FFT d'un signal

Bonjour
J'ai réalisé une application qui échantillonne des signaux à 5kHz (avec une carte NIUSB6210).
Je souhaite garder uniquement le signal à 50Hz de ces signaux (avec une FFT) et avoir la courbe à 50 Hz.
Est ce que ce traitement et possible?
Si oui comment?
Merci
Cordialement
Damien

Bonjour,
Merci d'avoir posté votre question sur le forum de national instruments.
Ce traitement est possible, il vous fait consulter l'aide labVIEW et aller dans recherche d'exemples. Il vous faut utiliser un passe bande en choississant le bonne ordre etc...
J'ai trouvé un VI sur la FFT. Vous pouvez le consulter et prendre exemple.
Brice S.
National Instruments France
Pièces jointes :
Frequency Analysis of a Filter Design - DFD.vi ‏32 KB

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Display FFT of input signal which receive from VISA read ?

Hi.
I receive data from MCU whcih sampling frequency is constant and we assumed it is S. now I get data from MCU with using VISA read block. I can display data, perform some functios like filtering, multiply and save them in txt files. all thing is OK.
but when I want to see FFT of incomig data on chart , it doesn't work !! I use this VI for calculate and display FFT of input signal (from VISA read) and display it which in Y-Axis is magnitude and X-Axis is frequency.
in your mind why this is not work and what I must done for show input signal FFT and frequency components ?
Thanks.

If I understood this correctly, you are getting the correct results when outputting to a text file but you are not able to see the results when trying to output to a chart?
This discussion may be of some help to you.
Rob S
Applications Engineer
National Instruments

FFT pour un signal analogique en temps réel

Bonjour,
Je souhaite calculer la FFT d'un signal sortant d'un générateur de fonctions?
Comment je dois le faire sur Labview? J'arrive pas à trouver les bonnes fonctions qui transforment ce signal qui est en temps réel en un signal manipulable par la fonction FFT disponible. sur la palette de fonctions de Labview.
Merci d'avance pour toute remarque, aide ou suggestion.

Greetings Sir,
You can find many examples of the correct use FFT functions inside the LabVIEW Example files. The FFT functions come in two version,the 'default' and the 'point by point'
The links above show the wiring options of each of these functions.
Be aware that these vi's do not come with the LabVIEW Base version, they are available in the LabVIEW Full and onwards.
Kind Regards,
Wouter
Applications Engineering
National Instruments Belgium

How to do FFT of vibrations signal?

Hello,
I'm trying to measure the time and frequency spectrum of a vibrations signal.
My time plot seems to be ok, but my FFT graph doesn't look right.
An expected vibration FFT graph of a bearing should look like peaks at various frequencies, but my FFT graph looks only like a linear line.
Does anyone know what's wrong with my attached program?
Thank you.
Y.Yang
Attachments:
3 axis white and green sensors v2.vi ‏122 KB

Your code looks fine. I double checked how you're acquiring data, and it seems fine. I'm pretty sure that you are acquiring an FFT and the graph is just not scaled correctly. Do the following:
Right Click Waveform Graph 2 >> X Scale >> AutoScale X
Right Click Waveform Graph 2 >> Y Scale >> AutoScale Y
I've also attached an example of what I've done to get the FFT from a simulated signal.
Che T.
Applications Engineer
National Instruments
Attachments:
FFT.vi ‏67 KB
fft.png ‏43 KB

FFT avec un signal d'entrée variable

hello to all
who has an idea how I can change the input FFT Example (windowed FFT) which in underneath the tree Labview 2011
see Figure telque entry that I would change is cerculer black
Pièces jointes :
pa.jpg ‏346 KB

Hello basmasag,
I'm not sure what you want to do, but you can modify this part of the diagram with some E/S from modules installed on your target for example.
You can put instead of 'sine wave generator', 'square wave generator' for example too.
I hope it's help you.
Regards,
Jérôme Henrion | NIF - Application Engineer | PXI Captain
Certified LabVIEW Developer

Apply FFT on recording signal

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since the ios 4 we can take advantage of the accelerate framework which holds the vDSP API and this has a FFT in it. I want to use that for the microphone input and filter low tones with a low pass filter. can somebody explain me how to apply the FFT to my microphone input and then deal with the data from the FFT or does somebody has a link for me explaining that?
would really appreciate that, thx.
palmdale

I dont think I more info about this. Try googling yor way around or try looking into the free books apple has provided for developers. You might find something in these books.
http://www.tuaw.com/2010/12/19/apple-posts-free-developer-documentation-for-iboo ks-users/

Perform FFT on stationary signal

so once i capture all my data (mains 50Hz) and i want to perform an FFT on it, how do i then do it while its stationary. ive almost got some code finished, however, its still going to be streaming isnt it?. i need to make stationary. any advice?
cheers.

