Diadem sampling rate

Hello!
We are using Diadem Version 11.0 and the hardware cDAQ with the voltage measurement hardware NI9229/9239.
I am trying to sample the voltage by at least 4khz, but I always get an error message that says that this rate isn't supplied by the hardware (the handbook says different things). I also changed in the options from windows timing to software timing or automatic timing, and also from hardware to software timing. Didn't work out. How can I maybe change the resolution to get a higher sampling rate?
Please help me , thank you very much in advance

Hi Lorenzo!
The problem is the NI 9229/9239 cannot sample at 4KS/s. If you check the NI 9229/9239 data sheet , it is said in the fist page you must use a sampling rate (Fs)= (50KS/s) / n; where n=1, 2, 3... 31; so it is not posible to sample at 4KS/s.
You can use sample rate (Fs)= 4,166 KS/s (using n=12); or you can use sample rate (Fs)= 3,846 KS/s (using n=13) but not Fs=4kS/s

Similar Messages

  • How can I add a curve with a different sample rate behind another curve to show it like one in the report

    I saved two curves with different sample rates with signal express in waveform.
    Now I want to add the curves behind and show them in a report. 

    Hello MReizner,
    Both the DIAdem VIEW and REPORT panel use the time information from your Waveform channels (make sure they actually have the waveform symbol, not the numeric data channel symbol in the Data Portal) to plot the data in the same axis system.
    In the example below I have two waveforms, one sampled at 5 Hz and one sampled at 1 Hz, both in the same axis with the same time channel. All I did was drag the data from the Data Portal onto the axis. DIAdem automatically takes care of creating the correct time channel and plotting the data with the correct points if the data is stored as a waveform.
    I hope this answers your question, please let us know if further clarification is required ...
    Otmar D. Foehner
    Business Development Manager
    DIAdem and Test Data Management
    National Instruments
    Austin, TX - USA
    "For an optimist the glass is half full, for a pessimist it's half empty, and for an engineer is twice bigger than necessary."

  • Audio sample rate does not match (HDcam to dvcam)

    I'm trying to import clips and keep getting this message:
    "The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape."
    Footage was originally shot on HDCAM and transferred to DVCAM elsewhere. Using FCP 5, am importing via firewire from a Sony DSR-11 deck. Using DV NTSC 48kHz Anamorphic as capture settings (though I've tried everything that I thought might possibly work with no success). The audio does not seem to drift over the course of several 5 minute or so clips. Clip settings show audio at 48 kHz (don't know if that's from capture settings or from actual data). Seems to me all audio should be 48 kHz 16 bits, so can't figure out what's going on. I have to export an EDL for the project to be finished in HD. Read some similar threads that ended in December, seemingly without much resolution. My broader concern is why this is happening; my immediate concern is do I need to worry about this right now since the media files will need to be recaptured in HD anyway. Any thoughts?
    Thanks

    A little more info. I'm having this problem on 4 tapes (from different cameras) that were transferred to DVCAM in a squished format to appear full screen on a 4x3 monitor. Video that was letterboxed and I can bring in with the standard DV NTSC capture settings does not have this problem. Still have the problem if I try to import the clips from the squished video with standard settings. Any thoughts?

  • Sample rate off the audio input and out devices do not match - what to do?

    This is fundamental, I know, but nevertheless I can't find my way around it. I get this error message when trying to recor:
    Sample rate off the audio input and out devices do not match
    and am asked to do this:
    Use the appropriate operating system or audio device control panel to adjust the sample rates of the input and output devices to use the same settingt.
    I have defined the sample rate to 44.1/16 bit in accordance with my inbuild soundcardt.
    I am trying to record from LineIn.
    When running on a M-Audio sound card I don't face any problems.
    HP 8560W, sound card IDT/High definition audio Codec
    Any suggestions?
    Knud
    Copenhagen

    You're sure you have set BOTH the input and output settings to 44.1 16 bit?
    Which version of Windows are you running?  There are a number of posts on this forum about how to fully access both the Windows Mixer and the Mixer for your soundcard.  Especially, you need to ensure that all "Windows Sounds" are turned OFF.

  • Separate sampling rate for two different channels for a USB-6009 daq

    Hi, 
    I am using a USB-6009 and incorporating the 'daq assistant' to change the sample rate.  I am trying to find a way to set the sampling rate to two unique values for two separate channels.  I've tried setting up two daq assistants and adjusting the sample rate different for each channel, though this does not work.  Is there any way to set the sample rate high for all channels then reduce the rate for a different channel - or an alternative?  I would appreciate any input on this, thank you!
    - Anthony
    Solved!
    Go to Solution.

