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
Stefan Johansson, SP
Attachments:
sweep.JPG ‏397 KB
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.
I measured the resitance of the Thermistor on the TBX-68T and it was about 5000 Ohms, as expected.
Just to make sure: When using the TBX-68T, do I need to hard-wire a thermocouple to Channel 1/auto-zero and another to channel 0/CJC? Because I connected a TC to channel 0 right now, but I wasn't 100%
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?
Thanks again. Hope to here from you soon.
Attachments:
EBlackMainDAQ.vi ‏107 KB
Save_Data8.vi ‏45 KB
Build_String_Array5.vi ‏33 KB

Similar Messages

  • Multirate sampling using CLK IN with PXIe-5922

    Hi
    I am working in an application where I need to change the sampling rate of an acquisition between 500 ks and 5 Ms without stopping the acquisition. The idea is to be acquiring continuously and be able to change the sampling frequency live depending on external events
    I was thinking about doing this using a PXIe-5922 and controlling its sampling frequency using the external sample clock input (maybe I would have to change the decimation factor by software as well?) connected to a signal generator (5404?) or some other clock generator whose frequency I can change without stopping the signal (maybe some specific development on PXI-7833R). My concerns are:
    Reading the 5922 spes I am not sure if I can change its sampling frequency between those values (500 kHz and 5 Mhz) using the external sample clock input without stopping the acquisition (my experience with other DAQ is that when you have to do any programming of the instrument the acquisition stops during the programming phase)
    If the above can be done, I guess my second problem will be to determine in the acquired data where a change of the sampling frequency has happened (as the driver will not know if I am using an external clock)… but I am not yet much worried because of this J
    I have not buy the hardware yet until I am sure I am in the right way... so any help will be much appreciated
    Best regards
    Guillermo

    Guillermo,
    Keep in mind the Reference Clock is used to generate the internal Sample Clock.  Changing the frequency of the Reference Clock will result in the same Sample Clock frequency.  Also, changing the Reference Clock frequency should not be done while acquiring since the device's clock circuitry needs to be configured based on the Reference Clock frequency.  I don't think this solution will produce desirable results for these reasons.  Unfortunately, the 5922 does not accept an external Sample clock from the Clock In terminal since the Sample Clock needs to be extremely high quality to meet the device's specifications.
    Do you really need to run continuously across different sampling rates, or could you afford a small amount of reconfiguration time between frequencies?  If you can, then you can also avoid the additional complication of not knowing where in your acquisition the sampling rate changed.  Could your application run continuously, then reconfigure the sampling rate when a trigger is received?
    Jeff B.
    NI R&D Group Manager

  • I'm having trouble with a multi channel acquisition and download with PXI-5922

    I am having trouble with getting data out from both channels of a PXI-5922.  My software seems to be working, but when I do the fetch there is no data for the second channel.  I am using the same format for all of the channelList inputs (0,1).  I've checked the instrument handles and anything else I can think of, but this is my first time trying to collect data on 2 channels of a digitizer.  Any help would be appreciated.

    Hi schliepe,
    Try checking your code against an existing example. The
    Developer Zone Example: Independent
    Channel Configuration with NI-SCOPE is a good VI to reference. Also note
    that there are several examples that come with the NI-SCOPE driver. They can be
    found in Windows by going to Start »
    Programs » National Instruments » NI SCOPE » Examples. Additionally, if you find that the examples are not working either, verify that both channels are functioning by running the
    NI-SCOPE Soft Front Panel.
    Please post back if you have any questions. Have a great
    day!
    Ryan D.
    District Sales Manager for Boston & Northern New England
    National Instruments

  • 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

  • TC measurements with PXI 4351 DAQ Card

    (I am using version 6i of LabVIEW). I am trying to create a VI that will use a PXI 4351 Card with a TBX-68T to take 4 thermocouple continuous measurements. I am also acquiring 8 voltages with a 6030E and TBX-68 and writing them to a file. My voltage data acquisition is modeled after "Buffered Continuous Analog Input.vi" I would like to aquire voltages and temperatures in the same vi, but using different devices and accessories. (I need a higher frequency for the voltage acquisition, and more precision with the temperatures). Is there a subvi I can use to collect the temperature data in the same vi as my voltage? I don't want the thermocouple DAQ to interfere with the voltage DAQ, either.
    I tried the NI435x thermocouple.vi
    but I couldn't get the Built-in Cold Junction Sensor to work or figure out how to combine it with my exisiting voltage acquisition. I also got NaN for my readings, and I checked the wire connections in all the sub vi's as suggested by another solution.
    Any comments would be great. Thanks in advance.

    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.
    I measured the resitance of the Thermistor on the TBX-68T and it was about 5000 Ohms, as expected.
    Just to make sure: When using the TBX-68T, do I need to hard-wire a thermocouple to Channel 1/auto-zero and another to channel 0/CJC? Because I connected a TC to channel 0 right now, but I wasn't 100%
    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?
    Thanks again. Hope to here from you soon.
    Attachments:
    EBlackMainDAQ.vi ‏107 KB
    Save_Data8.vi ‏45 KB
    Build_String_Array5.vi ‏33 KB

  • Quadrature Encoder measurements with PXI-6143 S-Series DAQ

    Has anyone used a S-series PXI-6143 DAQ to take encoder position measurements? I believe there are 2 counters available, if you have any examples of how to access and read from these counters it would be greatly appreciated.

