PXI-5610 RF Frequency Response adjustment

Hello,
I have noticed that adjustment for 5610 in RF frequency response is taking long time (about 2-3hr) with Cal exec 3.5, and I have tried to get the PXI chassis in a good enviroment and with some extra fans around since the adjustment has the legend "RF Freq. Response accumulated Meas. (Normalized to 45°C)
I am wondering if the enviroment has some important point here or it needs some update in a particular application inside Cal Executive 3.5?
I also have this question, Win7 has some advantage over WinXP in this particular or general adjustment?.
Br,
Omar

Hello Omar_Rdz,
Thanks for using NI forums! Calibration Executive usually takes a considerable amount of time for some of the PXI modules. By reviewing the Cal Exec manual it says that for the 5610 module it could take up to 240 mins (around 4 hrs). But definitely the temperature is a factor that can affect the performance not only of the calibration procedure but the whole system. Have you tried to execute the calibration when the controller and the chassis are cold? Also try to avoid dust accumulation in both the modules and the chassis because these can impact on the general performance of the system.
Answering your question regarding the OS, Win7 has a better performance compared to WinXP but it will also depend on what type of controller you are using. In order to give you a better answer, could you please tell me what controller and chassis are you using?
Regards,

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Frequency response measurements with pxi-5922

I’ am using signal express and the pxi-5922 digitizer together with the AWG pxi-5441 to analyse the frequency response of a buffer amplifier. See the attached signal express file. Many different ways to measure the frequency response have been tried and this is the best I came up with. It is basically two tone extract steps in a sweep loop. But I’ am still uncertain if this is the best way to do this kind of measurement. The fact that the detected frequency differs between the two channels worries me, even when the two channels of the pxi-5922 are looped. Is there a more accurate way to determine the frequency response?
Best wishes
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Frequency Sweep funkar.seproj ‏81 KB

Claudia-
Thanks for the response.
Regarding the CJC- When I switch it on, the temperature readings I get are very random, roughly negative 1 degrees. (I am operating right now at room temperature, and will be using J-type TC's to measure ~43 degrees C). Also, when I use the built-in CJC, the aquisition rate seems to slow down considerably. When I use the "user specified" everything seems to be ok, including the aquisition rate.
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sure.
I've attached my main vi and two sub vi's that I am using for the voltage aquisition part of my project. (Note:the current measurements are just voltage measurements multiplied by the recipricol of the resistance it was measured across, ie. 10).
I would like to keep this file as is because it writes to a file exactly the way I want it to. I'd like to have the temperature aquisition with the 4351 in the same vi as the 6030E so that they both stop and start at the same time. I am just not sure how and where to log the temperature data since there will be fewer data points than the voltage data. Any suggestions? Should I write two separate files? can I somehow append them?
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NI PXI 5610 procedure in Cal Executive

I am currently working with the calibration executive software version 3.3 to calibrate some PXI-5610 cards. For the AmplitudeAccuracy_Super10MHz it has me manually input two measurements. In turn it appears to perform a calculation to determine the As found dB reading on the calibration report. However, I cannot seem to understand why two measurements are made and why the calculations appear inconsistent. For the same test at frequencies from 20MHz to 2.7GHz with a power level of -50dBm I input similar numbers for both values on all frequencies but received vastly different values.
Here is a specific example:
1005MHz @ -50dBm 256average 1st value: -50.54dBm 2nd value: -65.17 As found reading on report: -.5837
1500MHz @ -50dBm 256average 1st value: -50.67dBm 2nd value: -65.12 As found reading on report: -.8231
Does anyone know what the calculations are for this procedure and why/what I might be doing incorrectly to make my measurements inconsistent?

