Waveform averaging

Hello.
I have to acquire noisy sine signal (about 30 kHz) with NI PXI-5105 60 MS/s digitizer. I tried to make a VI using LabVIEW help but got the following result:
Apparently, it is necessary to make waveform averaging but I don't know how to do such a thing.
Here shows how signal is indicated:
I found here the following example from RavensFan, but could not use it:
There is a type mismatch:
Is there somewhere a detailed description of the LabVIEW's waveform / complex waveform / … data types with examples? I so stupid and standard LabVIEW help did not help me.
Thanks in advance for your help and sorry for my poor english.

Try:
Extract Single Tone Information VI
Owning Palette: Waveform Measurements VIs
Requires: Full Development System
Takes a signal in, finds the single tone with the highest amplitude or searches a specified frequency range, and returns the single tone frequency, amplitude, and phase. The input signal can be real or complex and single-channel or multichannel. Wire data to the time signal in input to determine the polymorphic instance to use or manually select the instance.
and
Averaged DC-RMS VI
Owning Palette: Waveform Measurements VIs
Requires: Full Development System
Calculates the DC and RMS values of an input waveform or array of waveforms. This VI is similar to the Basic Averaged DC-RMS VI, but this VI gives more precise control over the individual DC and RMS calculations.
Do you really want the RMS or the amplitude of the 30kHz signal??
Have a look at the (power) spectrum of your signal .. will help you identify (and eliminate) the noise.
Greetings from Germany
Henrik
LV since v3.1
“ground” is a convenient fantasy
'˙˙˙˙uıɐƃɐ lɐıp puɐ °06 ǝuoɥd ɹnoʎ uɹnʇ ǝsɐǝld 'ʎɹɐuıƃɐɯı sı pǝlɐıp ǝʌɐɥ noʎ ɹǝqɯnu ǝɥʇ'

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Hello everybody ,
im working on a VI to average several EEG waveforms ...which is attached ..
actually each waveform entering this ' waveform averaging' VI is actually an EEG reponse to each sound click we present to the subject.
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Attachments:
AverageWaveforms2.png ‏6 KB

Hi, I'm not LV expert but I think I run into some thing like yours before. What I did was checking any missing pulse or abnormal in the EEG waveform and set trigger pulse once I found it. Take a look at your signal amplitude. If your signal too low (ex: <1.25V), your counter gate won't trigger. Once you record your EEG, amplified it and use trigger VI (take a look at the LV example) to set your trigger event pulse.
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Sorry,I really don't know if my question belongs in this category.Please see the attached file.There are 105 data points and they are plotted as the waveform.
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Attachments:
waveform.jpg ‏53 KB

Greetings!
I understand your question fine There are a couple of things you want to do. First, you might just try a different INTERPOLATION. The graph pallette gives you a few choices in this regard, the default is linear. If that doesn't work sufficiently, you want to try waveform averaging. Depending on how many points you average, you can eliminate "local" peaks while retaining longer peaks. There is a Waveform Average vi. in the NI code archives..(I think it's the one I submitted myself a while back!) Keep in mind that when you average a waveform, your peaks will be offset by half of the averaged samples....no way of gettin around this without "cheating". Hope this gives you some ideas. And yes, this can be done in Matlab quite nicely, and t
here is probably already a waveform average in the Matlab Tools.
Goood Luck
eric
Eric P. Nichols
P.O. Box 56235
North Pole, AK 99705

How to calculate average of step-shaped waveform?

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(ex. attached image)
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Attachments:
Untitled.jpg ‏6 KB

I did something like this in my Matlab days. I can give you a general idea on how I accomplished it.
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Josh
LabVIEW CLD
Software is never really finished, it's just an acceptable level of broken

Average value for waveform chart

Hello everyone.
Can i make a time interval, where i can measure average value for a waveform chart ?
for example fom 0:30 to 0:45
I have uploadet my VI
Thanks in advance
Oesen
Best regards
Oesen
Solved!
Go to Solution.
Attachments:
Trykmaaling.vi ‏83 KB