You can accumulate data in a buffer until you reach the amount of data you want to analyze. Then clear the buffer and start over. Or you can use the Point-by-Point VIs which will do the accumulation for you. If you open the PtbyPt VIs you will find that they build an internal buffer.
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Frequency Response VS FFT for measring frequency response of a audio ouput signal.

We have purchased the Sound and Vibration Toolkit and I have some questions.
From the frequency response example I looked at you measure the input then the ouput and the Vi gives you the difference.
I want to measure the audio frequency response a radio.
So all I have is the output I dont have the audio input to use as a reference.
How would I measure frequency response with the frequency rasponse VI with only the output audio signal?
I was also looking at the FFT example to measure the audio signal frequency response.
But from what I can tell so far this only does 1 channel I want to do both channles at the same time.
Is there a way to do FFT on 2 channels at once and have them output on the same graph?
Thanks for any help you can provide.

Hi,
I was looking through the examples and If you look at the SVXMPL_Multichannel FFT (simulated) examples, this shows how to take an FFT with multiple signals. You just pass in an array of the waveforms. If you need to put your signals together you can use the merge signals VI, which is located in the Express»SignalManipulation Pallette. Let me know if you need any further help with this issue.
Have a great day,
Michael D
Applications Engineering
National Instruments

Phasenabweichung & Frequenzbestimmung von 2 Signale

Mein System wird 2 Signale aufnehmen: die Kraft(N) und die Geschwindigkeit(m/s) für unterschiedliche Frequenzen, die von DIAdem 9.0 eingestellt werden, zwischen 5 und 150 Hz. Für jede Frequenz werden Messungen und Aufnahmen durchgeführt. Die signale sind annährungsweise als SINUS anzusehen. - 1) Wie kann ich die FREQUENZ der beide Signal messen, um die Einstellung der Steuerung zu kontrollieren ? Müssen für jede Messung (z.B. für jedes 1 Hz) eine FFT durchgeführt und gespeichert werden, und wie ? - 2) Wie kann ich die Phasenverschiebung zwischen Kraft und Geschwindigkeit bestimmen und auswerten ? - 3) Wie kann ich die Messung der unterschiedliche Phasenverschiebungen für die 145 Messungen (150-5 Hz) Automatisieren ? - 4) Wäre eine stetige (kontinuirlische) Messungen der Phasenverschiebung in Abhängigkeit der Frequenz möglich ? Vielen Dank, und MFG aus Hessen. Der Jungingenieur aus Frankreich.
Jean-Pierre, Junior Engineer Germany

Wenn ich Sie richtig verstanden habe, so lesen Sie zwei analoge Spannungen mit einer bestimmen Abtastrate (welcher ?) über Ihre Datenerfassungkarte ein.
"Die Signale sind annährungsweise als SINUS anzusehen. - 1) Wie kann ich die FREQUENZ der beiden Signal messen, um die Einstellung der Steuerung zu kontrollieren ? Müssen für jede Messung (z.B. für jedes 1 Hz) eine FFT durchgeführt und gespeichert werden, und wie ?"
-> Wenn Sie 100%ig wissen, dass es sich bei Ihrem Signal um ein Sinus handelt, so können Sie einerseits eine FFT (in Diadem -> Analysis -> Signalanalyse -> FFT) über Ihr Signal laufen lassen, oder Sie bestimmen programmatisch den Nulldurchgang der Sinusfunktion. Für die letzte Möglichkeit wäre es von Vorteil, dass das vorgegebene Signal bestimmter Frequenz für mehrere Perioden vorliegt. Sie können durch den zeitlichen Zeitabstand der Nulldurchgänge so einen Rückschluss auf Ihre Frequenz ziehen. Beachten Sie bitte, da es sich um ein Sinus Signal handelt, eventuell eine Richtungserkennung (Nulldurchgang von - auf + oder von + auf -) zu implementieren oder einfach die berechnete Frequenz zwischen zwei Nulldurchgängen ohne Richtungserkennung zu halbieren.
"2) Wie kann ich die Phasenverschiebung zwischen Kraft und Geschwindigkeit bestimmen und auswerten ? - 3) Wie kann ich die Messung der unterschiedliche Phasenverschiebungen für die 145 Messungen (150-5 Hz) Automatisieren ? - 4) Wäre eine stetige (kontinuirlische) Messungen der Phasenverschiebung in Abhängigkeit der Frequenz möglich ?"
-> Sie können die Phasenverschiebung zwischen zwei Signalen z.B. mit einem Referenzpunkt bestimmen. Wenn Sie beispielsweise die Phasenverschiebung zwischen zwei Signalen mit einem Oszilloskop bestimmen möchten, so setzen Sie je einen Cursor z.B. bei den Nulldurchgängen (von - nach +) auf den entsprechenden Sinus und können dadurch, durch den zeitlichen Versatz auf die Phasenverschiebung Rückschluss ziehen. Schwierig wird es, wenn Sie einen Phasensprung zwischen Ihren beiden Signalen bekommen (d.h. ein Vielfaches von 360°). Diesen Phasensprung müssen Sie programmatisch bestimmen.
Leider weiss ich nicht genau, welche Hardware Sie in welcher Diadem Version einsetzen und auch nicht welche Signalpegel Sie bei Ihren analog Inputs erwarten ? mV ? V ?
Eine weitere Möglichkeit, sich die Phasenverschiebung zwischen Referenz- und Eingangssignal berechnen zu lassen, ist der Einsatz eines Lock in Verstärkers. Allerdings wird dieser meistens bei kleinen Eingangsspannungen eingesetzt (nv...mV). Die vom Gerät ermittelte Phasenverschiebung zwischen den beiden Signalen kann dann beispielsweise über die serielle oder die GPIB Schnittstelle abgefragt werden. Diese Alternative ist mit weiteren Kosten verbunden.
Um Ihre Abläufe automatisiert ablaufen lassen zu können, können Sie sich ein DAQ Schaltplan in Diadem erstellen, , dass die Ausgabefrequenz mit entsprechender Amplitude steuert und auswertet.
Mit freundlichen Grüßen

