FFT - Frequency Response

Hi everyone!
I'm having a problem on my time domain and fft plot.
In the attached VI below, i made a Chirp signal and perform FFT (left side) and I wanted to have a plot which is moving similar to the Chirp signal I got from the net, which is at the right side of this VI.
I wanted my plot to be moving, or in real-time, I know this is just in the settings, but i cant figure it out. I've tried countless time already to obtain a plot similar to those at the right side of this VI. Also the amplitude of my FFT, is very small from the original amplitude value of 1.
Thanks, for the response.
Attachments:
sample_sweep.vi ‏113 KB

Hi gijude,
it could be so easy with some elementary math:
Reading online documentation like WIKIPEDIA can be a great source of wisdom…
Best regards,
GerdW
CLAD, using 2009SP1 + LV2011SP1 + LV2014SP1 on WinXP+Win7+cRIO
Kudos are welcome

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 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

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

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.

Loudspeaker Frequency Response

Hi all! I'm making a program to measure the frequency response of a loudspeaker. I'll be using a sweep to test the loudspeaker.
Particularly i would like the program to be like this, which is real time. How can i implement this in LabVIEW. This is the screenshot
Details about the plot.
First graph:
Shows a plot in real-time of a frequency sweep with a constant sine sweep amplitude of 1 V. When sweep is started, the graph shows a plot of FFT moving from left to right, with peak of FFT at maximum amplitude of 1 at corresponding frequency of the sweep.
Second graph:
Shows the plot of the Sound Pressure Level in dB versus freqeuncy.
Please refer to the picture and video link below.
https://www.youtube.com/watch?v=sKC3ioWXG38, skip to 4:10

Assuming your idea is to sweep a frequency into an amplifier connected to the loudspeaker, and measure the frequency response with a microphone:
You need to monitor the signal at the loudspeaker terminals to account for any non-linearity's in the signal generator and amplifier.
You need to know the frequency response of the microphone, this is difficult, and that is why calibrated microphones are expensive.
You need an anechoic chamber so that the results are not affected by any room resonances.
Your sound level plot is in dB (A). My understanding of the A weighting is so that the human perceived loudness is constant across the audio frequency range. If you are concerned about loudspeaker performance, is it worth discarding the complexity of this additional frequency response curve?
This will be a difficult project. Please let us know how you get on.

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

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.

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

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

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".

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.

Why do I get a Track out of memory error while running open loop frequency response?

MatrixX Build 61mx1411: I get a "Track out of memory" error when I run the Open Loop Frequency Response from the MatrixX pull down tools. What can I do to prevent this? We are running on an HP B1000 with 768 MB of RAM under HP-UX 10.2.

In the old days of Mx say Version 5 and prior the user actually selected the amount of memory that would be allocated. Depending on the size of the model etc. you would have to allocate memory. In version 6.0 and going forward there is no need for the user to manually allocate the memory.
Build {rstack=50000,istack=200000,sstack=50000,cstack=500 000}
If this is a command in a script file that you are running and the error is resulting from that then I would try commenting out everything after the letter d in the word build and then starting it back up.
i.e. only use Build
I don't believe that there is a way to manually allocate the initial SystemBuild Stack size.
I believe initially the stack size is set to 10010.
However, one way
you can manually set the initial SystemBuild stack size,is to create a large StateSpace as soon as you start up SystemBuild. This will prevent piece-meal reallocs while using SystemBuild.
You can created a new SuperBlock in SystemBuild and then drop down a StateSpace Block with 199 inputs and 199 Outputs and 1 State and entered ones(200,200)as the StateSpace Matrix without any problems. This would resize this internal stack to at least 40000.
You really should not have to do this but if that helps then you might think about doing this in your startup.ms file you could use SBA or load the file then you could delete the superblock and begin working.
"Bob" gave me this little tid bit.
Please let me know if any of this is of use.
Garrett
Garrett Thurston
[email protected]
Phone: 781.993.5540

Room frequency response measurements

Does anyone know of a Shareware program available that use your PC's sound card to play pink noise while recording from the microphone, and then display the room's frequency response?
I have a high-end stereo system and am looking for something that will show me what my room modes are doing.
Thanks in advance.

thanks but I'm not looking for pink noise to drown out others. It's to use as a measurement. see my post again.
1.33ghz PowerBook g4, 17-inch Mac OS X (10.4.7) 1.5gb ram

FFT - Frequency Response

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