Oesen,
When you are displaying data on a chart, the newest data is appears at the right of the previous data. When the chart history buffer is full (default length = 1024), adding a new point deletes the oldest point. Visually it appears that the data slides to the left as new data is added to the right end.
My question relates to how the user would "tell" the program what data to average. In the image you posted you marked a segment of the data to be averaged. Suppose the user marks point "A" as the beginning of the segment. While the user is deciding where to put mark "B" additional data is acquired and point "A" disappears because it is beyond the history length. What does the program do in such a case? A slight variation is the situation where both "A" and "B" have been marked but some of the data is deleted from the history before the averaging or other processing has begun.
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Attachments:
Gait Waveform.pdf ‏39 KB
Trial Data.zip ‏1189 KB

Hi MJG3,
What you are trying to do is definitely possible with DIAdem and while it is a lot easier than it would be in other languages, it isn't necessarily simple - especially if you don't have much experience with VBscripting. I took a look at the data that you attached and you will definitely need to create an algorithm to filter out each different section of your data to create separate graphs. In order to actually create these graphs, it is probably easier to record a macro in the Scripting tab that creates a subset of data and then adds it to your report (or wherever you need it) than to write it from scratch. Then, from there, I would use the algorithm you develop to determine where to put each data point.
As far as the actual algorithm goes, I have a general idea of how you might want to do this. You could use a 'filter' (if/else if) statements that look at the following parameters to determine whether you are walking, standing, localized, or inactive.
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Finally, any data that is not accounted for is 'localized' which appears to be more random than the other three data sets.
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I wish you the best of luck on your project!
Regards,
Trey C.

Does anyone know how I can average several waveforms?

I am trying to filter a nasty square wave,without affecting the rise or fall times of it. I have tried standard filtering, which helps with the noise, but is destructive to the rise and fall times. I have also tried taking the median of the signal, which is effective to a point, but I would like to try taking several single cycle or multiple cycles and averaging the the group of captured cycles. Just so its clear, I don't want the rolling average of each of the cycles, I want the point by point average of the group of averaged signals, as if they were laid on top of each other and each point is then averaged. Can anyone help me with this? I will attach a waveform example below, 3 captured cycles in wdt form,
just in case anyone has any has any good ideas about filtering without affecting rise or fall times.
Attachments:
0_to_31_1deg_open_1_gain ‏119 KB

The signal is a square wave with finite rise and fall times that vary depending on grey level. The rise time also is not the same as the fall time. There is little overshoot or ringing to deal with, but there is some 60 Hz. noise associated with the refresh rate. I had an idea similar to yours. Instead of actually capturing a full signal,what about capturing just a single transition. If I were to start capturing a little before I caused a grey level change (which corresponds to a signal level change from high to low or the reverse) and continue the capture for a little after the transition, there would be a high point to average, and a low point to average, and if a straight slope could be attached between the
two time points that matched these everaged values, then a rise or fall time could be determined. For that matter, it might even be easier to just quatify that time in between thoses two averaged values. I looked around the functions library quite a bit and could figure out no way to do this. If you have any ideas, they would be appreciated. I am including a print screen copy of the waveform, see what you think.
Attachments:
wavefrmexmple.doc ‏140 KB

Would anyone know how to program the Labview code for the following: Recording an analog input signal for 100msec and then repeating this 100 times in order to finally have an averaged signal (waveform)?

I have added a vi, but it doesn't quite work. Would be very grateful if someone could help, thanks!
Attachments:
Average_0411041_2.vi ‏77 KB

The attached VI collects 100 ms of data and does a rolling average. Since your sub-VI were not included I created this from scratch. There are comments in the code. I hope this is what you need.
Dirk
Attachments:
rollingaverage110x100.vi ‏47 KB

Channel name lost when averaging waveform

Hi, I am writing an application where data is acquired for DAQ continuously. Data is averaged for example every 1ms and then written to file. Problem is that channel names are lost when doing averaging. I am using 'averaged DC-RMS.vi'. I could copy the attribute before averaging and then write them back afterwards, but doesn't sound like good solution for streaming data application with several channels. Is this a bug or feature? Is there any other way to maintain channel names?
BR Marko
Here is sample picture:
Attachments:
Lost Channel name.png ‏29 KB

Hi Marko,
I have toiled with this same problem, without finding a
solution. It seems trivial enough, yet the answer still eludes me. Could you please post your code for others to
learn from.
Thanks
Alan