Simple FFT graph problem

Hi to all,
The problem is: I have a 100Hz sine in a text file (sampled at
800Hz), and when try to see the FFT of that signal, the dominant
frenquency should be around 100Hz, but doesn't happen. I think that's
because the "Convert to Dynamic Data" VI, but how can I solve this? The
VI and the signal file goes in attachment.
Thanks in advance.
Ricardo.
Attachments:
analise_sinal.vi ‏81 KB
sine_100hz.txt ‏685 KB

Before converting the array into the Dynamic Data type, convert it into a waveform type. Feed the data into the Y element of the waveform cluster and enter the inverse of the sampling rate into the dt element of the waveform cluster. Feed the waveform data into they Covert to Dynamic Data and change the properties of it to accept a single 1D waveform. Your frequency info will then appear in the FFT.
Looking at your data, if this sine wave is at 100 Hz then it is not sampled at 800 Hz. It is sampled at a higher rate.
Randall Pursley
Attachments:
Data Convert.PNG ‏14 KB

How to measure/estimate Bias of the signal

Dear Sir
I am performing the FFT of the signal which collect in real time from the hall effect current tranceducer SCT-013-005. I need to measure/estimate bias of this signal. Can you please guide me how can I do that. (Attached is my vi which is developed in LabVIEW 2012)
I shall be thankful to you for your attention and consideration.
Kind Regards
Urfee
Solved!
Go to Solution.
Attachments:
How to measure Biase of the signal.vi ‏930 KB

Hi tronoh,
what about using the "Basic Averaged DC&RMS" function on your periodic signal?
Best regards,
GerdW
CLAD, using 2009SP1 + LV2011SP1 + LV2014SP1 on WinXP+Win7+cRIO
Kudos are welcome

What is the output on the FFT PSD function?

I am trying to evaluate the FFT PSD function relative to frequency. I expect f(0) = the variance of all signals, but that is not what is given. Does anyone have any information about this. I ran a test array with a 1 Hz sin wave, with 10 samples/second, 120 total data points--the FFT PSD gives f(0)=0.

Hi,
The LabVIEW help file gives the following information concerning this function:
The FFT Power Spectral Density VI completes the following steps to compute power spectral density.
Computes the FFT of time signal.
Forms the power spectral density of time signal.
Averages the current power spectral density with the power spectral densities computed by the VI since the last time the averaging process was restarted.
Returns the averaged power spectral density in power spectral density
The single-channel version of this VI can perform single-channel measurements in both one-shot mode, meaning a single call, and continuous mode, meaning multiple calls with history. The single-channel version can perform multichannel measurements only in one-shot mode. If you want to make multichannel measurements in continuous mode, use the multichannel version of this VI.
The single-channel version of this VI maintains internal state information for a single channel only. Calling the single-channel version to process another channel without using the restart averaging control to clear the history results in an unexpected behavior of this VI. The unexpected behavior results from the VI passing the internal state information from one channel to another.
Note The single-channel version of this VI is primarily intended for continuous processing of a single channel. Do not generalize this behavior to the multichannel case and use the single-channel version in a For Loop to continuously process multiple channels by indexing an array of waveforms. Refer to the examples\measure\maxmpl.llb and the examples\analysis\measxmpl.llb for examples of how to use a multichannel version of a VI like the FFT Power Spectral Density VI in a loop.
Refer to the Noise Waveforms and PS Density VI in the labview\examples\measure\maxmpl.llb for an example of using the FFT Power Spectral Density VI.
Amanda Howard
Americas Services and Support Recruiting Manager
National Instruments

Problems with obtaining actual results using the FFT VI.