Hi Jerimiah,
I am taking a look into your question and I wanted to clarify a few things. What hardware are you using to make the measurements? Is it the recommended hardware or an unsupported model? Could you provide a screenshot of a couple rows of the AmplitudeAccuracy_Super10MHz section of the calibration report if you have it? It would also be helpful to have a screenshot of the prompts to enter the two measurements. I am not sure of the calculation that is made based on your inputs, however I suspect the difference may be related to the passband flatness. Let me know what other information you can provide and I will continue looking into an answer.
Steve B

PXI 5610 problem

When running the PXI-5610 adjustment procedure in Calibration Executive , the main power suddenly shut off . then what problem will happen to the PXI 5610?

Thanks a lot for your reply.
After the PXI controller 's suddenly power off when performing adjustment , we restart the OS and found the PXI 5610's error infomation in the MAX : Error -200437,External calibration constants are invalid,and he card cannot be used in our Auto-Test-System anymore. I guess the calibration data in the EEPROM was missed or uncompleted,but i have no relevant softwre tools to read or write PXI-5610's EEPROM.
Could you give me some suggestions to solve this BIG problem? Thanks.

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?
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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).
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Frequency response: "very poor" on SB audigy

just encountered this very strange thing. Right Mark Audio Analyzer 5.5 reports and diagrams a "very poor" frequency response on my SB0570 Audigy at 44 Khz and 6 bits (not only) loop-back test.
http://img86.imageshack.us/img86/6409/rmbenchmark3pv.jpg
is this normal?

walkietv wrote:
ahaa! thanks. the problem is the guide is pretty old and the mixer and EAX look different. Could you tell me please exactly what should i change. I mean what to put in REC, Source in the mixer. I checked the EAX console-only time scalling was acti've. Any other setting in EAX? thank you!
- all options on all EAX panels tabs needs to be set 'OFF' (i.e. not enabled).
- if you're running the loop back test (line-out --> line-in), the Rec -source shall be set to AnalogMix() (or if you see the Line-In mentioned there as separate source then select it).
- you can set the monitor while recording by pressing the '+' sign above the Line-In slider (Surround Mixer) --> disable this option (i.e. no monitoring while measuring)
Other settings:
- windows audio performance options --> set acceleration and SRC to 00%
- on Surround Mixer, set master volume 00%, Source: Wave 00%, Line-In 50% and rec source 50% (adjust the signal level to ~ -dB by moving the Line-In or/and rec sliders)
jutapa

Raw Frequency Response Data

Hi
I'm working on a model of a LIDAR system that currently uses a PXI-5620 for data acquisition. I would like to include the card's frequency response in my model. I have found the spec. sheet which contains several frequency response graphs. I was wondering if anyone knows where I might be able to obtain the data points used to generate these graphs. It would save me time as well as improve the accuracy of the model.
Thanks
Marc

Hi,
Sorry, but we do not provide these informations to our customers. You have to generate these data points yourself.
Cheers,
Philip

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)

Fail to calibrate PXI-5610

Sory if this isn't the correct Board to post this question but I couldn't find a Calibration Executive Board
I'm trying to calibrate a PXI-5610 with the Calibration Executive 3.4.1 and I have the latest drivers instaled on my pc, the probem is that the calibration doesn't start and it gives this error:
"Open VI Reference in cex_Instr_RFSigGen_Init.vi->_NI 5610_RFSigGen Setup.vi->_NI 5610_RFSigGen Setup.vi.ProxyCaller
VI Path: C:\Program Files\National Instruments\Calibration Executive\InstrumentDrivers\LabVIEW\RFSigGen\rssma\cex_Instr_LabVIEW_RFSigGen_rssma_Init.vi
LabVIEW: The VI is not executable. Most likely the VI is broken or one of its subVIs cannot be located. Open the VI in LabVIEW using File>>Open and verify that it is runnable."
What can I do to fix this?
ps: I had to perform the calibration with Calibration Executive 3.2.1 because the client had some urgency...
Can anybody help
Thank's in advance
Bruno Ranhel

Hi BRanhel,
Please take a look at this knowledge base. This is for Calibration Executive 3.3.1 but it could work for the 3.4.1
Best Regards,
Vincent.O
National Instruments France
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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.

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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
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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.
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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

PXI-5610 RF Frequency Response adjustment

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