Svt frequency response (mag-phase).vi

Hello people
I want to find out the peaks from the specrum which get from magnitude output of svt frequency response (mag-phase).vi.
but I always get errors when I use the svl spectrum peak search.vi, it looks like the peak search.vi can not accept spectrum type such as frequency response spectrum.
what can I do to find out the peaks@freqency from the magnitude output of svt frequency response (mag-phase).vi
Thanks in advance
Tim

Dear Lisa,
Are you still here for supporting.
I have installed Sound and Vibration Suite 2011 (SV) to use with LabVIEW Professional 2009 SP1. I have a concerns.
After I install SV to the computer and launch my VI in LabVIEW 2009, I could not find the SV menus to use (in block diagram windows). I found a link
http://digital.ni.com/public.nsf/allkb/9A4BC69D802AD4D9862574FA004F0C20
for the same issue but I could not find the path as they mentioned. The addons SV still missing in the installed directory.
by the way, I can follow the path to find out VIs that I currently need to use. However, some VIs have different name. For example
I could not find "SVFA Power Spectrum.vi" but "SVT Power Spectrum.vi"; "SVFA Magnitude and Phase to Real and Imaginary.vi" but "SVT Magnitude and Phase to Real and Imaginary.vi"
Similarly, I could not find "SVT Frequency response (Mag-Phase).vi" but "SVFA Frequency response (Mag-Phase).vi"
Do they have the same function? It makes me confused.
and I could not find "SVT FFT Spectrum (Mag-Phase).vi"
Can you please explain me what should be the mistake while I installed SV 2011 (I can not find SV 2009 and for interation with UFF58 file, LabVIEW 2011 is required)
Thank you so much.
Thinh Vo

Similar Messages

Frequency response and phase difference

Hi,
I generate "Sine-Swept" with a function generator, which is the input of my device.
Then, I measure the output(vibration) via a photodiod (which is as voltage).
Now, I want to obtain FRF and Phase difference between these two wave (output and input).
In most of the examples, the input and swept has been generated by the software itself, but if the input has been acquired externally and also the swept has been done via a function generator, how would be the FRF?
I have installed "Sound and vibration toolkit", Could you please give me a heads up, that how can I do that?
In advance, I really appreciate your help,
Petar

Hi,
I attached the project which I've recorded the data from DUTs. Also, I attached 2 snap-shots of the project, in case of the project file couldn't be launched.
I have connected the signal from a function generator to "ACH0" channel, and performed the swept-Freq in the output signal of function generator. (Red graph in 1st chart). Please note that the swept sine wave is generated via the function generator itself...NOT VI
The generated signal by the function generator is the input, which derives the oscillation of a drop,
A photodiode measures the oscillation of the drop via laser-scattering by drop vibration. SO....the signal generated by the photodiode is the "Response"....(white wave in 1st chart)(connected to "Ach1" channel.
Now, I want to obtain FRF, which the input is the signal generated by the function generator, and Response is the output of the photodiod'
As it's seen in the files...the FRF and phase difference is n't what is expected, also the coherence between in and output is zero!
Please help me out with this problem...
Thanks
Petar
Attachments:
12.seproj ‏419 KB
1.JPG ‏183 KB
2.JPG ‏177 KB

Where can I find the amplitude and phase frequency response when the AC coupling is set on the 4472

My friends,
I need to know the amplitude atenuation but most important, the phase distortion introduced on the low frequency components of the signals when the AC coupling is setted (i.e. when the high pass filter is connected) on the 4472 and 4472B DAQBoard.
I can construct the amplitude frequency response by generating and aquiring a sine waveform of a knew amplitude. But I cannot construct the phase distortion introduced by the circuitry.
However, I assume that this crucial information should be available in the DAQ Manual or in the website of NI, but untill now I haven´t found it.
Thankning in advance,
crimolvic from Chile

crimolvic,
Here are the Specifications and Datasheet for the 4472. They indicate a phase non-linearity of less than 0.5 degrees across all frequencies.
For information on how this varies with frequency, see the attached
spreadsheet. This response was the result of testing on a single
4472. Although this is classified as a "typical" response, it is
not gauranteed.
Have a great, day!
Travis
Attachments:
4472 Phase Linearity.xls ‏21 KB

3 Simultaneous Frequency responses using PCI-4552

I need to obtain 3 simultaneous frequency responses (Magnitude and Phase) using a PCI-4552 board. Ch 1 is the reference for the 3 FR's. I succeded in obtaining independent FR's, but I cannot configure the vi with the 3 FR's together. I attach my best try, please take a look at it.
Attachments:
zoom4.vi ‏378 KB

