Reception of samples at other frequency of digitization

Hi!
There are samples of a sinusoid frequency of 500 Hz received at frequency of digitization Fs = 10 кгц (I receive by means of Sine Waveform). It is necessary for me, I use these samples , to receive values of samples at frequency of digitization Fs-dFs distinguishing from 10 кгц on some value (dFs) .I need to receive values of these samples by means of function Sinc, however the sinusoid received as a result differs from a sinusoid received at the same frequency of digitization (Fs-dFs) by means of Sine Waveform. Help me, please!
Alex
Attachments:
Samples Sinс.zip ‏24 KB

Hi Alex,
I see that you have posted a VI here to take a look at, but I am very unclear what your goal is from the information you have provided.  I understand you might not be receiving your desired outputs, but it will be very difficult for me to advise any changes in logic, without a better understanding of what you need.  Please provide everything you can to clarify what it is you are working with and what it is that you need from this application.  In the meantime, I would suggest trying to search from within some of the resources we have at ni.com/support.  Here you might be able to find one of our toolkits will provide this exact functionality.  Thanks.
Jason W.
National Instruments
Applications Engineer

Similar Messages

  • Sample Clocked Buffered Frequency Measurement

    I have a circuit board clock which I want to test.  The clock should output at ~12 MHz.  I'm using a USB-6343 and would like to use the Sample Clocked Buffered Frequency Measurement mentioned in the X series manual but am unclear on how to set it up.  I believe I need an external clock but am not sure if I can use this card to generate this clock and what rate I should use; but more generally I'm looking for an example on how to set up this measurement in LabVIEW using DAQmx.
    In the end I'd like to get an accurate average measurement of the clock frequency over ~1 second if possible and am willing to use whatever method would best work with the 6343 card.
    Thanks in advance

    Hi Gollum,
    There's a few examples that you could take a look at that come packaged with LabVIEW. If you open the NI Example Finder  (Help » Find Examples), then in the Browse tab click the following: Hardware Input and Output » DAQmx » Counter Input.  In there we have a few examples on how to read in the frequency using DAQmx calls.  The property that is mentioned in the Sample Clocked Buffered Frequency Measurement (CI.Freq.EnableAveraging) can be located using the following method:
    In LabVIEW go to View » Class Browser, when the Class Browser window open, for Object Library choose DAQmx and for Class choose DAQmx Channel.  In Properties & Methods expand the following: Properties » Counter Input » Frequency » Measurement Specifications » Enable Averaging.  The Class Browser can be used to find several property nodes, and is very useful.  You can play around and find various properties that can be changed in there.
    Hopefully this helps.  
    Matt S.
    Industrial Communications Product Support Engineer
    National Instruments

  • What are supported sampling frequency and digitization rates for Zen Micropho

    Some MP3's are playing back slowly. It is inconsistent, though, since files with the same digitization rate and sampling frequency will behave differently - some right speed, some slow.
    The bulletin board says:
    "My tracks don't play at the correct speed e.g. they play too slowly, why?
    Chances are they are encoded in an unsupported sampling frequency..."
    How do I find out what the supported sampling fequencies and digitization rates are?

    Thanks for the info. I guess I was looking for something more specific - the exact bitrates and sample rates that Creative claims to support. Would you know where official and comprehensi've data can be had? There must be a tech spec somewhere.
    It is common these days in business to see a recording of, say, a conference call or seminar presentation at 32k bitrate/025Hz, or even 24k bitrate/8000Hz, posted to a company's website for download by those who could not be there, and MP3 players are increasingly used for their replay. Companies use low digitization rates because there is no need for hifi and the files are much smaller: less storage, faster download.
    I'd be surprized to think that Creative don't have compatibility with the standard range of rates offered by ubiquitous programs like Audacity and dBpower, the latter being one they themselves recommend!

