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

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• acquisition of data with different sampling rates high

  I have a few questions on the use of the OMB-DAQ-3005 with different sampling rates high.

  For our application, we have 8 analog inputs. Which two are a quick response and should be sampled frequently.  We have an encoder quadrature (CPR 1000 running at 1800 rpm).  We plan to sample X 4 encoder.  For the analog inputs for the quick response, we want to trigger a sample of each pulse or each a few pulses, thus creating a timestamp with the position of the encoder with respect to position index as well as two fast analog inputs.  We have data correlating the analog inputs with the position of the encoder.   Other analog inputs, we want to measure relatively slowly (for example once every 5 dry or similar).

  How can I go on the configuration of the two (or more) sampling rates different wherein I can taste entered at different frequencies?  Also, is there a way to reset the encoder count after outbreak of the index as I have the position of the encoder with respect to the index?

  Maybe you'll find someone here who uses the OMB-DAQ-3005, but this forum is really more designed for LabVIEW programming issues.

  I've never used the OMB-DAQ-3005, but out of curiosity, I took a glance at the Manual of OMB-DAQ-3005.  The answer to both your questions are:

  1. you cannot run a hardware DQA Multiplex (like this one) at independent rates by channel.

  2. the OMB-DAQ-3005 supports an Index Z feature to reset the counter - look for documentation on how to configure any software interface you are using.  If you get stuck, you can try to discover media appropriate for instrument channel.

  Best regards

• 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

• 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.

• 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!!

• 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

• 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

• 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.

• on the sample rate of PCI-6723

  According to the specification of PCI-6723, faster sampling rate is 45kSample/s, 32 channels working simultaneously. But he's always fine when I put the sampling frequency to 200kSample/s, 32 channels working simultaneously. This configuration will damage the material?

  Ok. I assume you mean update rate of sampling frequency not.  As long as you use the onboard buffer you can reach 204Ks/s 32 channels.  Since you do not get an error the device and things work it is probably what you're doing.

  You had asked the rate could only be achieved you would have been a mistake.  And the material would not be damaged by incorrect update rate adjustment.  You're ready to go!

• To the bottom of the sample unnecessarily high resolution in the book?

  Hello

  I said in the forum Photoshop my workflow is strange and I don't have no need to do any resizing photoshop because there is the option at the bottom of the sample images of a unnecessarily high effective resolution is it in InDesign.

  So if I have a lot about 17.4 inches wide color jpeg photo and I want to put in JPEG 5.4 "black and white in a book of say 15 chapters and then export the book in pdf format, how can I avoid using photoshop? And where is the control at the bottom of the sample at 300 dpi or less?

  Thank you

  Garrett

  (I'm down here too... sniff.)

• 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?

• Plugin data for a possible evolution of file Extension?

  Hello

  I have files with the same file format data in files from there. Problem: the file extension is never the same, only the first 2 letters:

  My extension is: *.pa + a number incrementing

  Example *.pa01, *.pa02; *. PA03, etc..

  I tried to define * .pa use expansion, but it does not work. DIAdem allways aks me what Plugin data should be used for the file.

  Someone at - it an idea to set a file Extension variable?

  Thanks in advance

  Hi Marco,.

  Unfortunately, we do have an elegant solution for you.  It was the case that you can define extensions of use with one or more '? ' character, but that doesn't seem to work more in DIAdem 11.0.  My only suggestion is that you can list on all variants in the file extension string:

  *. PA01, *.pa02; *. PA03...

  Who has at least 100 different file extensions, you must set this one right channel

  I hope that this limitation will get some attention in the future.

  Brad Turpin
  Tiara Product Support Engineer
  National Instruments

• How to import the samples B2B zip file

  I'm new in the B2B and download examples of B2B of Oracle's Web site.
  I want to import b2b - 101 - Custom_1.0_orders_generic_file.zip and select the check box replace existing metadata.
  But B2B shows the error:
  Import file b2b-101 - Custom_1.0_orders_generic_file.zip failed. Error -: B2B-52321: no meta data found to import.
  
  How to import some B2B tutorial?

  You must unzip the example and follow the readme.txt file

  Rgds,
  Nitesh jegou

• Sampling rate higher for the measurement of precision meter

  I have a BNC 6259 M Series DAQ USB. I am currently using the DAQ Assistant to perform simple cash rising measured with the measuring mode single sample on request. I tested my VI with a known square wave with a function generator signal and it clearly lacks a few edges. I think that the solution lies in faster sampling. However, I was not able to understand how to use clocks to set up continuous sampling mode. I tried the forums and I found articles that were close, but not quite exactly the problem I am facing with as (http://forums.ni.com/t5/Digital-I-O/trying-to-use-NI-6251-s-DIO-port-as-input-and-output/m-p/448035#...) or (http://forums.ni.com/t5/Multifunction-DAQ/Using-Counter-of-PCI-6024E-with-Quadrature-Encoder/m-p/984...). Any guidance here would be great.

  In fact, the calendar should not have anything to do with the edges being detected.  Configuration of a sample clock for a county of just edge task allows you to enjoy deterministically in the account register and has no impact on the edges which can be counted.  Also, the analog examples really have nothing to do with what you seem to be asking questions on.

  ... So it leaves the question unanswered as to why you might miss the edges.  Perhaps the following information could shed some light on the question:

  1. it is possible that you do not configure the counter exactly as you think you are.  Can you post the VI you use?  As a point of reference, count digital events shipping example does not use any clock sample timing and just questioned the value of the register count with a software loop, but the meter should not miss all this edge on the input source.  There are examples that are timed by the material available as well, but this is not necessary, unless you need a constant specified dt between your counter samples.  To use the examples of the timed sample, you will need to generate a clock of either another subsystem on the map or use an external clock.

  2. assuming that the configuration of the counter is not the issue, there may be a problem with the method that you use to determine if you are away from the edges.  How do you know that you are away from the edges?  The function generator produces only a finite pulse amount?  You start the meter before start out impulses?

  3. If the two points above do not raise red flags, it seems likely that the meter is registered just not some of the impulses of your FGEN.  Can I assume that the output of the FGEN is 0 - 5V TTL?  What is its frequency and duty cycle?  The maximum external source for the meter on the M-series products: DAQ (like the 6259) is specced at 20 MHz, but this depends on a clean signal with good connections.  At frequencies above it, the bandwidth of the front-end of the PFI lines becomes limiting.  If you have an available specification document for your FGEN I'd like to be able to see it.

  I hope this gets you throw on the right track to solve the problem - impatience comes back with more information.

  Best regards

• 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.