I took a look at your VI. It looks like you have two issues.
First, you are using a single stop button's input to stop several loops. The problem is that the output of the button will not allow the consecutive while loops to execute until the first one has finished (they are waiting for the prior loop to finish to get the final value of the button). If you want to read the value of a single button simultaneously in several while loops, you need to use local/global variables.
You should have an example:
Stopping Parallel While Loops with Reset.vi
... included with LabVIEW which will demonstrate how to properly do this.
Your second issue is that the base analyzer can only do two simultaneous channels. If you really need to do more, you'll need to grab al
l the channel's raw data using the DAQ API instead of the DSA API, and feed the individual captures into the "Frequency Response Function (Mag-Phase)" vi. This should give you the same results, but you'll be using the host computer to do the analysis instead of the board.

Frequency response function modal analysis

After reviewing the signal analysis functions in DIAdem I have realized them to be a bit limited for modal analysis. I have a couple hammer impact tests that I need to process a frequency response function for, and since this is brand new to me I'm not seeing anything in the embedded function list that is going to help me. I was wondering if anyone out there has a couple of pointers on generating a FRF plot for modal hammer impact tests. I did notice that the ChnFFT2 command allows me to generate a transfer function, coherance, and FFT Cross Spectrum channels for analysis. Though I might be confused and this may be everything I need. My FFT2 settings are below.
[code]
FFTIndexChn      = 0
FFTIntervUser    = "NumberStartOverl"
FFTIntervPara(1) = 1
FFTIntervPara(2) = 2500
FFTIntervPara(3) = 1
FFTIntervOverl   = 0
FFTNoV           = 0
FFTWndFct        = "Rectangle"
FFTWndPara       = 10
FFTWndChn        = "[1]/Time axis"
FFTWndCorrectTyp = "No"
FFTAverageType   = "No"
FFTAmplFirst     = "Amplitude"
FFTAmpl          = 1
FFTAmplType      = "PSD"
FFTCrossSpectr   = 1
FFTCoherence     = 1
FFTTransFctType = "Spectrum H0"
FFTCrossPhase    = 0
FFTTransPhase    = 0
Call ChnFFT2("[1]/Time axis","'[1]/H_1' - '[1]/H_4'","'[1]/A_1' - '[1]/A_4'") '... XW,ChnNoStr1,ChnNoStr2
[/code]

Standard modal analysis has something denoted as FRF. I have a labview application note "The Fundamentals of FFT-Based Signal Analysis..."
Frequency Response Function
The frequency response function (FRF) gives the gain and phase versus frequency of a network and is typically computed as
where A is the stimulus signal and B is the response signal.
The frequency response function is in two-sided complex form. To convert to the frequency response gain (magnitude) and the frequency response phase, use the Rectangular-To-Polar conversion function. To convert to single-sided form, simply discard the second half of the array.
You may want to take several frequency response function readings and then average them. To do so, average the cross power spectrum, SAB(f), by summing it in the complex form then dividing by the number of averages, before converting it to magnitude and phase, and so forth. The power spectrum, SAA(f), is already in real form and is averaged normally.
Refer to the Frequency Response and Network Analysis topic in the LabVIEW Help (linked below) for the most updated information about the frequency response function.
http://zone.ni.com/devzone/cda/tut/p/id/4278
So the options for FFT2 are
No
DIAdem does not calculate a transfer frequency response.
Spectrum H0
DIAdem calculates the transfer frequency response by dividing the FFT of the output signal (A) by the input signal (E): FFT(A)/FFT(E). DIAdem averages the amplitudes of the individual transfer functions.
Spectrum H1
DIAdem specifies the cross spectrum and the auto spectrum for each signal pair. DIAdem calculates the transfer frequency response by dividing the averaged spectra: Middle(cross(A,E))/middle(auto(E)). DIAdem does not average phases, because phases can delete each other.
Spectrum H2
DIAdem specifies the cross spectrum and the auto spectrum for each signal pair. DIAdem calculates the transfer frequency response by dividing the averaged spectra: Middle(auto(A))/middle(cross(E,A))
If you assign the values Spectrum H1 or Spectrum H2 to the variable FFTTransFctType, DIAdem averages and divides the cross spectra and the auto spectra and calculates the amplitudes last.
Which state auto spectrum when FRF is power spectrum.