  • Sample Sweeping Audio Frequency

    I am able to do the harmonic analyzer on a specific frequency but wanted to get input on two things:
    Analyzing a sweeped signal (say an audio range of 20Hz to 20kHz) - Do most programs just sample every few periods  within the file and then analyze the next?
    How do programs do "real-time" analysis.  I assume there is some sort of aperture time of sampling so sample every 1 second of data?  In addition, I have only devised a way to analyze a wav file (converted to an array).  Is it possible to do "real-time" sampling from the incoming audio?

    Hi ngay528,
    Essentially, you can't do a THDN analysis on a signal with a continuously changing frequency, as by definition, THDN required one fundamental frequency to get any useful information on it.  In order to get proper data from a THDN the best option is to acquire your data in chunks at discrete, or stepped frequencies.  If you know where these frequency changes are occurred you can then perform a THDN on those individual chunks of data. 
    Regards,
    Hassan Atassi
    NI Community Project Engineer

  • Sample Rate vs Frequency

    "The increments across the x-axis of the graph are controlled by the sample rate.  In this example, 8000 units along the x-axis represents
    one second of audio. "
    Actually not. One should sample at twice the highest possible input frequency.
    Renee
    "MODERN PROGRAMMING is deficient in elementary ways BECAUSE of problems INTRODUCED by MODERN PROGRAMMING." Me

    You see basically an FFT describes the frequencies and amplitudes of samples or events. If one undersamples one cannot hope to describe the event and that's what it's all about. If one samples at a rate less than twice the maximum frequency one cant cant
    hope to describe the event because one cant be certain that data is take fast enough to describe it. I'll give you a worse case example.
    Consider an electrical sine wave occurring at exactly 1 khz. The signal oscillates from +1 volts to -1 volts at the rate of one khz, Let us consider what would happen if we sampled this signal at 1 khz, We would see a constant voltage at the input although
    the signal is changing sinusoidally at a 1 khz rate, If we sampled at 1.5 times the signal we'd see variations. And if we sampled at twice the maximum frequency we could be sure we're sampling at a rate which will allow description of the input.
    Renee
    "MODERN PROGRAMMING is deficient in elementary ways BECAUSE of problems INTRODUCED by MODERN PROGRAMMING." Me
    Not quite.
    When you're taking a true analog signal and "digitizing it", it will never have the resolution of the analog signal ever again.
    The faster the sampling rate, the closer the approximation but the actual digital result is and will always be a square wave (not triangle wave, not sine wave, a square wave).
    The faster it samples it, the closer the "stair" to the next one, thus more closely approximating the original sine wave, but it can never be completly true to the original signal.
    This isn't related only to sound but to any true analog waveform being "digitized".
    Please call me Frank :)
    Well maybe you could do something like a Bezier curve or
    Bézier Spline that would only need 4 points for each half of the wave in order to match the wave or something digitally? I don't know much about it but it seems to me you could somehow match an analog waveform digitally.
    Please BEWARE that I have NO EXPERIENCE and NO EXPERTISE and probably onset of DEMENTIA which may affect my answers! Also, I've been told by an expert, that when you post an image it clutters up the thread and mysteriously, over time, the link to the image
    will somehow become "unstable" or something to that effect. :) I can only surmise that is due to Global Warming of the threads.

  • Audio sample at variable frequency

    I think this is the right forum for that question.<BR>
    Do anyone know if it is possible to play an audio sample an modify the frequency while playing according to a variable value?
    Any example?

    write yourself a Mixer that can change the frequency of the data flowing through it.</p>
    Mixer
    <p>
    Or find one on the internet that does it.
    <p>
    matfud

  • Sampling/T​riggering externally with DSA cards with even sample spacing in frequency space

    Hi there,
    Is it possible for the NI DSA cards to trigger on a signal to record at a desired time, at an un-even spacing in the time domain. For my application I need to have the samples evenly spaced in frequency space.
    I would set the first channel for acquisition from the source. Would use the external trigger port, or the second channel for the trigger signal? 
    This is for a optical coherence tomography system.
    Thanks! 