    All tasks on a DAQ board that use a sampling clock must use the same clock.  Therefore, you cannot have two tasks on the same DAQ board sample at different rates.
    Alternatives would be:
    1. to combine all of the channels into a single task and just accept the extra data
    2. get an extra DAQ board to use in parallel
    There are only two ways to tell somebody thanks: Kudos and Marked Solutions
    Unofficial Forum Rules and Guidelines

  • How to acquire data from 2 chs of the same DAQ card at different sampling rate

    I am using single DAQ card (either 6013 or 6014) in my system i want to acquire data from 2 (or more) channels with following requirements
    1. sampling rate of each channel should be independant of each other (say one is 20 Hz and other is 15 kHz)
    2. data from all the channels should be acquired simultaneously.
    3. coding must be done using DAQmx VIs
    I have tried out following things
    1. I created separate task for each channel: i found out that two tasks can not run simultaneously even though the channels are different
    2. I tried out single task with two channels included in it. and i used 'channels to Read' property to determine from which ch. i want to acquire data: this method works fine if the sampling rates are same. but if i change the sampling rate of one channel it gets reflected in other channels as well.
    can somebody help me out to solve this problem.
    i will appreciate if somebody can post the sample code as my deadline is approaching
    Tushar Jambhekar
    [email protected]
    Jambhekar Automation Solutions
    LabVIEW Consultancy, LabVIEW Training
    Rent a LabVIEW Developer, My Blog

    Hi Dennis Knutson
    Thanks for your suggestion.
    Tushar Jambhekar
    [email protected]
    Jambhekar Automation Solutions
    LabVIEW Consultancy, LabVIEW Training
    Rent a LabVIEW Developer, My Blog

  • Multiple sample rate question

    I have one project with edits of different files all over the place recorded at 44.1 kHz. I have another project with edits all over the place recorded at 88.2 kHz. I wish to combine all tracks from these two projects. I realize I will need to convert the sample rate of all the 88.2 kHz files, but I don't want to mix down any track. I want all of the edits to be cut at the same places in time, just with the referenced files being at a different sample rate.
    I've tried everything. It seems that it is impossible. If I convert the sample rate of the referenced files, then open the project and change it's sample rate, the edits are now cut at differnt locations in time. It seems I have to mix down each 88.2 track, then sample rate convert these to 44.1.
    But I need to have all the edited segments as they appear in both projects in the same project. Any ideas?

    Don't record at 88.2! Why would you do that?!
    I'd back up and archive your current session first as you are probably going to have to do some very destructive messing.
    I haven't put this to the test but maybe you could try converting the route files in a third party software like ProTools. You'll get a bunch of prompts when you reopen Logic but this may work. But for me the simplest (albeit time consuming route) would be to convert all of your regions to new original sound files (you can do this from the audio pop down on the arrange page). Then in your audio pool select all unused audio, delete and go through each new file and convert the sample rate. It's a bit of a car crash situation needless to say next time preparation preparation preparation!
    Finally when you have been through this kind of process make sure that you are operating at the correct new clock rate before continuing otherwise this stuff still aint gonna work.
    Hope this is of help.
    Good luck.

  • IIR Filtering and response .vi: Butterwort​h filter magnitude response depends on sampling rate -why?

    Hi folks,
    I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
    I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
    Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
    This is fixed and prescribed for the application.
    However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
    To compare the 2nd and 3rd order's magnitude response of the filter I used the example IIR Filtering and response .vi:
    I was very astonished when I the found that the magnitude response is significantly influenced by the SAMPLING RATE I tell the signal generator in this example vi.
    Can you please tell me why - and especially why the 3rd order filter will be worse for the low frequency parts below 1 Hz of the signal. I was told by people experienced with filters that the 3rd oder will distort less the amplitudes which is not at all true for my relevant frequencies below 1 Hz.  
    In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
    THANK YOU for your ANSWERS!!!
    chris
    Solved!
    Go to Solution.
    Attachments:
    butterworth-filter-differences.png ‏285 KB

    Hello Lynn,
    thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
    However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
    I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
    So I assumed that everybody has it in his/her examples shipped with LV and it is not necessary to post it.
    I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
    So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
    I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
    Thanks for any further comments
    chris
    Attachments:
    IIR Filtering and Response_CH.vi ‏55 KB
    butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

  • Sampling Rate- How do I check and adjust the sampling rate on my Laptop?

    For the purposes of matching the sampling rates on my laptop to my usb microphone, (recording in garageband) how do I find out what the sampling rates are on my 2009 macbook?  Many thanks!

    GarageBand '11: Set the audio resolution: http://support.apple.com/kb/PH1873

  • How can I programmatically determine the capabilities of a card under NI-DAQmx (e.g. max sample rate, number of AI/AO/CTR channels, etc.

    Is there a DAQ_Get_Device_Info() equivalent for NIDAQmx? I need to iterate thru all the devices on my system, and build up a list of device capabilities. The system may include M-series and E-series cards.