Frequency response - sound and vibration

Hi,
I need to find the frequency response of the DUT. I am using the NI example from sound and vibration toolkit to do
so (LabVIEW 8.5\examples\Sound and Vibration\Getting Started\SVXMPL_Getting Started with Swept Sine (DAQmx).vi) now my problem is
to tabulate the phase difference between the stimulus and response when i do it i get constant values. Even though i dont give any response
signal to the channel i get the wave same as stimulus, is it suppose to be like that !!!!!!!!!! I will attach my vi so that it gives the idea where
i am measuring the phase difference pls check it and help me with this.....thank you in advance.
Attachments:
DAQ Freq resp (req)_trial.vi ‏121 KB

What is the DAQ card you are using for executing this Frequency Response?
This DAQ error is related to the output frequency of your signal higher than the possible output rate of the card. Basically, you are trying to update at a rate higher than the capability of the card.
For doing Frequency response, generally you need to have a NI DSA card (446x/447x/449x) with one Analog output and two analog inputs (minimum). Generate the swept sine signal from the Analog output channel, give this signal as input to your DUT and also to one of the Analog input channel. This input to your Analog input channel will act as your Reference signal. Then the response signal from your DUT, connect it to another Analog input channel. You would get a very good response results.
The reason why you need to have a DSA board is that, for doing Frequency response, we need to acquire both high frequency and low frequency components without much loss. This is possible only if your DAQ board has a higher dB value (in the range of >110dB) which is present only in DSA boards.
I have completed a Frequency Response Analysis just a week back with the same example programs. So there wont be any problem with that vi.
Regards,
Sundar Ganesh

Frequency Response Function NxM output format

Hi,
When the Frequency Response Function for magnitude, phase, and coherence
is used in the NxM "all cross pairs" polymorphism, the format of the outputs
are still 1-D arrays of clusters instead of a 2-D NxM sized array of clusters.
Does anyone know how the indexing of this 1-D array would correspond to
indexing of the would be 2-D array? Or is it just [index/M, index%M].
Thanks,
Kevin

Hi Kevin,
First, thanks for posting this request. You've helped us find a gap in our context help. The documentation on X-Y pairing should be improved and the VI documentation should be extended to cover all instances of this polymorphic VI. Here is some text that should be added to the "X-Y pairing" description:
"all cross pairs" means that the pairs are formed by matching each signal from array X with ALL signals from array Y, continuing in this fashion for each signal in X. If there are N signals in X and M signals in Y, the total number of cross pairs is N x M. The paired signal results Xchan-Ychan are returned in the order 1-1, ... 1-M, 2-1, ... 2-M, ... N-M. "
Hope this helps,
Mike C
LabVIEW Math & Signal Processing
National Instruments

Calculate frequency response using FFT and inverse FFT

Hi,
Attached is the program using FFT and inverse FFT to filter a time domain signal. The frequency response of the LPF can be obtained by using the chirp signal from 0 to 5kHz. However, I don't know why the signal obtained from a sine wave input is so strange. The amplitude is wrong and has a envelope outside. Please help to point out what's wrong with that.
Bill
Attachments:
fft filter.vi ‏87 KB

If you check the help text for sine wave.vi you'll see that it generates the sine wave based on the following formula:
yi = a*sin(phase[i])
for i = 0, 1, 2, …, n – 1 and where
a is amplitude,
phase[i] = initial_phase + f*360*i
This means that when you input a=1, f=0,1 and initial_phase=0 you will get a sine wave that is based on samples at every n*36 degrees; i.e. at 0, 36, 72 etc...due to this sample rate you never see the full amplitude (+/- 90 degrees), the wave is clipped at the top. If you input an initial phase of 64 degrees you will get the full amplitude, but the wave is still deformed due to digitalization...
The lower the frequency you put in, the closer the digitalized representation will be to the true sine.
Use the Waveform Generator VIs from the analyze palette if you want to have more control over the wave generation (sample rates etc.). (Not available if you have the base package.)
MTO

Make frequency response analyser using frequency generator and counter

Hello
Can we make a frequency response analyser using a Frequency generator and frequency counter?
How to add modulation with it? Modulation frequency is to be varied as per the input to given to the carrier!
The outputs are Frequency, magnitude, and the phase as like solartron FRA.
somebody have an Idea for this
awaiting for the solution
thank you
"Thanks with regards "
by
..........Gireesh..........

