on the sample rate of PCI-6723

Similar Questions

• If I use a few channels on PCI-6259 to increase the sampling rate, channels numbers can I be contiguous?

  We are looking to buy a card PCI-6259 usable on a Linux machine. We would use NIDAQmx to access the card. If we were to use only a few channels to increase the sampling rate, do I need to select specific channels?

  For example, the card is 1 MHz. If I select the channel 1 and channel 2, I can taste each channel at 500 KHz. could I choose 8 channel and channel 13 and still be able to sample each channel to 500KHz? Or need of specific channels to use when a subset of channels are selected?

  Thank you

  -Tom

  Thanks for the quick response.

  -Tom

• DMM (NI 4070), how to correctly set AC Freq (bandwidth) by the sampling rate

  using a NI4070 multimeter and I see the max connection is 300 kHz by respect it.  But I don't understand how to set the min and max, acFrequency according to the sampling frequency or speed reading.

  6 1/2 digits resolution, the speed can vary from 0.25 s/s to 100 s/s and this range corresponds to a lower end on the connection (minimum acFreq) from 1 Hz to 400 Hz.

  (Q1a) - is the playback speed, controlled by the minimum setting of IviDmm_ConfigureACBandwidth?   or vice versa?

  Otherwise, I do not see how to control the rate of reading or the sampling frequency.   IviDmm_ConfigureMeasurement only allows you to control the range and resolution.

  (Q1b) - is there a way to directly control the sample rate (digitizer) or playback speed (dmm)?

  (T2) - the upper limit of the bandwidth of AC always seems to be at 300 kHz... is there still a reason to reduce this maximum value?

  (T3) - Finally, unlike the traditional niDmm function, the resolution via the IVI configuration should be passed as absolute value; does directly when number of digits and the beach?   For example if I want to 6 1/2 digit to 300V range, I guess that by the specifications that the resolution should be set at 0.001 V... followign, if I want 5 1/2 digits to 1V range, the resolution should be set to 0.00001 V?

  Hi Rjohnson,

  I'll try to answer your questions as best as I can:

  Q1A.  The ConfigurACBandwidth function is used by the driver OR DMM to calculate the good aperautre for the measure.  So yes, by adjusting your minimum frequency, you will affect your reading speed.

  Q1B.  Your reading rate will depend largely on your measuring cycle.  To get a fast measuring cycle, there are a few things that you can adjust.  You can programmatically control your time aperature, as well as your time to settle.

  Q2. I can't find a reason to change. This parameter is only used for error-checking and verifies that the value of
  This setting is less than the maximum frequency of the device.

  Q2B.  I think what you say is right, but I'll need to check on that - I'll let know you as soon as.

  Hope that helps.  "" "I would recommend checking the explanation of the Cycle of the DMM measurement in DMM help' devices ' NI 4070" DMM Measuments "DMM measurement Cycle.

  Take care!!

• How to increase the sampling rate in this VI?

  Hi all

  I have recently inherited this mess of a VI and can not figure out how to increase the sampling rate. I tried to change the "ms of waiting ' clock, but it does not add more data points." The main VI, as well as the Subvi, which contains a Daq Assistant to a load cell and LVDT is attached. Any ideas on how to improve the sampling without a complete overhaul would be greatly appreciated!

  Thanks in advance!

  If you are grateful, feel free to give congratulations and mark the topic as resolved.

• How to improve the sampling rates high in samples mega 3rd spartan Board

  I am trying to acquire data in spartan 3rd Council mega samples rate can you post me examples to this effect

  So far, I've got to get the frequency in Kilosamples of sampling, but megasamples requires complex code

• The sample rate conversion

  Hello

  If I scan documents to Audition 2.0 48 kHz and then convert to 44.1 using software for recording on CD, would there be loss of quality?

  Thank you.

  If it has voice recordings and you need an original CDs can just master at 44.1 k and have done with it. Since there is no content in your files anywhere near the limit of Nyquist (this is half of the sampling frequency and represents the highest frequency at this rate, you can save), then all you need to do is an extra oversampling when the format of distribution requires.

  The speech contained in frequencies up to 12 kHz (hopefully) and even if you add music the answer will not exceed 20 kHz - and in any case, this is where the human ear cup (in children - rather less than that of adults). Therefore recording at 44.1 k means that you will capture all this, and you will be able to register without creating large files by scanning of a plu top noise load audio - which is all that recording at a higher sample rate will reach.

  If you need pace to a different distribution format, then you have lost nothing - just do it when you need to.

  I record a lot of original acoustic music and unless specifically requested to (which is never arrived, I might add...) I always master to 44.1 k. It's been proven by research academics caution that no one can tell the difference, then what is the point of a high sampling rate, besides wasting disk space unnecessarily?

• cwdsp. Sine wave is where the sampling rate?

