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

Similar Messages

Frequency Response Function & FFT & Inverse FFT (problem of unit Volts-RMS)

Hello everyone,
I am currently working on a VI in order to compare two analog signals : the first one corresponds to the output signal (my reference) which is sent by my data acquisition card to a shaker and the second one corresponds to the input signal recorded by an accelerometer fixed on the same shaker. The final goal of the VI is to correct the analog output signal by using the analog input recorded signal in order to have the vibrations on the shaker which corresponds to what we really want.
To summary, I have a problem of unit with the Volts-RMS...
So this is my method for the VI :
First, I have to calculate the Frequency Response Function between the two analog signals (output and input). For it, I use the " Frequency Response Function (Real-Im).vi " which returns the complex values of the FRF in Volts-RMS (but I don't want to use this unit).
Then, I want to calculate the FFT of the analog output signal (my reference). There are two different blocs which can be used : " FFT Spectrum (Real-Im).vi " and " FFT.vi ".
The " FFT Spectrum (Real-Im).vi " returns the FFT complex values of the signal in Volts-RMS and the " FFT.vi " returns the FFT complex values in Volts (or say me if I am wrong, thank you). I really would like to use the second one because of the unit.
Then, I divide the FFT just calculated with the Frequency Response Function calculated just before.
For the end, I calculate the inverse FFT of that with the " Inverse FFT.vi " which use the complex values with the same unit than for the " FFT.vi ".
I don't want to use the Volts-RMS unit because I absolutly want to use the blocs " FFT.vi " and " Inverse FFT.vi ".
The problem is that I don't find a bloc which use the same unit for the Frequency Response Function. The " Frequency Response Function (Real-Im).vi " returns only the complex values in Volts-RMS unit. Maybe it is possible to convert it correctly? Or maybe there is an other bloc which can be used in order to calculate the Frequency Response Function with the same init than for the FFT and Inverse FFT ? Because I can't mix everything for the moment...
Thank you for your help,
Best regards,
Sebastien

Hello Preston,
No, I have not use the Sound and Vibration toolkit. I have only used the signal processing toolkit with the two toolboxes " Waveform measurement " and " Transforms ".
But I think that what I have done for the moment in my VI is correct (I have finished the complete VI). But I am not sure of the units (Volts, Volts-RMS...) and I would like to understand.
I have tried with the Sound and Vibration toolkit for the frequency response function (because you say me that it deals with all the unit conversion) and I can obtain the same results than with the " Frequency Response Function.vi " of the toolbox " Waveform measurement ".
But I would like to understand the units (see my previous post please). For example, for the FFT (the result is a complex), why sometimes it is in Volts, sometimes it is in Volts-RMS ? Is it possible to convert it ? How ?
If you want, I can attach on the forum my VI and that will maybe help you to explain me. Maybe it will help other people interested.
And if someone else can give me other precisions or advices about it, do not hesitate.
Thank you for your help,
Sebastien

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 function

Hello Everybody,
I want to calculate a frequency response function out of a acceleration and force signal. I measured this signals with an sinus sweep function from 1- 1000 hz.
Is there a way in diadem to solve this problem?
Thank you for your help!
Bye Vincent

Hi Vincent,
sure, you can use the FFT bloc (Analyze >> Signal Analysis >>FFT) and select the channel you want to use.
I hope this helps.
Marc

Microphone frequency response function

Hi all,
I am trying to do an impact test with 3 types of sensors to investigate surface properties of a material.
The first sensor is the sensor in the modal hammer.
The second sensor is a geophone , whilst the last sensor is an air-coupled sensor in the form of a microphone.
Aside from the microphone which is directly connected to the laptop by USB, the instrumented hammer and the geophone is connected to the NI 9233 which is connected to the NI USB-9162 which finally connects to the laptop.
This hardware doesn't support analog triggering therefore software triggering is used.
I am using a vi example from http://www.ni.com/example/28438/en/
I have successfuly setup the LabVIEW VI to read the raw data coming in from all the sensors and to obtain the frequency response of the geophone due to stimulus signal created by the impact hammer.
I have tried to obtain the frequency response of the microphone due to the modal hammer however no results show up in the graph for the frequency response of the microphone.
My question is this, what is going on and how do I fix this so that I may obtain the frequency response of the microphone?
I have attached my vi for your consideration.
Solved!
Go to Solution.
Attachments:
Y2014M01D28 IRnMicFRFRecAll .vi ‏190 KB