Trouble capturing waveform from PXI-4472

I'm really a very green newbie at this stuff, so bear with me...
I've got a PXI-4472 data acquisition board and a PXI-5411 waveform generator. I've connected the arbitrary out of the 5411 to the channel 0 in on the 4472. An external oscilloscope shows a 1v-amplitude sine wave being generated.
I created a very simple VI to show what the 4472 is capturing. It connects a NI-DAQ channel I generated to the standard "AI Acquire Waveform.vi", then out to a Waveform Chart, all within a while loop with a Stop button. Problem is, all the waveform chart seems to be showing is the running average of the waveform instead of the form itself (solid line, a tad above zero).
I can hook the 4472 input channel up to a DC-out power supply, a
nd when I vary the voltage, the waveform chart changes as well.
So my question (whew!): What's wrong here that's not allowing me to capture a waveform from the 4472 (in turn from the 5411) and display it on my waveform chart?
Thanks in advance for the help.

Never mind.... it was a sample rate problem. I upped the sample rate and it came out ok.

Triggerring PXI-4110 to measure 1 current value while HSDIO PXI-6552 generating waveform

Hi,
Some question about PXI-4110 to measure current while PXI-6552 is generating the waveform.
1. Let say, I need to measure 3 points of current values, i.e. while PXI-6552 is generating sample-1000, 2000 and 3500. On the edge of sample 1000,2000 and 3500, the PXI-6552 will send a pulse via PFI line or via PXI backplane trigger line. My question is, is it possible to trigger PXI-4110 (hardware trigger or software trigger) to measure current values at these points ?
2. Let say I need to measure the current on 0ms (start of waveform generation by PXI-6552) , 1ms, 2ms, 3ms, 4ms... and so on for 1000 points of measurement, code diagram as shown at the figure below. It is possible for the VI "niDCPower Measure Multiple" to measure exactly at 1ms, 2ms, 3ms .. ? How much time will have to spend to complete acquire 1 point of measurement by "niDCPower Measure Multiple" ?
Thanks for viewing this post. Your advice on hardware used or software method is much appreciated. Thanks in advance.
Message Edited by engwei on 02-02-2009 04:24 AM
Attachments:
[email protected] ‏46 KB

Hi engwei,
1. Unfortunately, the 4110 does not support hardware triggering. Therefore you cannot implement direct triggering through the backplane or anything like that. However, there are a couple of possible workarounds you can try:
a) Use software triggering: Say your 6552 is generating in one while loop, and your 4110 is to measure in another while loop. You can use a software syncrhonization method like notifiers to send a notification to your 4110 loop when your 6552 has generated the desired sample. This method, however, will not be very deterministic because the delay between the trigger and the response depends on your processor speed and load. Therefore, if you have other applications running in the background (like antivirus) it will increase the delay.
b) Use hardware triggering on another device: If you have another device that supports hardware triggering (like maybe an M-series multifunction DAQ module), you can configure this device to be triggered by a signal from the 6552, perform a very quick task (like a very short finite acquisition) then immediately execute the DCPower VI to perform the measurement. The trigger can be configured to be re-triggerable for multiple usage. This will most likely have a smaller time delay then the first option, but there will still be a delay (the time it takes to perform the short finite acquisiton on the M-series). Please refer to the attached screenshot for an idea of how to implement this.
2. To make your 4110 measure at specific time intervals, you can use one of the methods discussed above. As for how long it will take to acquire 1 measurement point, you may find this link helpful: http://zone.ni.com/devzone/cda/tut/p/id/7034
This article is meant for the PXI-4130 but the 4110 has the same maximum sampling rate (3 kHz) and so the section discussing the speed should apply for both devices.
Under the Software Measurement Rate section, it is stated that the default behavior of the VI is to take an average of 10 samples. This corresponds to a maximum sampling rate of 300 samples/second. However, if you configure it to not do averaging (take only 1 sample) then the maximum rate of 3000 samples/second can be achieved.
It is also important to note that your program can only achieve this maximum sampling rate if your software loop takes less time to execute than the actual physical sampling. For example, if you want to sample at 3000 samples/second, that means that taking one sample takes 1/3000 seconds or 333 microseconds. If you software execution time is less than 333 microseconds, then you can achieve this maximum rate (because the speed is limited by the hardware, not the software). However, if your software takes more than 333 microseconds to execute, then the software loop time will define the maximum sampling rate you can get, which will be lower than 3000 samples/second.
I hope this answers your question.
Best regards,
Vern Yew
Applications Engineer, NI ASEAN
Best regards,
Vern Yew
Applications Engineer
Attachments:
untitled.JPG ‏18 KB

Can I use one waveform to subtract from the other one in labview?

