Does Strain task in DAQmx include lowpass filter?

I am using a load cell with a 2.232 mv/V with an excitation voltage of 10V. When I put a scope to the output, there is a lot of high frequency noise which causes me to have to use an amplifier with a lowpass filter built in.
I would like to be able to accomplish this same task without having to purchase the amplifier (I have 3 load cells per machine, 6 machines so cost does add up). I am currently using a PCIe-6320 for some encoder measurements on this same application and did find where I can use the analog inputs to measure Strain using DAQmx.
My question is does the Strain measurement in DAQmx have a built in lowpass filter to clean up the signal? If not, is there a solution to this that does not require a Toolkit or additional module for signal processing? We have the basic LabView 2009 SP1.

aeastet wrote:
I would also look at why you are getting so much noise. How do you have your channels hooked up to the system?
Yes.
Using a strain guage module the bridge is built into the terminal block and the hardware does the mv/V thing so there must be some place the noise is entering the system, long, non-shielded, non-twisted pairs, bad ground...
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

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Noisy Analogue Signal and Lowpass Filter Questions

Hi everyone,
I am getting analogue signals from a strain gauge that are a bit more noisy than I would like... Normally, I jump right in an smooth the signal using a lowpass filter of whatever order gave me the result that was acceptable, without overkill and smootig the signal to nothing... However, I am settingup a new DAQ device and would like to a little more informed about what and how am doing this...
So, here are my questions:
1. How do I determine the filter order? I usually use a 2nd order filter for smoothing.. I understand the comncepts of roll-offs and phase delay as they relate to the order numbers... On what basis should I decide which order filter to use for this specific signal?
2. I am still thinking that using a lowpass filter is the best approach... I assume the lower cutoff freq is in Hz (i.e. 10Hz) and is NOT a normalised freq [i.e. cutoff freq / (sample rate/2)]?.. The default value is 0.125, which seems too low for a Hz value? Also, on what basis should I decide what cutoff freq to use for this specific signal?
3. I need to synch the filtered strain gauge signal with other, unfiltered signals... But as the filter will add a phase delay it will affect signal synchronisation... I heard somewhere that a 4th order filter is zero-phase, but I can't see how that could be right as the filter only operates in one direction... Any suggestions for what I could do to maximise the timing alignment between the filtered and unfiltered signals?
All comments welcome!
Regards,
Jack

Jack,
I was the Instrumentation Engineer at a university for 24 years and worked with several execise physiologists, biologists, and neuroscientists on similar projects. My group was the "lab/tech staff" you have little of. There were 3.5 (engineer, 1.5 electronics technicians, and machinist) of us for the entire university.
OK. The strain gauge signals probably have no information above a few tens of Hz (hummingbirds) or a few Hz for most larger species. EMG is usually filtered in the amplifiers to slightly larger bandwidths (few hundred Hz) unless you are trying to see the individual action potentials. Does the dynamometer have specifications on the bandwidth or sampling rates used?
Since your bandwidths are realtively low, I suggest that you sample all the analog inputs (and the trigger outputs, if the boards have the capability) from the same clock. You can later decimate or average data on channels where less data is needed. Record the trigger signals on an unused analog input if you have one available. It can be very handy to have the information about when the triggers actually occurred in the same dataset as the response data.
Delay equalization is a complex topic. Suppose that you are sampling all your channels at 1 kS/s. Suppose you have a 10 Hz low pass filter, 2 pole Butterworth, on a strain gauge channel and a 50 Hz filter, 4 pole, elliptical, on an EMG channel and you want to synchronize the filtered data with some unfiltered channels. The two filters have very different phase and delay responses over the 0-500 Hz frequency range. To equalize you would need to determine the maximum delay at any frequency in the range for both filters. Then you would need to design THREE all-pass filters. One would have a phase response which compensates the 10 Hz filter. Another would compensate the 50 Hz filter. These would need to be designed so that the total delays through the low pass plus all-pass filters is the same. Then you need a third all pass filter with that same total delay to compensate the unfiltered data. This is may be a non-trivial job for an expert, depending on the low pass filters.
What kind of noise are you seeing and how is it affecting your measurements? Can you post some data?
Lynn

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It's written that for the NI 62xx serie M DAQs, a lowpass filter can be programated (to prevent noise or aliasing), but I can't find where I can configure it ...
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Mathieu

Bonjour,
Effectivement, la valeur du filtre de la carte NI-6280 est 40kHz.
En simulant la carte, j'ai pu faire le test avec le VI joint.
Si ce VI ne fonctionne pas, quelle est le code d'erreur obtenu et à quelle fonction apparaît-il ?
Salutations,
Isabelle
Ingénieur d'applications
National Instruments France
Attachments:
filtre 6280.vi ‏73 KB

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Attachments:
Counter_1_Meas Buffered Semi-Period-Continuous_main_lv09.vi ‏34 KB
SemiPeriod_Reconnect Counter on Error.vi ‏35 KB

Lowpass filter in modulation

hello
i'm using LABVIEW to implement LSB-SSB modulation, this requires a VSB signal passed through a lowpass filter, the problem i couldn't get my required spectrum, i wonder if the problem is in the filter or elsewhere, i hope you can help me in that , i only need to know how the filter in labview works , thanks
here is the right attachment (the previous one was wrong sorry
Solved!
Go to Solution.
Attachments:
ssb1.png ‏71 KB

The realities of real filters!
1. It is not possible to implement an ideal filter in the real world. *
2. All filters, hardware or software, have a transient when the signal is first applied (or changed).
3. The narrower the bandwidth or the sharper the cutoff, the more significant the transient.
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Once I am adding VAR_VALUE_1 to the static parameters, the system generates two messages error:
Program error in class SAPMSSY1 method : UNCAUGHT_EXCEPTION
System error in program SAPLRRS2 and form GET_SID-1
Best Regards,
Hanit

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