaux with fast spikes on top (a burst).
Both idealised and noisy versions are presented here for you to peruse at your leisure.
It's my job to extract the spike times from the noisy signal but this is so far beyond my experience level I have no idea where to begin. Fortunately, I am a total ninja at Matlab.
Could someone kindly provide me with the name of the procedure, filter or smoothing function which is best suited for this task. Or even the appropriate forum to ask such an asinine question.
Presumably, it needs to increase the signal to noise ratio? The problem here seems to be determining the difference between noise and a bona fide spike as the margin between the two is quite small.
UPDATE: 02/07/2013
I have tried the following filters in Matlab with mixed results. It's still very hard to say what is noise and what is a spike.
Lowpass Butterworth filter,
median filter,
gaussian,
moving weighted window,
moving average filter,
smooth,
sgolay filter.
This may not be an adequate response for stackoverflow - but one way of increasing a signal to noise ratio in your case is to average parts of the signal.
low pass your signal to remove noise (and spikes), and find the minima of the filtered signal (from your image, one minimum every 600 data points). Keep the indexes of each minimum,
on the noisy signal, for each minimum index, select the consecutive 700 data points. If you have 50 minima, you should have a 50 by 700 matrix,
average your matrix. You should have a 1 by 700 vector.
By averaging parts of the signal (minimum-locked potentials), you will take advantage of two properties: noise is zero-mean (well, it should be), and the signal of interest is repetitive. The first will therefore decrease as you pile up potentials, and the second will increase. With this process however, you will lose the spike times for each slow wave figure, but at least have them for blocks of 50 minima.
This technique is known in neuroscience as event-related potential (http://en.wikipedia.org/wiki/Event-related_potential). It may not fit perfectly your signal, or the result may not give nice spikes, but you may extract the spike times for some periods of interest (given the nature of your signal, I would say that you would need 5 or 10 potentials to see an emerging mean activity).
There are some toolboxes that do part of the job (but I would program it myself given the complexity of the task). These are eeglab or fieldtrip. They have a bunch of filter/decomposition options too, as well as some statistical features.
I have discrete empirical data which forms a histogram with gaps. I.e. no observations were made of certain values. However in reality those values may well occur.
This is a fig of the scatter graph.
So my question is, SHOULD I interpolate between xaxis values to make bins for the histogram ? If so what would you suggest to be best practice?
Regards,
Don't do it.
With that many sample points, the probability (p-value) of getting empty bins if the distribution is smooth is quite low. There's some underlying reason they're empty, which you may want to investigate. I can think of two possibilities:
Your data actually is discrete (perhaps someone rounded off to 1 signficant figure during data collection, or quantization error was significantly in an ADC) and then unit conversion caused irregular gaps. Even conversion from .12 and .13 to 12,13 as shown could cause this issue, if .12 is actually represented as .11111111198 inside the computer. But this would tend to double-up in a neighboring bin and the gaps would tend to be regularly spaced, so I doubt this is the cause. (For example, if 128 trials of a Bernoulli coin-flip experiment were done for each data point, and someone recorded the percentage of heads in each series to the nearest 1%, you could multiply by 1.28/% to try to recover the actual number of heads, but there'd be 28 empty bins)
Your distribution has real lobes. Because the frequency is significantly reduced following each empty bin, I favor this explanation.
But these are just starting suggestions for your own investigation.
I have FFT outputs that look like this:
At 523 Hz is the maximum value. However, being a messy FFT, there are lots of little peaks that are right near the large peaks. However, they're irrelevant, whereas the peaks shown aren't. Are the any algorithms I can use to extract the maxima of this FFT that matter; I.E., aren't just random peaks cropping up near "real" peaks? Perhaps there is some sort of filter I can apply to this FFT output?
EDIT: The context of this is that I am trying to take one-hit sound samples (like someone pressing a key on a piano) and extract the loudest partials. In the image below, the peaks above 2000 Hz are important, because they are discrete partials of the given sound (which happens to be a sort of bell). However, the peaks that are scattered about right near 523 seem to be just artifacts, and I want to ignore them.
If the peak is broad, it could indicate that the peak frequency is modulated (AM, FM or both), or is actually a composite of several spectral peaks, themselves each potentially modulated.
For instance, a piano note may be the result of the hammer hitting up to 3 strings that are all tuned just a tiny fraction differently, and they all can modulate as they exchange energy between strings though the piano frame. Guitar strings can change frequency as the pluck shape distortion smooths out and decays. Bells change shape after they are hit, which can modulate their spectrum. Etc.
If the sound itself is "messy" then you need a good definition of what you mean by the "real" peak, before applying any sort of smoothing or side-band rejection filter. e.g. All that "messiness" may be part of what makes a bell sound like a real bell instead of an electronic sinewave generator.
Try convolving your FFT (treating it as a signal) with a rectangular pulse( pulse = ones(1:20)/20; ). This might get rid of some of them. Your maxima will be shifted by 10 frequency bins to teh right, to take that into account. You would basically be integrating your signal. Similar techniques are used in Pan-Tompkins algorithm for heart beat identification.
I worked on a similar problem once, and choosed to use savitsky-golay filters for smoothing the spectrum data. I could get some significant peaks, and it didn't messed too much with the overall spectrum.
But I Had a problem with what hotpaw2 is alerting you, I have lost important characteristics along with the lost of "messiness", so I truly recommend you hear him. But, if you think you won't have a problem with that, I think savitsky-golay can help.
There are non-FFT methods for creating a frequency domain representation of time domain data which are better for noisy data sets, like Max-ent recontruction.
For noisy time-series data, a max-ent reconstruction will be capable of distinguising true peaks from noise very effectively (without adding any artifacts or other modifications to suppress noise).
Max ent works by "guessing" an FFT for a time domain specturm, and then doing an IFT, and comparing the results with the "actual" time-series data, iteratively. The final output of maxent is a frequency domain spectrum (like the one you show above).
There are implementations in java i believe for 1-d spectra, but I have never used one.