I'm using the FFT express VI and graphing the results. My experiment involves decreasing the amplitude of a 100Hz acoustic pressure wave, ideally to zero, by phasing it with another signal of the same frequency and amplitude. According to other analysis I believe that the amplitude from the FFT of the signal is increasing and then decreasing. However, I am not able to visually see this in real-time on the graphical display.
The change in amplitude is happening quickly, every few ms. Is this timing to fast for the process to maintain proper resolution? Is there a way to make the output from the FFT more sensitive?
Thanks for the help.

The period of a 100 Hz signal is 10 ms. If your amplitude is changing every few ms, an FFT is not going to reflect this accurately. The FFT is basically defined for constant amplitude during the segment of signal being analyzed. The FFT also typically requires several cycles to give useful results. If it changes during this time, other techniques must be used to detect that. There is a large body of literature on time-frequency analysis dealing with this topic.
Lynn

Calculating FFT using the FFTAverageType variable in DIadem

Hi,
I have some doubts about how should I use the averagingType variable. I want to define this variable as "Arithmetic" and get the combined FFT of two trials of the same process, i.e I have two signals which represent the same process.
What I do is just calculate the FFT of each signal separately. Then I calculate the amplitude of each FFT signal and finally i combine these two signals by doing the mean for each amplitude. I.E if i have signal1FFT = (1 , 2 ......23 ) and signal2FFT = (2,4, ...43 ) the resulting signal should be ( 3/2 , 6/2 ...... 66/2 )
Am I correct about what this averaging type variable does? When i try to do the same in other mathematic programs I get different results but I don't know exactly what I am doing wrong,
Thank you so much for the help!

There are several ways to approach averaging in the FFT environment.
FFTAverageType is should be "Arithmetic" which means that all FFTs are weighted equal.
The Exponential averaging is kind of a running average for special calculations which are typically done during a running measurement.
Much more important is the parameter in "Calculate average from:" (Amplitude or Autospectrum)
Amplitude is giving a linear result and Autospectrum is giving the result based on squared values. The squared result is the correct result from an energetic point of view.
Example:
If you have a sine with a peak of 2 for one second and a peak of 4 for another second, you might want to get and averaged peak of 3.This might be wrong.
The RMS value of a sine with a peak of 3 would be 2.121.
The RMS which is representing the energy in a signal is the root of the average of the squared values. The RMS of a signal with half the time 2 and half the time 4 is 2.236. This means, that a sine with the peak of 3.162 for two seconds has the same RMS as the signal with a peak of 2 and 4 for one second each.
If you average FFTs in DIAdem you need to decide between the linear average (based on the peak Amplitude) and the energetic correct squared average (based on the Autospectrum). Right or wrong depends on your circumstances.
Usually, other mathematic programs are giving you the choice to decide or they might just use one or the other method.

Extraire une fréquence précise d'une FFT

Bonjours,
Voila ma question, J'ai effectué une FFT sur un signal temporel échantillonné à 1024 S/s , j'ai donc tout mon spectre de 0 à 500Hz, maintenant j'aimerais en sortir l'amplitude à 50Hz +-3Hz (ceci afin de voir la contribution du réseau électrique dans le spectre de mon signal mesuré) et je ne sais pas vraiment comment m'y prendre, iest-ce qu'il y a un vi déja existant qui fait cela?
J'ai essayé d'utilisé le vi "Multiple Tone Exctract", cela marche mais je le trouve un peu lourd pour mon besoin.
Cordialement
Sébastien Kuenlin
Résolu !
Accéder à la solution.

Bonjour et merci d'avoir posé votre question sur le forum NI.
Effectivement un VI Express tel le "Multiple tone extract" est gourmand, c'est pourquoi il est possible d'utiliser la fonction "Extract Single Tone Information", de plus bas niveau, comme dans l'exemple proposé ci-dessous et en pièce jointe.
En espérant avoir répondu à votre question.
Mathieu TUR
National Instruments France
Mathieu_T
Certified LabVIEW Developer
Certified TestStand Developer
National Instruments France
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Pièces jointes :
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