  Hello

  According to the method above (CWDSP. Sine wave), the parameters are the following:

  (n, a, f, Phase)

  n As Variant - [Input] number of samples to generate.
  Amp as Variant - [Input] Amplitude of the signal that results.
  f As Variant - [Input] frequency of the signal resulting in standardized units of cycles/sample.
  The phase as a Variant - initial phase [output] in degrees of the generated signal. Output, the Phase is the phase of the next portion of the signal. Use this setting in the next call to this function to simulate a generator of continuous functions.

  We are not lack of sampling frequency?

  example:

  I want to generate the next sine-

  FREQ = 1 kHz

  sampling frequency = 10 kHz

  (Number of samples) block size = 1024

  Amp = 1

  How will you use this function for this signal?

  I think (but I'm not sure of it...) is: CWDSP. SineWave (1024, 1, 1/10, 0)

  There is an example: "power spectrum".  In this example, they do not mention the sampling frequency and the signal is generated as follows:

  CWDSP. SineWave (1024, 1, 0, 1000/1024)

  No mention of the sampling frequency.

  Thank you

  Hey Rafi,

  Both of your assertions are correct.  The frequency of de.1Hz at no time is the equivalent of what you would get from sampling equipment of a wave of 10 kHz to 1000 s/s; in both cases, you will see a cycle of the wave every 10 samples, as you are pointing out.

• determination of the sampling rate and the frequency waveform data record

  Hello

  I write a simple program that collect data from a triaxial accelerometer input, convert it to a frequency spectrum, and then save the time domain and the frequency of the waveforms in an external file separated.  I don't understand how to set the sampling frequency, however.  On the DAQ Assistant, I updated the acquisition mode "Samples continues" and read samples is 2 k, which corresponds to the total number of data points that are collected.  How can I program sampling for awhile, it 30 seconds, for example?  Wouldn't be better to set up a trigger, as it will continue to collect data up to what I told it to stop?

  I also want to save waveform data in a separate file that can be easily seen by other computers that have not installed Labview.  I have currently the program put in place to convert a text string of the waveform of the time domain and then save it in a text file or a spreadsheet.  It works fine, but I would also like to record the frequency wave, which is a different type of data.  How can I do this or is there a better way?

  My program is attached.  Thanks for your help!

  Here's how you can use the shift register to build the table, and also where you can choose to play exactly 100 samples per while the loop iteration.

  

  Brian

• Writing data to extend the acquisition of data for the sampling rate high file

  These are the tasks that I have to do to take noise measurements:

  (1) take continuous data to USB 6281 Office, in a sample of 500 k (50 k samples at a time) rate.

  (2) save data continuously for 3 to 6 hours in any file (any format is OK but I need to save in a series of files rather than the single file). I want to start writing again file after every 2 min.

  I enclose my VI and pictures of my setup of the task. I can measure and write data to the file continuously for 15 minutes. After that, I see these errors:

  (1) acquisition of equipment can't keep up with the software (something like that, also with a proposal to increase the size of the buffer...)

  (2) memory is full.

  Please help make my VI effective and correct. I suggest to remove him "write in the action file" loop of consumption because it takes a long time to open and close a file in a loop. You can suggest me to open the file outside the loop and write inside the loop. But I want to save my data in the new file, after every 2 min or some samples. If this can be done efficiently without using Scripture in the measurement file then please let me know.

  Thank you in advance.

  This example here is for a single file and a channel, you should be able to loop over that automatically. The background commentary should be the name of the channel, no group namede the name of the channel in the control.

  

• Setting the sampling rates

  I just got hearing CC 2014 and when I try to record it says the frequency of sampling of the input and output devices do not match. How to set those rates at the point 8.1 of Windows?

  Very well.  You must go in the Windows sound control panel, and then select the Read tab.  Select your output device (if you have no external card it although that may not be), and then select Properties.  Then click on the Advanced tab and there will be a drop-down list for flow of sample and depth.  I would say 16 bit / 44100 Hz (CD quality) for pure audio which will end by if he's going to end up in video on CD or MP3 or 16 bit / 48000 Hz.

  Then, go to the Windows Sound Control Panel, click the recording tab and follow the same procedure (highlight, properties, advanced) and select the same settings in the drop-down list

  This should sort out you.  Longer term, consider an external USB audio interface - it'll be a lot better, and if it has ASIO drivers, it will stop Windows applications to change the audio settings without telling you.

• On the NI PCI-6221 fast sampling rate question

  Hi I was wondering if someone can answer a question of sampling rate on this card to PCI-6221 (http://sine.ni.com/nips/cds/view/p/lang/en/nid/14132).

  Especially if I wanted to transmit simultaneously (analog output) and data acquisition (analog input), what is the sample rate max I could use. Kind regards.

  Since the 6221 is multiplexing the analog input, your question for I / simultaneous ao is possible for one channel of the only. If your "simlutaneously" can include delays (e.g., 100us), you may be able to work with several AI channels as well...