I cannot fix your VI because I do not have DAQmx or the SVFA toolkit. What I have attached is a simulation which shows some of the concepts.
The left side simulates acquisition of two channels (hammer and geophone) via the DAQ device and one channel of sound (microphone). Don't worry too much about the details. I just threw this together quickly. The hammer signal is a square pulse, the geophone signal is one cycle of a sine wave with noise, and the microphone signal is one cycle of a triangle wave plus noise. Each of the pulses is dleayed from the start of the acquisition by different amounts to represent the trasmission delays of the sound and vibration. The values chosen are arbitrary and do not simulate any physics.
I used the sampling rate, block size, and duration controls from your VI, so the numbers of samples and the sampling rates should be the defaults from your VI. The three graphs one the left show the three signals as generated. Note that the Microphone graph has X-axis autoscaling turned off. Also the data is all in waveforms or arrays. I avoid the use of the Dynamic Data Type produced by Express VIs because it effectively obscures the data structure.
The right side shows one way to do the triggering. I used Basic Trigger Level Detection.vi from the Waveform Monitoring palette and the Get Waveform Subset.vi from the Waveform palette. Note that there are separate trigger VIs for the DAQ channels and the sound channel. Because they are not started at the exact same time and they do not have the same dt (= 1/sample rate), the waveforms cannot be combined to use a single trigger. The geophone signal is synchronized with the hammer signal so one trigger from the hammer can be used to get both subsets.
Note that with the default delays the microphone pulse occurs 30 ms after the geophone pulse but in the subsets it occurs about 4 ms before the geophone. This is due to the independent triggering.
Since you do not have a common signal or trigger for both devices (DAQ and sound), you cannot know exactly what the timing relationship is between them. On every run the differences in the start times will vary. So you will need to do some kind of calibration to determine the time delays. I am thinking of some kind of periodic stimulus which will produce several pulses to both the geophone and microphone. The first pulses will have indeterminate delays but subsequent pulses should have reproducible delays. The relative spacing and directions during the calibration runs should be the same as the hammer position for the real experiment.
Lynn
Attachments:
Sound and hammer sim.vi ‏28 KB

Multichannel Frequency Response Function Block?

We are currently using the FRF block to evaluate our DUT with two
channels (X and Y). However our DUT has 8 output channels. Ideally I
would like to be able to wire up the existing FRF block with
multi-channel support (ie 1x X Signal, n x Y Signal) and get the new
block to correctly perform averaging. Currently if I wire up the FRF
block in a loop it averages CH1, CH2, CH3 etc which obviously is the
wrong result.
One solution was to have 8 FRF blocks - but this seemed a bit of an
overkill. Is there a n Channel FRF block anywhere - or has someone
modified it for n Channels? (if not, I suppose I will have to do the
modifications!).
Many thanks,
Marc

Hi Marc,
You asked if there was an N channel FRF block anywhere. LabVIEW 8.0 added multichannel support for the FRF. Modes include 1xN,
Nx1, and NxM, where NxM can be interpreted as a pairwise FRF (first
element of X with first element of Y, ...) or as all the combinations
of X and Y array elements.
Barring an upgrade to LabVIEW 8.0, the simplest solution is
probably your
approach of placing N FRF VIs on a diagram. This ensures that the averaging is the same as the 1x1 case with little programming effort.
For the 1XN case it should be fairly simple to export the averaging
state and manage that yourself. If you use one of the other
measurement VIs as a starting point (e.g. FFT Spectrum (mag-phase).vi)
please note that the state cluster that is used is intended to cover the
general case where each "channel" could have a different number of
points, different sampling rate, different timestamp... This is the
reason for the array-of-clusters-of-arrays approach. If you make a
simplifying assumption that each channel will have the same number of
points, same timestamp, etc. then you could utilize a much simpler state cluster
that contains some 2-D arrays to hold the multi-channel averaging information.
Hope this helps.
-Jim

Frequency Response Analyser or Impedance Spectroscopy

Hi
I am looking for a Labview program to allow the freq. response of say a
tuned circuit.
I will be using a GPIB Function Gen. From 1Hz to 10MHz and measuring the
peak-peak value across a shunt resistor that is in series with the tuned
circuit.
Thus as the frequency increases then the tuned circuit will respond
accordingly. If we take samples every 100Hz then we can produce a spectra
of the tune circuit.
I actually going to use this for measuring the char. of an electrochemical
cell.
Any Suggestions
Wayne

