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.

Similar Messages

Frequency response of real analog filter using mydaq

Hello!
I am trying to find the frequency response of a analog filter using the NI myDAQ. There are several examples of this in Labview, but they are usually analyzing digital filters, not real world filters. There is one example, see attached vi, that simulates measuring a realworld signal using a simulated frequency generator and a simulated multimeter. I believe that this could be done for real using the mydaq.
Can someone please help me convert this VI to run off of the mydaq?
Thanks!
Attachments:
Frequency Response.vi ‏21 KB

Thank you Kyle, this is exactly what I was looking for. I searched the examples that came with labview 2010, the signal processing module, the 8.6 DSP module and the signal anaysis toolkit, but I had not yet installed the elvis cd so I didn't run across these VIs. Thank you once again!

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
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not gauranteed.
Have a great, day!
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Attachments:
4472 Phase Linearity.xls ‏21 KB

FS7 audio Frequency Response 50Hz - 20KHz?

Hello guys, I'd like to point out an issue that I couldn't find on the forum regarding the audio frequency response.I've always worked with pro camcorders that recorded 20Hz to 20KHz, which is the standard frequency us humans can hear. So I was surprised when I saw in the FS7 manual that the audio Frequency Response is 50Hz to 20KHz. Is this a typo? Or is the camera actually limited to 50Hz for the lower freqencies?I would assume that the internal mic would be limited to that range, yes, but not the internal recording of the sound? Why limit the camera to 50Hz? Any light on the subject would be appreciated. Thanks!

The tests I posted are for the mic inputs, not line inputs. I did test both, but since I often run & gun and cannot support all the extra gear to make the FS7 mic inputs on a par with Sony's EX3 mic inputs, I'm requesting that the high pass filter added with firmware 2.0 be removed in futre updates. We have the wind filter already. No additional filter is needed. While Genelec monitors are pretty good, they're not noted for deep bass response. Most studios use the ubiquitous NS10, which are only good to 80hz, which is why few people notic the deficiency. The custom built system in my screening room has a -3dB point of 7hz. I noticed that even piano sounds thin on this camera. that doesn't surprise me, since the rolloff starts at 200hz. All I'm asking for is the same quality I'm used to with my EX3's mic inputs. Only 0.5dB down @ 20hz and no thinning of the midrange. When I do large budget orchestra shoots (rare) I use 24/96 sampling, 8 channels and large diaphragm studio condensers. My recordings earned critical praise from Peter Aczel at The Audio Critic. I was a sound engineer for 4 decades before I got my first digital camera. In addition to that, I was consulted as an independent expert on infrasonics for chapter 17 of Ethan Winer's THE AUDIO EXPERT, published by Focal Press. I was responsible for the information on subwoofers and how they operate. Acoustic and speaker design share a long history with me, and my 'day job' is amplifier design, modification and repair. I'm too intimately aware of what's going on in the signal path of an audio system and when modern digital systems deviate intentionally from DAT-quality, it really bothers me. I made a big deal about this in 2007 with the Sony HVR-V1U, and Sony must have listened, because they got it right with the EX1 and EX3 cameras. But sadly, they've reverted to these games with the FS7, though not as badly as 2007.

Frequency response at low frequency

I'm working on a bandpass filter, and I'd like to get the frequency response showing that the frequencies outside the lower and higher cutoff frequencies are being cut off. However, the correct plot is shown only when my cutoff frequencies are high (roughly from 1000-8000 hertz). When I use low cutoff frequences(roughly 4-5 hertz), the plot is incorrect. So how can I get the frequency response to my low cutoff frequencies? Thanks.
P.S. In the code, some parts are irrelevent. In the front panel, the only relevent part is the frequency response plot at the lower right corner, and the specs above it; in the block diagram, only the upper half(with IIR and FRF) is relevent. Thanks.
Attachments:
BME_Pressure_Sensor_V1.00.vi ‏591 KB

Hi Manson
There is a bug in your diagram since you connected the number of samples where you should have connected the sampling frequency.
The sampling frequency is related to the pace at which you take the measurement.
Usually, Fs = 1 / dT
where Fs is the sampling frequency and dT is the time interval.
It should work better.
In any case, to have a better resolution in the low frequency range of your spectrum computations, you have to increase the number of points of your data because there exist the following relationship between dF (space between 2 points in you spectrum), dT, and N (number of data points) :
dF = 1 / (2 x dT x N)
Doc-Doc
Doc-Doc
http://www.machinevision.ch
http://visionindustrielle.ch
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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 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.

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

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

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

Missing content of wrapped response on a filter

Hi,
I've posted this to the w.d.i.workshop group last week; no replies:-(
Perhaps someone here could help? Here's the post and some additional
info I came accross in the mean time:
I wrote a response filter that is supposed to wrap the entire page
before sending it to the browser and then store it in a database.
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I've figured out that the missing data come from skin.properties in my
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Cheers,
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Did you ever get this resolved? I have the exact same problem. I get everything else fine except for the head section.

Frequency response of a filter

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