    Hi Anthony,
    You may have to excuse some of what I say since I am still a bit uncertain of what you need to accomplish due to my lack of understand of OCT. If you wanted to start your acquisitino at a given time by using an external trigger, most cards have a trigger input which will allow you to do so. However, using an external clock will be much more difficult. While it will be difficult to use an external clock signal with a DSA device, resampling may be a better option. Here is an article which shows some of the resampling functions which, by my understanding, may accomplish what you need to do. Let me know if this puts us in a better direction.
    Regards,
    Kent
    Applications Engineer
    Digital Multimeter Home

  • IPhone 4 reception - no different to other phones?

    No flaming!
    The problem that people are describing I can replicate on my iPhone 4. I thought I would test with another phone and see if the death grip could cause the same problems.
    The antenna on this unit is obviously to the right of the phone - watch for yourself. I have not experienced many problems on the SE unit despite being able to recreate this.
    http://www.youtube.com/watch?v=QZ4xWVSVLwI
    The only difference is that the iPhone does not switch frequency when the antenna is blocked - that can be resolved
    Comments?

    I've conducted a test, to verify the theory that it is really the contact made between the two antennas that is provoking the reception degradation...
    I realized that there are two points of contact possible, one at the bottom left, but also one at the top, so I told myself: "If it's really making contact witht he two antennas with the hand, that is generating this issue to appear, then it should also happen if I do this on the other point of contact, that is at the top left corner making sure to cover the two antennas"
    Guess what? NOTHING happen if you do so... So apparently making contact between the two antenna is not the reason of this issue!
    That would explain why this issue still appears when you deactivate both WiFi and Bluetooth and grip the phone at the bottom left corner...
    So it's clearly more complicated than anyone tried to find out here!

  • 3g iphone - reception, software upgrade and other things.

    I've read a lot of disgruntled people's posts on here on how bad Apple is, how crap the iphone works etc. I thought I might just add my 2 cents worth.
    I bought my phone the day it came out (no waiting in lines for me, it's who you know). Anyway, my iphone has worked FLAWLESSLY. Not a single problem. I have just upgraded to 2.1 without a hitch. My reception is always good, and that is on a crappy Australian Optus network (I can differentiate between the phone and network problems unlike most). I have had one or two dropped calls, but that was inside a building were the 3g dropped out to 2g while I was talking on the phone.
    I have since recommended the phone to 5 other people who, like myself are absolutely satisfied with our product (no, I don't work for Apple nor am I an Apple Fanboy).
    I have used countless smartphones over the years.....I can remember when scratching my head why a Windows mobile phone would not sync properly with a Windows PC (hands up all those who have been there eh?). People need to look at it for what it is, a cutting piece of hardware that has the ability to develop and grow....unlike ANY other phone out there. When was the last time you upgraded your Nokia or HTC phone (without having a mental breakdown).
    I for one am very happy with my purchase and unlike others, don't expect my phone to do my taxes or wash the car for me. Time to bring some reality back to some of these posts.

    This has never happened to me with every firmware update released since 3rd party apps became available.
    All 3rd party apps on your iPhone should be in your iTunes library on your computer. Is Sync Apps selected under the Apps tab for your iPhone sync preferences with iTunes and was this selected before installing the firmware update?

  • How to lock sample frequency for 5673 and 5663

    Hello,
    In general I'm trying to lock transmitter and receiver together.  It seems easy to lock the carrier frequency, however no matter what I do, I seem to have a drift in my sampling frequency (on the order of 1ppm). 
    Is the sampling clock in the 5663 digitizer tied to the same clock reference as the LO?  ... I didn't think I would need the TClk mechanism here since I don't care about delay ... 
    Any thoughts are greatly apreciated.  Below are a few specifics. 
    I'm using the 5673 as transitter and 5663 as receiver.  I've noticed the folowing:
    1. When each are using a 'Reference Clock Source' = OnboardClock I have a noticeable carrier offset at the receiver (eg 5.8 GHz carrier has  ~7 KHz offset) and this is fine. 
    2. When each are using a 'Reference Clock Source' = PXI Clock, with the chassis physically tied, I see no noticeable carrier offset at the receiver. 
    3. When 5673 is using 'Reference Clock Source' = OnboardClock, 5663 using 'Reference Clock Source' = ClkIn (ClkIn/Out physically tied), I see no noticeable carrier offset at the receiver. 