Thanks Carlos
The latter method I did not think of but will try it.
Cheers
Wayne
"JuanCarlos" wrote in message
news:[email protected]..
> Wayne,
>
> Looks like a sweep sine analysis should give you a good idea about the
> frequency response. Make sure that you measure the input and output
> signals of the circuit; this way you can just compeir RMS values and
> get the frequency response.
>
> Another method that you may want to look into is a broad band
> frequency response. Basically you send white noise to the circuit;
> then you acquire the input noise and the output noise; the you
> calculate the FFTs of this data and compair it; after some averaging
> you get the frequency response graph of your dev
ice. LabVIEW has a VI
> called Frequency Response Function that does a large part of the job;
> together with the "Frequency Analysis of a Filted Design.vi" example
> you can get a good idea on how to perform this test.
>
> I hope this helps.
>
> Regards,
>
> Juan Carlos
> N.I.

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.

TS-11 Frequency Response At 1/8" Jack

Hi,
I tried to play back sound from my new Pavilion TS-11-e100sr using 1/8" jack. The sound was audibly equalized for laptop speakers or maybe earphones and was terrible when it played back through outer sound system. I have measured output voltage at 1/8" jack playing back generated sine waveforms of constant level. The result is shown at the picture attached. As you see, there is an enormous 6dB boost down 100Hz and -3dB cut at mid range around 300-400Hz. At high frequencies there is also progressive 3dB boost up to 16kHz. I tried to find where I can switch off this EQ but I couldn't. Can you please instruct me what should I do to obtain flat frequency response at the output?
This question was solved.
View Solution.

I myself found a solution. I've replaced two Realtec drivers with Microsoft and this gives flat response close to AC'97 specs.

Microphone's frequency response

Hi,
Is it possible to plot a microphone's frequency response using sound and vibration? If it's possible, can anyone tell me how?
Thanks

Hi popcorn,
If you have the Sound and Vibration Assistant with SignalExpress, there is a step you can add under Analysis - Frequency Domain Measurements called, Frequency Response.
Here's a link to an article that discusses various functions, Sound and Vibration provides. Although the article discusses LabView, many of the same functions are found in SignalExpress as well. The Frequency Response Function can be found at the bottom.
http://zone.ni.com/devzone/cda/tut/p/id/3030
Aki T.

How to equalize an analog output by a known frequency response?

I'd like the analog output of my system have a flat response. What I'm going to do is first measure the stimulus signal, then using a filter to compensate the frequency response to make the following output signal flat. The difficulty is how to build a filter according to the frequency response. I know it's easy to do by using digital filter design of signal processing toolkit. But I need to do it by LabVIEW and the response is changed frequently. Any suggestions?
Bill

The filter does the signal "adjusting." Filters are typically characterized in the frequency domain, but the work on the signal fed them, which usually occurs sequentially in time. The lookup table is just to select the appropriate filter so you do not have to do a lot of calculations at run time. As an example, suppose you have just treble and bass and only one cut and one boost setting for each. You measure the stimulus and find the bass is too high and the treble too low. You select the bass cut and the treble boost filters for this run. If this is expanded to octave (or third octave) filters and 16 gain/attenuation settings in each band the lookup table approach saves time and may also provide a compact means of recording the equalization settings with your test data (rather than filter coefficients which do not actually indicate the response without characterizing the filter).
Lynn

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".

Frequency response of a filter

I have the filter coefficients of the filter I require in my program. I need to find the frequency response of this filter. Is there any function in LabVIEW that helps me to do this?
I guess I need a function which is similar to the freqz function in matlab for this.
Solved!
Go to Solution.

Thank you guys! I found out what I wanted. But thanks for guiding me.
I'll post the answer so that others can use it
First I found out the transfer function ( from the filter coefficients) of the filter by using:
Digital Filter Design toolkit => Utilities => From TF ( DFD Build Filter from Transfer Function.vi)
The output filter got from this was wired as the input filter to:
Digital Filter Design toolkit => Filter Analysis => Freq resp ( DFD Plot Freq Response.vi)
I got the required frequency response .
@Sd.Kfz.10 I coudn't use the FIR filter and IIR filter where coefficients are given as inputs (in signal processing toolkit) because I wanted the response of the filter alone. These FIR and IIR filter requires the input signal array.
I was using this for the linear predictive coding for speech recognition. I modelled the vocal tract as a autoregressive model (all pole filter) using a the AR modelling.vi in the ADSP toolkit. I wanted to see the frequency response of the modelled filter but I only had the filter coefficients.

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

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

Frequency Response Function NxM output format

Similar Messages

Maybe you are looking for