Hello Gireesh,
You can use a function generator to generate frequencies and use the modulation tollkit and other tools availabe with LabVIEW to do the modulation part . Or you can use the analog output ports on the daq card to generate different frequency signals for the same purpose .This should pretty much serve your purpose.
http://zone.ni.com/devzone/cda/epd/p/id/5646
http://digital.ni.com/manuals.nsf/websearch/AF3615F31CE9656C862576070020B8F7

Frequency response and transfer function

Hi
What is the difference between transfer function vi in basic labiview frequency palette and FRF(Mag & Phase) vi in sound and vibration tool kit. Aren't the two vi's for the same purpose, but I am getting different results using these two vi's.
Which vi should be used for calculating transfer function between two signals x and y.
Ankit

Hi Ankit,
You are correct that the FRF (Mag & Phase) VIs are very similar. The one provided with the Sound and Vibration Toolset has several extra features such as the Spectrum Info output.
For a given measurement with given settings you could choose either VI, however you should see the same result from both. If you are not seeing the same result, I expect the input parameters are not identical. Changining input parameters such as averaging can have significant impact on your results. Verify that you have identical parameters and I expect you will be able to get your results to match.
Regards,
Jennifer O.
Applications Engineer
National Instruments

Questions about the frequency response step

Hi,
I am using Signal Express 3.0. I am not clear about the transfer functions in the frequency response step with different averaging modes. What I got from the help file is that H(f)=Sab(f)/Saa(f) which is cross spectrum over auto spectrum where a is the stimulus signal and b is the response signal. When RMS averaging is used, I am wondering whether the transfer function becomes magnitude of cross spectrum divided by magnitude of auto spectrum. When vector averaging is used, everthing is in complex numbers.Is it doing averaging of the time-domain signals, frequency domain signals or the results from transfer functions?
Thanks a lot.
Ningyu Zhao
Solved!
Go to Solution.

The algorithm to calculate the spectrum is the same in both modes. However, the averaging mode can have a huge effect on the result. RMS averaging is done on the spectrum itself, after the calculation. Vector averaging is done on the input signals before the calculation. With vector averaging, the signals must be coherent (have the same phase) or the result will be wrong due to the signal being averaged away.
This account is no longer active. Contact ShadesOfGray for current posts and information.

MacBook core duo Sept. 2006 - the audio is a mystery. Any analogue out has bass boost with bass distortion. With digital out, by USB or Airport Express Airtunes, the frequency response is normal. Somewhere, Apple put in an analogue bass boost.. why?!

Since new, my Macbook core duo has sent all audio to all analogue outputs with frequency distortion. Some physical hardware or firmware in the analogue section adds a bass boost to the frequency response of the audio.... And, this boost adds bass frequency distortion to most, if not all, of the audio analogue output.
This happens for all audio sources, players, iTunes, videos, streaming audio/video, movies... any sound source available to the Macbook. No-one with whom I have talked about this problem, has ever heard of it. Not elegant ideas for reducing it are to use equalisers on players and iTunes. iTunes even has a preprogrammed "bass reduce" on its equaliser, as if it knows already that this inherent bass boost "feature" has been built into the Macbook, and cannot be defeated by the user.
The digital audio is of sufficiently high quality to play well on a very good to excellent sound-system; so, it's a shame someone has mucked around with the frequency response of the analogue conversion, by designing the frequency distortion right into the computer. This motherboard, (which includes the sound-section), has been replaced, along with the speakers, and everything sounds exactly the same as it was before.
To receive a flat frequency response and no frequency distortion, I listen on one receiver using Airtunes, (it doesn't matter whether or not the analogue or digital output jack is wired analogue or connected via optical cable.... the freq. response is flat); and I listen on a high-end stereo system using an USB output port to an A/D converter, passing the analogue result using long patch-cords connected to the "line-in" jacks of the stereo. The bummer is that most of my audio sources are not derived from iTunes... hence, I cannot use Airtunes with Airport express for them. I've tried using "Airfoil" without luck. The sound becomes distorted with sibilance and other frequency anomalies.
Question: has anyone else discovered this sound imperfection in any Mac product? And, does anyone know if Apple is aware of it? Finally, has anyone found an Apple fix for the problem; or, at least, come up with better solutions than I have?
Thanks heaps for any impute and answers you may supply!! junadowns

Army wrote:
This might not help you a lot, but if you want a stable system, try using packages from the repo wherever possible, look at the news before you update your system and don't mess things up (like bad configuration etc.).
When it comes to performance, you won't gain much by compiling linux by yourself! Just use the linux package from [core] or if you want a bit more performance, install the ck-kernel from
[repo-ck]
Server = http://repo-ck.com/$arch
(this has to go to the bottom of /etc/pacman.conf)
(use that one which is best for your cpu (in your case this might be the package linux-ck-corex).
Hmmm, Linux-ck-corex doesn't even load.. I am now trying to install the generic one. Hope it works.
Edit: I will first try linux-lqx...
Last edited by exapplegeek (2012-06-26 18:33:31)

Frequency response requirements for headphones with CMSS on XFi ???

