OLED display & signal sampling rate

Dear Ni technicians
I am new to LabVIEW Embedded module. I use LabVIEW 2009 and Luminary LM3S8962 evaluation board. Recently my project need to use this board for data acquisition and display the input the waveform on the OLED display. However I find that the sampling rate is as low as around 10Hz, hence I can only get the waveform below 2Hz without aliasing. While the datasheet claim that the sampling frequency is about 16MHz. Is it because the display rate "drag" the "sampling rate"? if so, how could I improve it?
Here I attach the code and the actual result in the ppt file.
Many thanks!
Best regards,
Haiwen
Attachments:
testing.ppt ‏4333 KB

Hi, I'm working on it, I wonder if at this time if you manage to increase the sampling rate?
Atom
Certified LabVIEW Associate Developer

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Display file type, bit rate, sample rate in 12.0.1?

I just upgraded to 12.0.1 running on an old dual quad core running OS10.7.5
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One important aspect was to view the file type, bit rate, and sample rate as I'm a band/audio producer.
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Hello,
i am using first time labview signal express together with a NI pci 6251 & NI Scc 68 device.
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hello markus_umd,
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Hello everyone,
I'm having an issue detecting low frequency signals with a high sampling rate. Shouldn't I be able to detect the frequencies as long as the sampling rate is at least 2 times the highest frequency I will measure? The frequency range I am measuring is 5-25 Hz, and I use Extract Single Tone.vi to measure the frequency. The sampling setting I am using is 2 samples at 10 kHz. Is there a method I can use to make this work?
Attached is the vi.
Attachments:
frequencytest.vi ‏21 KB

You are sampling at 10Ks/S, but only taking two samples. What do you expect to see? If your signals are binary (On or Off) you would only see either an on or an off, or if the rise/fall time was fast and you were Extremely lucky, one of each. If you want to see a waveform you have to sample for at least the period of a waveform. So you should take samples for at least 0.2 seconds to capture an entire waveform at 5Hz, ideally longer. Think of looking at a tide change at a dock. If you want to see the entire tide change you will probably have to measure repeatedly over 24 hours, not just run out on the dock, measure the height twice and leave. That wouldn't tell you anything other than at that precise moment the tide height was X, but not that it was at high tide, low tide, in between, etc.
I type too slowly, I see that a more technical answer has been given, so mine will be the philosophical one!
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How to sync. 50Hz sample rate with 5Hz external digital trigger signal?

Hello there,
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I hope the problem description is clear enough....
Thank you in advance for any advice.
Regards,
Oliver

You have to use the counter together with the analog input to achive this. You could configure the counter to generate 10 pulses with 50 Hz whenever the counter sees a rising edge on the inpput of the counter. The output of the counter would be used as the external clock for the analog input.

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Hi Neolker,
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Best regards,
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NI USB 6211 : sample rate and sample read

Bonjour,
je viens vers vous car je souhaite faire une acquisition continu d'un signal d'une entrée analogique. Sur cette entrée analogique est branchée un accéléromètre et un transducteur qui me renvoie donc une tension (120Mv/g) . J'ai donc réalisé à l'aide d'un exemple labview un programe. Cependant, celui-ci est très très long .. Lorsque je varie la tension d'entrée, cela prend plusieurs secondes avant d'afficher la valeur exact. Une erreur 200279 s'affiche lrosque les paramètres de sample read et sample rate ne correspondent pas... j'ai été voir sur le net sur ce lien :
http://digital.ni.com/public.nsf/allkb/AB7D4CA85967804586257380006F0E62
Mais rien n'y change. je vous laisse mon Vi en pièce jointe... cordialement
Attachments:
Pièces jointes :
Test 1.vi ‏28 KB

Doublon et résolu ici
Valentin
Certified TestStand Architect
Certified LabVIEW Developer
National Instruments France
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Du 2 au 23 octobre, partout en France

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Do you want it to be a graph or a chart? They are 2 fundamentally different things. A chart maintains a history of previous points and you can add to it a point or several points at a time. A graph has no history and it only shows what you send to it at a given point of time.
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Message Edited by Ravens Fan on 10-03-2008 11:52 AM

Problème NI usb 6211, sample rate et sample to read

Bonjour,
je viens vers vous car je souhaite faire une acquisition continu d'un signal d'une entrée analogique. Sur cette entrée analogique est branchée un accéléromètre et un transducteur qui me renvoie donc une tension (120Mv/g) . J'ai donc réalisé à l'aide d'un exemple labview un programe. Cependant, celui-ci est très très long .. Lorsque je varie la tension d'entrée, cela prend plusieurs secondes avant d'afficher la valeur exact. Une erreur 200279 s'affiche lrosque les paramètres de sample read et sample rate ne correspondent pas... j'ai été voir sur le net sur ce lien :
http://digital.ni.com/public.nsf/allkb/AB7D4CA85967804586257380006F0E62
Mais rien n'y change. je vous laisse mon Vi en pièce jointe... cordialement
Attachments:
Test 1.vi ‏28 KB

Bonjour Geoff54,
Tu es sur le forum international donc si tu veux que quelqu'un te reponde tu devrais poser ta question en anglais.
Sinon il éxiste un forum francophone : French Forums
L'erreur que tu rencontres vient du fait que le buffer du PC se remplit avec les données que tu acquiers mais tu ne le vide pas assez vite et lorsque le buffer est complet il te renvoie l'erreur -200279.
Pour solutionner celà, il faut que tu viennes vider le buffer plus rapidement.
Tu peux faire celà en augmentant le "sample to read" à plus que 1000 ou bien en laissant la valeur par défaut qui est -1.
Bonne journée,
Valentin
Certified TestStand Architect
Certified LabVIEW Developer
National Instruments France
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Travaux Pratiques d'initiation à LabVIEW et à la mesure
Du 2 au 23 octobre, partout en France

Minimum sample rate for 10KHz square wave with MIO

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Hi Steve,
Are you sure you are sampling at 400 and 800 KS/s?
I thought those boards only go up to about 100KS/s.
(show your point on your plot. Are you getting 40 samples per period?)
If you really are sampling that fast, then you have partially answered your own question. There is a small capacitance at teh input to the board. A high impeadance source will combine the input capacitance of the board to give you the charge ramp you mentioned. Only solution to this situation is ussing a low impeadance output buffer between your signal source and the AI.
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IIR Filtering and response .vi: Butterworth filter magnitude response depends on sampling rate -why?

Hi folks,
I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
This is fixed and prescribed for the application.
However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
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In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
THANK YOU for your ANSWERS!!!
chris
Solved!
Go to Solution.
Attachments:
butterworth-filter-differences.png ‏285 KB

Hello Lynn,
thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
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I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
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chris
Attachments:
IIR Filtering and Response_CH.vi ‏55 KB
butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000. Is this possible in Audition ? I have been going through all the preferences and all the tabs and I can't find this option. All I find is a convert option, or the adjust option. But that's not what I want. I want to mix down this way.
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JimMcMahon85 wrote:
Can someone explain this process in laymens terms:
http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
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just what's the easiest way to test if a simple dithering setting is
working for 32-bit down to 16-bit in Audition?... Why is there no info
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I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
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I guess the answer to this question relies on someone having an external digital amp/decoder/processor that can display the source sample rate and bit depth during playback, together with some suitable 'demo' files.
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Can't change sample rate for digital input on Mac Pro

Hello all,
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Hi,
when E&M signaling is configured on digital interface like the VWIC is, 2 or 4 wires operation is not applicable because there are no wires at all, and reported only for compatibility with the analog E&M card.
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Please rate post if it helps!

OLED display & signal sampling rate

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