    Not sure if this is the proper forum.  I created a post in the 'NI Support' area and LabVIEW is were it got placed.  ... arg ...
    I'm going to try and move this to a more apropriate forum:  ... 'High Speed Digitizers' I guess.

  • How to lock sampling frequency for 5673 and 5663

    Hello,
    In general I'm trying to lock transmitter and receiver together.  It seems easy to lock the carrier frequency, however no matter what I do, I seem to have a drift in my sampling frequency (on the order of 1ppm).
    Is the sampling clock in the 5663 digitizer tied to the same clock reference as the LO?  ... I didn't think I would need the TClk mechanism here since I don't care about delay ...
    Any thoughts are greatly apreciated.  Below are a few specifics.
    I'm using the 5673 as transitter and 5663 as receiver.  I've noticed the folowing:
    1. When each are using a 'Reference Clock Source' = OnboardClock I have a noticeable carrier offset at the receiver (eg 5.8 GHz carrier has  ~7 KHz offset) and this is fine.
    2. When each are using a 'Reference Clock Source' = PXI Clock, with the chassis physically tied, I see no noticeable carrier offset at the receiver.
    3. When 5673 is using 'Reference Clock Source' = OnboardClock, 5663 using 'Reference Clock Source' = ClkIn (ClkIn/Out physically tied), I see no noticeable carrier offset at the receiver.
    Solved!
    Go to Solution.

    Hi Clayton, 
    The digitizer sample clock time base source is different from the Reference Clock source. I've copied  a description below of the difference between the two from the digitizers help. 
    Clocking:Reference (Input) Clock Source:
    Specifies the input source for the PLL reference clock.
    Clocking: Sample Clock Timebase Source:
    Specifies the source of the sample clock timebase, which is the timebase used to control waveform sampling.
    Yes, the default configuration of the NI 5663 is for the NI 5652 to export its internal 10 MHz reference to the NI 5622 so that the NI 5622 and the NI 5652 devices are frequency-locked. The NI 5663 can also be configured to lock to an external (10MHz only) reference source. The NI 5663 can also be configured to lock to the PXI 10 MHz backplane clock. Locking to the PXI 10 MHz reference does not require a cable, but this configuration does not provide the same frequency and phase noise performance as the NI 5652 internal Reference clock. All of this information is provided in detail in the NI RF VSA Help. See the directory below:
    Regards,
    Travis Ann
    Customer Education Product Marketing Manager
    National Instruments

  • What is the start frequency for the digitizer 5142

    Dears,
    what is the start and stop frequency of the digitizer 5142 which is 100MS/s?

    Hello Ahmed.Abdulbaky,
    When you say “start” and “stop” frequencies, I assume that you are inquiring about the maximum and minimum sampling rates of the 5142 digitizer.  Is this correct?  If so, then take a look at page 13 of the NI PXI/PCI-5142 Specifications, and you will see under Real-Time Sampling that the minimum and maximum sampling rates are 1.526 kS/s and 100 MS/s, respectively.
    To answer your question from your other post, yes, you can change the sampling rate of the digitizer through the NI-SCOPE API.  If you are using LabVIEW with the device, there is a VI called “niScope Configure Horizontal Timing.vi” that you can use to set the sampling rate, by wiring the desired sampling rate to the min sample rate input.  This VI can be found in the functions palette under Measurement I/O » NI-SCOPE » niScope Sonfigure Horizontal Timing.vi. 
    Chris_G
    Sr Test Engineer
    Medtronic, Inc.