    Attached is a program I've used in the past to determine number of AI channels. It could be easily modified to check for AO or digital or counter. Also, there is a ton of properties that you have access to (i.e. max sample rate, max/min voltage inputs, etc.) that are accessed as properties of the type of channel, or timing properties, as opposed to properties of the board. Check out the DAQmx C Reference Help (Usually at Start>>All Programs>>National Instruments>>NI-DAQ). Expand the NI-DAQmx C Properties, and look at the List of Channel Properties, and Timing Properties, etc.
    -Alan A.
    Attachments:
    Device_Info.vi ‏25 KB

  • Creative Audigy 2 NX Bit Depth / Sample Rate Prob

    This is my first post to this form
    Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
    The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
    I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
    Here are my system specs:
    HP Pavilion zd 7000
    Intel Pentium 4 3.06 GHz
    GB Ram
    Windows XP Prof. SP 2
    Thnx.
    -cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

    Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
    I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
    Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
    Thnx

  • High sample rate data acquisition using DAQ and saving data continuously. Also I would like to chunck data into a new file in every 32M

    Hi: 
      I am very new to LabView, so I need some help to come up with an idea that can help me save data continuously in real time. Also I don't want the file to be too big, so I would like to crete a new file in every 32 mega bytes, and clear the previous buffer. Now I have this code can save voltage data to TDMS file, and the sample rate is 2m Hz, so the volume of data increase very fast, and my computer only have 2G ram, so the computer will freeze after 10 seconds I start to collect data. I need some advise from you briliant people.
    Thanks very much I really appreciate that. 
    Solved!
    Go to Solution.
    Attachments:
    hispeedisplayandstorage.vi ‏33 KB

    I am a huge proponent of the Producer/Consumer architecture.  But this is the place I advise against it.  The DAQmx Configure Logging does all of it for you!
    Note: You will want to use a Chart instead of a graph here.
    There are only two ways to tell somebody thanks: Kudos and Marked Solutions
    Unofficial Forum Rules and Guidelines
    Attachments:
    hispeedisplayandstorage_BD.png ‏36 KB

  • Error message trying to synchronise audio and midi sample rate 42463

    I have a Rode Podcaster mic that is terrific. It has been working fine in GB. suddenly I am getting this when I try to record my podcast:
    'Error message trying to synchronise audio and midi sample rate 42463 recognised. Check conflict between Garageband and external device'
    Has anyone experienced this before? How do I resolve it? I'm still using my Version 6.0.5.
    Apple, is this a way of forcing me to upgrade to V10?

    Just click the icon on each track that stops it acting as a live input. It looks like a sound wave. Only have them clicked orange when using the track to record! I don't know if that will solve your problem!

  • Error while trying to synchronize audio and MIDI.  Sample rate 42804 recognized.  Check conflict between Garageband and external device.

    Sometimes, while playing back my software instrument songs, I get an intermittent pop-up error message, saying "Error while trying to synchronize audio and MIDI.  Sample rate 42804 recognized.  Check conflict between Garageband and external device."
    (Sometimes the five digit number is different, but remains a five-digit number beginning with "4".)
    Simultaneously, the song stops until I press the "Okay" button in the pop-up window.
    When I continue to play the song, the sound is jerky and clipped, and the playhead doesn't keep up with the song, and then suddenly jumps to the part of the song currently being played.
    There's also a sound of static.
    The issue seems to occur whether or not I have my MIDI controller turned on and plugged into my desktop Imac.
    Tony

    Hello,
    open your Audio MIDI Setup utility and set the input to 44100
    https://discussions.apple.com/message/12710638#12710638

  • Error while trying to sync audio and MIDI, sample rate 39100 recognised??

    When I press record a message appears saying 'error while trying to syncronise audio and MIDI, sample rate 39100 recognised. Check conflict between garage band and external device. I'm using a Behringer UMC-202 interface. The inputs are selected as the device but no audio is present either.

    Just click the icon on each track that stops it acting as a live input. It looks like a sound wave. Only have them clicked orange when using the track to record! I don't know if that will solve your problem!

Maybe you are looking for

  • Verizon Customer Service?  Now that's a question - with no good answers.

    Verizon has gone to great lengths to create in me something much more that a dissatisfied contract customer.  All the aggravation was exacerbated March 09 when my annual "UPGRADE" time came 'round.  Circumstances cause that I keep expenses to a minim

  • ITunes/iPod problem

    okay.. i have an iPod mini 4GB and whenever i normally plugged it into my USB port, a little box would come up and say what i would like to do. i'd normally just hit cancel but the last time i used it i decided to check the box to never bring up this

  • OAWebBeanFactory has been deprecated

    Hi, I'm new to OA Framework as well as java.. I'm trying to dynamically create a submit button on a standard page.. I'm using getWebBeanFactory.).createWebBean(pageContext,"BUTTON_SUBMIT") to create this in the processRequest of the Custom page contr

  • Implementing Oracle DCN with Coherence Cache in a weblogic 10 app server

    I m trying to implements a DCN ( Database change notification ) on oracle to notify a listener of an event of DB so I can update Coherence Cache. I followed the tutorial here and it is working fine using a sample program with a main method to execute

  • Why am I getting so many popups

    So every time I click on something on my web browser a new window opens with a random add. Then I have been getting tons of pop ups and my blocker is already enabled. Has anyone experienced this if so how do I fix it if I don't qualify for apple care