Hi,
I would like to know if someone could tell me what kind of heaphones are suitable for the CMSS mode with the XFi.
I mean between : flat response/free-field correction/diffuse-field correction.
Applying HRTF filtering should mean that headphones with flat response is the best option ( same configuration as binaural recordings).
But I have a big doubt that Creative team expects costumers to possess such a pair of headphones, as it is rather for scientific uses (psychoacoustics, audiology etc...).
So, if we look at the technical solutions for wide audience we have two options (FF correction and DF correction). Here is a trick because these corrections intend to reproduce some of the effects from HRTF (for two different environment configuration of HRTF measurements). It is why the frequency response of most of the headphones have a notch between 4Hz and 0 kHz.
To simplify, if we listen binaural sounds with classical headphones the effect of outer pinna is reproduced twice.
So I guess Creative have implemented a kind of normalization/equalization/correction process to deal with the non-flat frequency response of headphones, but do someone know if they have chosen diffuse field or free field correction ?
This post might seem a detail but the issue can be very important for the accurate localisation and the coloration? of 3D sounds with headphones.
Thank you, and please forgive my english!

The only possibility that I can think of is that 2/2. mode is NOT as simple as headphone mode with crosstalk cancelation. Perhaps the HRTF only kicks in for sound sources outside of the arc directly in front of the listener. If that were the case, you wouldn't percei've any distortion for sound sources in front of you.
Also, you are wrong regarding DirectSound3D. Keep in mind that Direct3D and DirectSound3D are not the same. The whole point of OpenAL and DirectSound3D is that they present an API to the programmer through which there is NO specification of the number of speakers. When using OpenAL or DirectSound3D, the only thing a programmer can do is specify the location of a mono sound source in 3D space relati've to the listener. The speaker settings for your DirectSound3D or OpenAL device will then determine how this sound is "rendered" by the soundcard. It is not under control of the game. For example, if you have 5. speakers and the 3D position is behind you, the SOUND CARD will make the decision to use the rear speakers. If you use headphones, the SOUND CARD will decide to apply an HRTF to create the illusion of a rear sound source. The point is that the game does not have control over how many speakers you will get sound from.
However, to further complicate the situation, there are SOME games (HL2 is an example) where DirectSound3D is used, BUT the sound output of the game itself IS a function of the Windows speaker settings. This is not how programmers are SUPPOSED to use DirectSound3D. I've written about this countless times. There is a good post on [H]ard|Forum about this. Do an "advanced" search with my username (thomase) looking for the terms "hl2" and "cmss".

How can i create an indicator to read the RMS value of my FFT using FFT Spectrum (Mag-Phase).vi

Hai to all,
how can i create an indicator to read the RMS value of my FFT using FFT Spectrum (Mag-Phase).vi.
as u can see in my block diagram(attached), i can use statistic to read the RMS value for the data but i cant's use it on the FFT Spectrum (mag-Phase).vi .
Thank you for helping.
Attachments:
block diagram.jpg ‏48 KB

hafizzuddin wrote:
thank you for the opinions, for now i am not using any express Vi..
In the long run you will benefit from this. Anyway is your problem solved, or do you need more help?
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

What is the frequency response for the new iPhone 5 earbuds?

I have recently purchased a new iPhone 5 with iOS 6.0.1. I was wandering about the specifications of the new earbuds. What is the frequency response? Thank you for the information.

Apple has released 2 sets of earbuds with the iPhone 5 release. There are the earpods and also the in-ear headphones. Here are the in-ear headphones: http://store.apple.com/us/product/MA850G/B/apple-in-ear-headphones-with-remote-a nd-mic?fnode=49. The frequency response on these is 5kHz-21kHz. I was unable to find the earpods response for definite but the best response I got was 30Hz-16kHz. Hope this answers your question.

Svt frequency response (mag-phase).vi

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