  • PCI-6023E DAQ card maximum sampling frequency

    Hello
    I am using PCI-6023E DAQ card in pc-based ETS solution (and writing appilication in LabView 7.1 with RT module). The card has 200kS/s maximum sampling frequency, but it can be set for much higher sampling frequencies and the waveform acquired appears to be correct (i.e. i've tried setting it fo 1MS/s and sampling 400kHz sine, which is obviously above Nyquist frequency for 200kS/s card, but on spectral graph, main peak is at 400kHz). Is the card driver doing some kind of free/coherent sampling?
    Moreover, when sampling frequency is set to 200kS/s, the card seems to be doing same thing - i.e. for 200kS/s and sample block size of 200kS, graph should be updated once in a second, but it's updating slighty slower.
    I'd really appreciate if someone could explain me (or gave me a link to materials) what exactly is happening here? Is driver doing some background work, or maybe it is problem with network latency/unstability ? What is the impact of this effect on real-time aquisition?
    Thanks in advance
    Jan Kienig

    Since the fundemental is 4 times the nyquist, then what you are measuring is an alias of the fundemental. This works well as long as the fundemental is a repetitive signal. Sampling every other peak and every other node looks the same as sampling every peak and node. Tektronix exploited this on their 7S series sampling heads. Another use of this phenomena is the effective demodulation of high frequency signals as long as the bandwidth meets nyquist. As with your card, if the input amplifier supported it, I could extract modulation information from a 500 MHz signal so long as the the bandwith of that modulation did not exceed 100 kHz.
    Parker

  • High frequency

    Hi everyone, im pretty new with Labview, and I need a help. I would like to simulate square wave with high frequency (40MHz).
    1. Can anyone help me, how can I simulate the signal (test2.vi) in a moving (time) axis?
    2. I would like to catch the datas and write it into .txt or excel. I found this example (0807-LVM_Beispiel.vi). Can I get the datas like this from my 40Mhz frequency generator?
    A little help would be nice. Thanks.
    Attachments:
    test.jpg ‏525 KB
    test2.vi ‏29 KB
    0807-LVM_Beispiel.vi ‏88 KB

    thanks for your help.
    Ive been searching, and now I've changed my test2.vi into test CIC.vi and modified it. If you may see it, im trying to simulate the signal with frequency 40MHz and, with sampling frq. 800MHz and sample rate 1000samples or max. ~300.000samples (and save it in a txt.file). Actually, my task is, that im gonna need to make a labview program that can take as many points as possible, that might be not periodic. I mean, we gonna use NI card (in near future, not bought yet) to take some input datas, and to get all the data-samples in high frequency range. Is it possible to make such a program (capture the data) without knowing in the first place which NI card that we are going to use. But i believe, we gonna buy a digitizer, digital I/O and might also mxi controller (I've looked at the offered device list) from NI. Thanks a bunch. Is it much easier after buying the devices? (like getting the device's driver and maybe program that support it).
    One other question, in my test CIC.vi, im saving my points into .txt file (only the amplitude). How is the trick to take also the x-axis points (time axis) and save it in the same .txt file.
    Attachments:
    test CIC.vi ‏49 KB
    CIC 1 sub.vi ‏21 KB
    Save 1Data.vi ‏18 KB

  • Pulse train generation fails with certain values for "number of samples"