    Hi Bentup,
    Measuring a quadrature encoder on your device is actually going to be a bit different than on an M series device.  The STC-2 chip, which is used in the M series, allows for A and B inputs specifically for encoder signals (used in the Angular Encoder task).  The original STC-1 chip which is used in E series and most S series boards (including your 6143) does not have these same inputs.  Instead, you may must a Count Digital Events task (taking advantage of the up/down line of the counter).  The Angular Encoder task is not supported on STC-1 devices.
    The article that lab_boy linked earlier actually mentions how to use an encoder with an E series device once you scroll down a bit (here's the link again).  This same procedure should apply to your S series--connect the A output of your encoder to the source of the Counter (this is the signal that you are counting).  Connect B to the up/down line of the counter (P0.6 and P0.7 for counter 0 and 1 respectively). 
    The downside with this method is that it is more susceptible to vibrations or noise on the encoder lines. If this is a problem for you, you can look into an external encoder conditioner like the one mentioned in the article.  I hope this helps!
    Best Regards,
    John
    John Passiak

  • Reflected power frequency response measurement

    My goal is to measure the reflected power from a DUT. I am using a Transmission/Reflection test set from Agilent to sample the reflected power and pass it to the PXI-5660. I sweep the signal over a range of 100 MHz to 500 Mhz and measure the reflected power signal at each point. A calibration sweep is taken first with a short at DUT end of the cable. I also sample the incident power with the T/R test set. The output signal is typically 0 dBm. My question is: What is the formula to calculate the reflected power from the signal measured? The signal I see on my application is not what it should look like.

    Hello Brian!
    Your best bet at finding that formula is probably Google.  I did a search and got many different relection formulas and how to derive them.
    Here is a derivation...
    http://www.physics.cornell.edu/sethna/teaching/sss/pythag/pythag04.htm
    Here is a discussion and definition..
    http://www.eagle-1st.com/notes/RPM/body.htm
    If neither of these are helpful if there something specifically with the 5660 I can help you with.  Let me know and have a great day!
    Allan S.
    National Instruments

  • 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

  • Questions about the frequency response step

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

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

  • 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".

  • 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,

  • 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)

  • Spectrum Frequency Error with PXIe-5641R and PXI-5600

    Hello,
    following situation: I am using the PXIe-5641R (in FPGA-mode) in combination with the PXI-5600 downconverter to get a spectrum.
    After trying out with different span widths I've experienced strange frequency errors in the range span <10M.
    To negotiate errors in my program I've evaluated also the NI example of http://zone.ni.com/devzone/cda/epd/p/id/6196 - unfortunately same error in frequency happens there.
    What is meant with frequency error, for example:  I have a transceiver which sends FSK with center frequency 443,92MHz. Readings with span-width = 12,5M shows overall good results.
    Reading with span width smaller 10M, eg. 3,125MHz, gives unexpected readings. 
    With a center frequency of 433MHz and a span of 3,125MHz the transmission is out of range of spectrum. When I add an offset of 2M to the calculated IF-frequency, I get right results.
    With a center frequency of 434M I've got readings at 434,92MHz, means at 1MHz too high (after adding an offset of 1M to IF freq readings are correct).
    With center frequency 435MHz readings are correct. The limit between correct and wrong is exact at 434,5MHz (resulting IF = 14,5MHz) readings at that center frequency are good, just one step below (434,49MHz) readings are 1MHz too high.
    Where is the error hidden? Like I said this behaviour happens also with the NI example.
    Thanks for helping!
    Message Edited by Lars.B on 05-20-2009 04:26 AM
    Message Edited by Lars.B on 05-20-2009 04:28 AM
    Message Edited by Lars.B on 05-20-2009 04:29 AM

    To get closer to the problem: I've managed to get constant right results in editing the VI which calculates the IF-frequency for the IF-RIO after setting center frequency and span of PXI-5600.
    But the new values make no sense at all. Documentation says that with spans higher 10MHz step size of center frequency is 1MHz, below it is 5MHz (with lower phase noise). After my modifications it seems that step size is 1MHz at all span widths higher 1MHz and 5MHz at span widths lower 1MHz. 
    Attachments:
    Get NCO Frequency.vi ‏8 KB

  • High frequency power measurements

    Hey,
    I'd like to know if there are developments in measuring electrical high frequency signals with labview without using an extern power analyser. At the moment i'm using a yokogawa power analyser but i'd like to know if it's possible to log HF signals without the help of a power analyser... Are there NI products on the market for this purpose?
    Thx,
    Andy