Here is the thing.
I am using DAQ to aquire amplified signal to measure its noise of output.However DAQ would contribute a little noise to the whole system.My thinking is to capture two waveforms of noise from DAQ itself and the whole system(amplified circuit and DAQ) and then make the subtration calculation between them.Is it possible to make it with labview?How?
Or can I aquire two groups of signal using DAQ at the same time? If it can't be realised is there some other way?
Thanks

If I'm reading you right I don't think your original concept is valid. If you have already done a good job with wiring, shielding, etc. You won't be able to simply subtract a data collection of "just noise" from a collection of "noise plus signal" and expect the noise to cancel itself out. It's random. A better method would be to collect the same signal several times and then average the results. Even that will only reduce repetitive components in the noise.
As smercurio said however, the most important consideration is to make sure your connection method, grounding, shielding, timing, etc is top notch to prevent noise from contaminating the signal in the first place. I generally use differential signals through good quality twisted pairs with the shield connected on the DAQ side only (to prevent ground loops).
Using LabVIEW: 7.1.1, 8.5.1 & 2013

Synchronize two waveforms with different time

Hi All,
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<>< Eric
Attachments:
IMS Software v5.4.vi ‏158 KB

Hello Eric,
I see that you are creating a waveform using the Build Waveform VI and then wiring it to the outputs in the Chromatography Mode tab - namely 'Waveform Graph' and 'Waveform Graph 3'. Is the 'dt' value constant for every iteration of the while loop? If so, your output should have constant time spacing between plotted values in the waveform graph.
So, again I am not completely sure what you mean by the IMS not operating at the frequency you input :-(. Are you seeing the problem in 'Waveform Graph' or 'Waveform Graph 3' or both? An simple example VI would be great to describe your issue.
Vivek Nath
National Instruments
Applications Engineer
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Counter/Timer Waveform

I would like to convert period measurements (RPM data) received from my counter/timer board into waveform data type so that I would be able to conduct frequency analysis on the data (e.g. FFT). Have tried with limited success interpolation & decimation techniques. Would be grateful for any suggestions.

Malfesto,
I had a look and can offer up a few comments:
1. I don't understand your comment about "truncation error" in the slow speed pulses case. Perhaps your are remarking about the "glitches" in the slow data embedded in the block diagram array? More on this a bit later...
2. At least at a quick glance, the interpolation seems a little funky. I didn't spend a ton of time wrapping my head around it so maybe it's actually ok, but I saw a finite difference that would act like a derivative of the period, and a scaling factor based on the average period -- things that make me go hmmmm.
Also, if you zoom on the graphs for the Slow Data, you can kinda see that the "Post Processing Graph" looks like a derivative of the "Pre Processing Graph". I suspect you need to rework your interpolation method.
3. When I did a similar project, I think I first converted all the periods to frequencies and also generated a "sample time" array that was the cumulative sum of the periods. (I toyed with tweaking the time array so that the timestamp would correspond to the midpoint of the interval that I had converted to a frequency. I don't recall now whether that tweak proved useful enough to bother with).
Once I had laid out arrays for frequencies and (unequally-spaced) sample times for those frequencies, I could generate an equally-spaced time array at a desired "virtual sampling rate" and interpolate to create new "virtual frequency" values. Since I was merely post-processing data, I think I also used the native LabVIEW array functions 'Threshold 1D Array' and 'Interpolate 1D Array' in the interpolation process.
Now, before going into an FFT you'll want to remove the huge DC offset. This may be as simple as subtracting the mean or median frequency from all entries if you are nominally at constant speed.
4. Your raw dataset for Slow Data has suspiciously periodic glitches in the periods. There is a very strong and dominating pattern that repeats every 16 intervals. Your typical period is registering about 10 million timebase cycles. However, in every set of 16 periods, you get one period around 27 million and another around 35 million.
I don't know how you gathered your "Slow Data" values, but there seems to be a systemic error in the measurement. That sort of one-sample discrepancy looks very suspicious. I'd urge you to determine the source of those glitches and try to eliminate them. They don't appear to be telling the truth about the motion you wish to characterize. The device you measure doesn't by chance start and stop or go back-and-forth during your measurement, does it?
Hope this helps.
-Kevin P.

Waveform averaging

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