    I'm generating a retriggerable analog output signal, and so I'm using a counter as the sample clock (see: Retriggerable AI Using Retriggerable Counter). I am finding that, above a certain number of samples, and only for certain values of the number of samples, the counter task gives me error -200305, "Desired finite pulse train generation is not possibe." The error crops up only when actually starting the task.
    The analog signal that I'm trying to generate will be about 800 kHz, so my counter is set to run at the same frequency. I find that the counter task works fine if the number of samples to generate is anywhere between zero and 671,088 samples. Setting the number of samples to 671,089 gives the error above, as does 671,090 samples and so on. However, using 671,096, the counter task works fine. After that, the counter seems to output fine only if the number of samples is divisible by 8.
    The only thing I can think of is that (617088 samples) / (800000 Hz) = 0.839 s. At the internal clock rate of 20 MHz, 0.839 s is 2^24 samples, and it is a 24-bit counter on this hardware. So if it's this internal counter rolling over, that's fine and I can work around that. But if that's the case, what I don't understand is why increasing the number of samples in increments of 8 samples still works.
    The hardware is a PXI-6733 board, running with LabView 7.1.1 and NI-DAQmx 8.1.

    Hmmm,  multiples of 50 & 100?  Now I'm puzzled again.
    Here's how to make sense of the 100 kHz timebase idea though, even if it turns out not to be the right explanation.  For a retriggerable finite pulse train, you actually use a pair of counters.  If you were to program it manually, you could set your output counter to generate a continuous pulsetrain at 800 kHz using the internal 20 MHz timebase.  This output counter would also be configured to use the other counter's output as a digital level-based pause trigger.  So the 800 kHz pulsetrain is only output while the other counter's output is, say, high.
    The other counter is configured for retriggerable pulse generation.  The pulse duration or high time should be set for (# pulses) / (800e3 pulses/sec).  This other counter can be configured to use the 100 kHz timebase, so its high time would then have to be an integer multiple of 10 usec.
    So let's see...  An 800 kHz pulsetrain is possible with a 20 MHz timebase (exactly 25 cycles).  A 700 kHz (28 + 4/7 cycles) or 900 kHz (22 + 2/9 cycles) is not.  So when you request those other frequencies, you actually get a near approximation.  I dunno if DAQmx can be queried for the actual value correctly or not -- I recall an early version that reported back whatever freq you had asked for rather than what it actually used.  Queries based on ticks (rather than time or freq) did return what was actually used, as I recall.
    Let's suppose a request for 700 kHz gets truncated to 28 cycles of the 20 MHz timebase making a 1.4 usec period.  Then 50 of those periods becomes 70 usec, which is evenly divisible by the 100 kHz timebase.  Bingo!  (Note: 70 is the least common multiple of 10 and 1.4)
    Now suppose the request for 900 kHz turns into 22 cycles of the 20 MHz timebase, or a 1.1 usec period.  Now it takes 100 of those periods to get to 110 usec, which is also evenly divisible by the 100 kHz timebase.  Bingo again!  (Note: 110 is the lcm of 10 and 1.1).
    Did you follow the method here?  It should help you figure out expected results for various output freqs and #'s of samples.
    -Kevin P.

Maybe you are looking for

  • Unable to open the member selection(pov)

    Hi, We are facing a critical issue and we are using FR 9.2 Version: Issue: We have FR Reports and Books in Workspace, when we try to open a Book or Report from workspace it asks for the POV to select. If we select member from the selection for the fi

  • Car charger, Mikegyver system

    Whether or not I buy a new MacBook is dependent on being able to charge it in my car. Has anyone used the Mikegyver MagSafe cable system for doing that? If so, are you satisfied with it? It seems to be the only option available besides using a DC pow

  • Tax Breakup in PLD

    Dear All, I require the following tax breakup in PLD Ex: Total................................................303743.40.......A Packing charges.................................2316............B Excise Duty@10%..............................30606......

  • Not able to execute the explain plan in development.

    Hi In TOAD when try to click "Explain plan current statement." icon, I getting the below error message. ORA-00604: error occurred at recursive SQL level 1 ORA-01950: no privileges on table space 'DEFAULT_DATA' can anybody give some idea to solve this

  • Can I install the LR4 upgrade and still keep my LR3 installed?

    I have LR 3 and I would like to upgrade to LR 4. Can I install the upgrade to LR 4 and still keep my LR 3 installation intact so that I can switch back and forth between the two versions if I want to?