  HAVE the multiplexes, workable sample rate given that the total sample (250 kHz) frequency divided by the number of channels that you use. AO is faster than HAVE it, so it does not reduce this number.

  hope this helps,

  Norbert

• PCI 5154 sampling rate

  Hello

  What is the sample rate max 5154 PCI for two channel inputs? The manual States the 2GS/s is for one channel only. So, am I not able to get a bandwidth of 1 GHz for the simultaneous measurement of two channels? Thank you!

  Hi gbhaha,

  First of all, TIS mode up to 20 GECH. / s using an ADC, while your real time sampling uses two converters a/n at the same time to a single channel.  Take a look at these diagrams that I linked in my first post for more details on this architects.

  About the difference in the bandwidth between the 5153 and 5154 - the 5153 has 500 MHz of bandwidth in its circuits, even when acquiring at faster sampling rates.  The 5154 1 GHz of bandwidth, this is why it is more expensive.

  Kind regards

• Different sampling rate with the same connector AIO, Labview FPGA

  Hello

  I use LV 2009 with the new Toolbox FPGA and an NI PXI 7854R. I acquire an analog signal with a sampling frequency of 600kS / s. I need as the sampling rate for the processing of the data, but I also need the signal sampled with a much smaller, variable sampling frequency to a FFT.

  I've attached a picture to clarify, in a simple example, I'm looking for.

  

  I tried with the structure case only take each ' iht iteration, but did not get the expected results.

  Does anyone have another idea how to solve my problem? Of the, "Resampling" express VI in the funtion FPGA palette does not help me.

  Thanks in advance,

  Concerning

  Hello

  the connector for the analog input is a "shared resource", so you should he alone in your FPGA Code.

  Find attached an example that shows how to perform this task of analysis.

  Concerning

  Ulrich

  AE OR-CER

• Audition 3.0 how to disable ASIO and the default Sample Rate recording?

  Hi people,

  New here, but I hope someone can help me with a few questions, I'll have with Audition 3.

  Firstly, some background questions.

  I use hearing parallel to a broadcast audio broadcast program called SpotOn,

  This software requires that I run the sound card, a RME Madiface XT in 48 k mode, and that all the outputs that it uses are defined as WDM Windows so that the windows kernel mixer can control them.

  This means that when I use the hearing at the same time, I have to configure it to use the "Audition 3.0 Windows Audio" driver to stop him from taking control of the sound card directly and change setting which prevent SpotOn to see its output.

  The problems I encounter are that hearing itself seems to randomly change mode in the edit window ASIO driver, I suspect that this happens when I import audio data from a key which is from 44.1 to modify for use in SpotOn. This often seems to not only make the outputs of the card its invisible to the windows kernel mixer but also change the sampling frequency of 44.1 sound card and stops work SpotOn.

  The second question I have is that the sampling frequency of default record when I record in edit mode is always 44.1 and if used it again change the map sound 44.1 and causes the same problems, I'd be very keen to know how to change this default to be 48 k if possible.

  Then.

  What I ultimatly looking is...

  1. a way to disable the ASIO drivers in hearing so that it is not only this option is available, and cannot use the Audition 3.0 Windows Sound Drivers.

  2. a way to make the sampling rate 48 k to stop people choosing 44.1 mistakenly when saving default record.

  Any help or advice that anyone can give would be much appreciated.

  Thanks in advance for your comments

  What other pilots ASIO sees your installation of 3 AA? If it's the Madiface one and you use only WDM drivers you can just uninstall the ASIO RME driver?

• Getting the error «for input and output sample rate does not match...» »

  I am trying to create an audio file, but when I click on the record button I get this error:

  «The sampling rate of the audio output devices and audio don't match.» Audio cannot be recorded until this problem is corrected.

  Use the appropriate operating system or control panel of audio device to adjust the sampling frequency of the output devices and audio entries to use the same settings. »

  I have the preferences set to use operating system controls and it seems to work because when I change in the Win 7 Pro CP it adjusts at the hearing.

  So I do not know which need to be addressed. I use the video card as the source and not the speakers thus mean the output must be la carte too?

  Please help, I'm confused.

  Thanks for your time!

  Got an answer to a previous question asking how to record and thought, this could be useful for others.

  rothmannnyc,

  

  February 18, 2013 16:33 in response to talbot

  It takes a Windows setting and a setting of the hearing.

  You will need to go into your settings from Windows recording devices - if you see a speaker in the lower right of your screen, right-click on it and choose recording devices.  Find your audio input source, select Properties, go to advanced.  You should see a drop down menu with options for the quality and frequency of sampling, mono/stereo.  In general, you'll want to 44.1 kHz for audio; 48 Hz for the video.

  At the hearing, go to Edition/Preferences/Audio Hardware and select the same frequency of sampling for output.  I went through the same question a few months ago, and it worked for me.  Good luck!