    Hello,
    In the case of frequencies up to 200 KHz, NI can provide several solutions using the 'standard' data-acquisition boards or digitizers (scopes), from a low to very high accuracy solutions.
    A good solution can be one of the high speed M-series boards (PCI-625x).  These boards have up to 32 multiplexed channels with a resolutions of 16-bit at a speed of 1 MS/s  (500 KHz).
    A better solution would be a S-series boards.  These data-acquisition boards sample all input channels simultaneously.  We have boards with 2, 4 and 8 channels and sample frequencies of 10 MS/s  (up to 5 MHz if needed).  S-series board are the boards with product numbers PCI-61xx.
    The best solution is to use a digitizer (scope).  Also here a lot of possibilities going from low to higher bandwidth and resolution.
    The most flexible is the PXI-5922.  A 24-bit digitizer if you sample @ maximum 500 KS/s.  This board only exist in the PXI platform.
    Then NI has 8-bit digitizers (normal resolution for scopes) from 15 till 125 MHz bandwidth.  If you need a higher resolution they have solution up to 14-bit (very high for scopes) @ 100 MHz.
    Please give your local NI Office a phone call.
    They have technical engineers who can discuss your needs and provide you a solution.
    Best regards,
    Joeri
    National Instruments
    Applications Engineering
    http://www.ni.com/ask
    Make our forums great:
    If you like the answer, don't forget to "Kudos!".
    "Accept the Solution" if your question is answered!

  • CTR works with PXI 8196,PXIe 8102, fails with PXIe 8100 - why?

    My client has reported a problem.  
    For years he has used a PXI 8196 RT Controller with PXI 6602 Counter card and my software has given good results.  They have 20+ of these systems and they have worked well.
    Now they are moving to PXIe 810x controllers, for cost reasons.
    WIth a PXIe 8102, the same code also works perfectly, measuring total counts over a period, as well as instantaneous frequency.
    With a PXIe 8100 - the exact same code reports DIFFERENT answers. The reported frequency is always 1% HIGHER than actual (For example, a known 4500 Hz input is reported as 4500 Hz on 8102, but as 4545 Hz on 8100.
    This happens on any channel, and swapping just the controller will make the problem come and go.
    Here is the CONFIGURE code, where the channels are set up (again, this has worked for years).
    Here is the SAMPLE code:
    Basically the CONFIG code configures the thing to count edges.   I do this because they need an accurate count over a 20-minute period, in addition to instantaneous frequency readings.
    The AVG TIME is a user-settable number defining how long a period to average, when showing the "instantaneous" frequency.
    So, I create a buffer for N samples, corresponding to that period.
    At SAMPLE time, I read the counter.  I replace the oldest value in the buffer with the newest, then subtract the newest - oldest to get the total counts in the sample period.
    The PULSES PER COUNT item is a scaler, to account for a 60-tooth wheel, or something.
    So, this same code has worked perfectly for years, until I plug an 8100 code in.  Then the result changes by 1 %, and EXACTLY 1%?
    The CPU burden on the bad controller is 31%.
    Any ideas?
    Steve Bird
    Culverson Software - Elegant software that is a pleasure to use.
    Culverson.com
    Blog for (mostly LabVIEW) programmers: Tips And Tricks

    Well, the controllers are not in my own hands.  I have an 8196 controller and on that, the CPU time is between 2 and 4%.
    But the 8100 and 8102 controllers are in my client's hands.
    I haven't gotten any hard timing numbers other than I saw the 31% figure reported on the video monitor.
    It's hard to believe that it would be EXACTLY 1% if it was CPU overburden.
    My software includes a calibration facility; here is a run from the good 8102:
    Here is a run from the 8100:
    This was with a reference digital freq generator.  You can see the one case where everything is within 0.1 Hz.
    the other case has everything EXACTLY 1% higher.  My only explanation is that the scan engine is running 1% slower.
    Steve Bird
    Culverson Software - Elegant software that is a pleasure to use.
    Culverson.com
    Blog for (mostly LabVIEW) programmers: Tips And Tricks

Maybe you are looking for

  • Power failure during installation

    This isn't a question, but a cheer. While installing Yosemite on my Mac Mini, the power went out at my house. Those of us old enough can remember the warnings that used to be attached to system installs---DO NOT TURN OFF YOUR COMPUTER DURING INSTALLA

  • Why "exceeding 80% of host limit" warning?

    I am a first time host, just started in Oct. '13--AC Meeting. Paid for 3 hosts. While 1 host was on with 5 attendees, she couldn't chat and got a warning notice that we were almost exceeding the 80% limit for hosts. How can this be if only 1 host was

  • Dynamic field catalog

    Hi all,      i have a strange problem, in table WITH_ITEM the field ITT and wt_qbshh , we have different kinds of tax codes and tax amounts. i was displaying these as two columns in my report. but my client ask me to display the wt_qbshh field in dif

  • Using an xsl page with an xsql servlet.

    I haven't been able to get my xsql page to display using an xsl page. I am trying to simply display one viewObject using the 'ViewObject Show' component. The xsql runs fine without trying to format it with the xsl page but nothing displays when I try

  • Audition CS6 Crashes constantly

    I have a certain Session (.sesx file) that I am now trying to modify after I have sent it back to Premiere. I need to make some adjustments to my mix such as raising the volume levels back up for all the tracks. Unfortunately, now Audition is consist