Question on Time Variant

Hello All
How is the time variants gets the data, Do we need to LOAD data from Source system, like how we do for Master Data Load ?
Thanks, looks silly questions, but good to understand ....
BI

Hi,
In rsa1 of BW goto the source systems. Right click on your source system and select Transfer Global Settings. This will do your job. You need not to LOAD any data from Source system.
Hope this Helps,
Thanks,
Amit

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Issues with using SV Time-Varying Loudness.vi

Hi,
I am rather new to Labview and am having issues getting my Time-Varying Loudness calcuations to come out correctly. Attached is my current VI and a .csv file of a set of my data (calibrated in Pa. Sample rate = 50kHz for 5 seconds). When i run the VI it does not output what I would expect. Here are my questions:
Time-Varying Loudness:
1) What are there so many signals? There are hundreds of lines, but I only expected one.
2) Why is my x-axis not a range of 1 to 24 like Barks should be?
1/3 Octave Band Analsysis:
1) Why does it look incorrect? Their shouldnt be a linear increase as a function of frequency...
The data is of a part being squeezed and making a squeek. So it is a transient noise that happens around 2 seconds. Additionaly, how would I go about making a waterfall (Time(s),Frequency(Hz),Amplidtude(sones)) of this signal? Is there any easy way? or do I have to make all three signals and add them to make one plot?
Thank you for your help! Let me know if you need more infromation,
-Troy
p.s. i included a picture of the results as well.
Solved!
Go to Solution.
Attachments:
TimeVaryingIssue.zip ‏668 KB

Hi Troy,
1) The SV Specific Loudness VI "chunks" data into 2 ms blocks, and then returns these as individual Specific Loudness vs. Sone plots. The colored lines you are seeing on your graph are representative of the 2,500 2 ms time periods within your five-second acquisition. Each of these plots contains 241 points, however they should occur in ten sample "steps", one for each sone. The digital filter's buffer takes a bit to fill and kick in properly, however, so you may want to give your first few rows( 0-.02 seconds or so) a closer look before using them- you may see a number of unexpected zeroes around the lowest sones.
2) Since you are using a chart, the x-axis will increment with every subsequent run (the previous data is retained.) If you only want to display the most recent data, I would recommend switching to a graph, which can also be found in the graph controls palette. Also, see above (#1) for why the x-axis is longer than 24 points.
3) Remember that dB is a unitless measure, and can only be used in reference to another value. A typical reference for sound is 20 uPa, however you will likely need to equalize your input data and determine/set a dB reference. You may want to take a look at SVL Scale voltage to EU.vi (EU stands for equalized units) and SVL Set dB reference.vi. In your case, I believe the Octave plot is showing your dB relative to a default value of 1. The values should be accurate with respect to one another, but you will need to provide a reference value to calibrate the scale.
4) I think that this VI serves as a good example of how to go about creating a waterfall plot:
Waterfall Display for Octave (DAQmx)
http://zone.ni.com/devzone/cda/epd/p/id/5562
You will need to break up your waveform into chunks, much like the specific loudness VI, but the basic concept is roughly the same (take waveform chunk, take octave measurement of chunk, append octave measurement output cluster to array of clusters, repeat for remainder of waveform, display)
Phew! That was quite a bit of information. Let me know if that makes sense, and don't be intimidated by the detail and/or unfamiliar functions or methods - you're definitely on the right track, or at the very least asking the right questions.
Tom L.

Frequency analysis of time varying data

I want to add a routine for frequency analysis of before and after strong impulse on a body.
Some vibration sensors will be attached to the body.
In this case, I was told I need to use a time-varying method like wavelet or any other to analysis instead of simple FFT.
I installed wavelet package but I have no idea how to use it.
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labmaster.

labmaster wrote:
I want to ask some curious things.
1. I have to analysis the spectrum of frequency in dynamic signal.
In case of presenting dB, how to know the frequencies?
2. In high sampling rate (1MHz), the low frequency element can't be analyzed. (too small window for 60Hz in my last example).
If so, generally, do I have to down sampling to ~1kHz for the 1MHz in wavelet or short time FT?
Take a look on how the Express VI you are using are made. The Express VI you are using is located here
C:\Program files\National Instruments\LabVIEW 2010\vi.lib\addons\Time Frequency Analysis\_express\_TimeFrequencySpectrogramConfig.llb
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To answer your first question. The dB setting will change the Y axis scaling, and not the X axis scaling. Which is the frequncy axis
The step between each frequncy bin will be( number of Frequency bins/(Fsample/2)). So using 1 MHz may reduce resolution. Perhaps 10 KHz will be bettter for you to use.
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Why you want to evaluate the time for inactive person.
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