Supported sampling frequencies

Similar Questions

• AI PXI-6255 sampling frequency

  Hello

  We use a simulated PXI-6255 device and with the LabVIEW example, apparently, that we can achieve a sampling rate of 20 kHz to 50 channels without warnings or errors.

  The spec is the number max of 20 kHz channels must be approximately 38 channels (750000 / 20000).

  
  
  
  
  
  
  
  

  1.25 single channel of MECH. / s multi-channel of 750 ksps / s (aggregation)

  The simulation cannot not representative of the unit or the real PXI-6255 does support a higher rate that we expect?

  Hi jharris66,

  It is not a limitation of the driver to the sampling frequency, so a simulated device reports any error in this case.

  In this example, I think that the limit of 750 kHz comes to the multiplexer.  I don't expect to see the same error messages with the real device, but your accuracy will probably not to spec because not allowing enough time settling between the channels.

  The limit of 1.25 MHz is the maximum ADC sampling frequency.  You can probably exceed this number somewhat, but once you go too far beyond 1.25 MHz you will eventually receive a synchronization from the hardware error when the ADC cannot follow.

  Thus, the specification is still technically 750 kHz for channels multiple acquisitions.  You can probably run at a higher rate than this (even on real hardware), but I wouldn't say that it is supported.

  Best regards

• How to change the DAQmx sampling frequency

  Hello

  I'm trying to: record streaming channels (acceleration 21 and 1 tension) using a DAQmx task, then convert the data to a PDM file. The program files and output to the TDMS file very well. The issue I'm having is that I can't change the sampling frequency. I want to record 500samples/s and I can not get the "real sampling rate" of change of 1651.61samples / s. I am trying to use the clock to do this and I succumbed. I also tried to change the settings of "Timing" in the task without a bit of luck. Here is a screenshot of the .VI I created. I've also attached a copy of the file VI. Any help would be greatly appreciated!

  

  Thank you

  Tony

  You will need to provide the model of your device. You can also look in the sheet/manual to see what the real supported sampling rates. Some devices have limited rates.

• Time interval in the output file is different from the sampling frequency

  Hello

  I'm a load at 1000 Hz. cell sampling I send information to the VI "write to a measurement file. When I opened the .tdms in excel, the time interval between samples is 0,00062 seconds instead of 0.001 seconds. Is it possible to change the time interval seconds a.001?

  Kind regards

  Johan Hendrikse

  It has sampling frequencies different - he has simply not an infinite number of them. I should have been a little more clear as to what I wanted. Have you read the specifications? Sampling frequencies are defined as 50 000 / n where n is an integer between 1 and 31. When you specify a rate, the driver will put it to the nearest that the material support. A little arithmetic shows that 50000/31.00062. How do you rate? Will this work with a DAQ Assistant?

• Average implement properly sampling frequency

  I am writing a program for the acquisition of data that reads the voltage and data current with a 3 phase generator, IE. There are 3 channels of voltage and data 3 (also represented under tension) current data channels come in my USB-6009.

  The waveform is not yet known but probably roughly sinusoidal until about a power of say 1.5 kHz frequency. I need to work on the (reliable) frequency in real time I know the speed of the generator. I use the 'harmonic distortion Analyzer' VI do basically this deep down it's a FFT on the data. I suspect it's too much an overload of treatment because if I increase to more than 1000 Hz sampling frequency, I'm starting to have problems - and if the data max 1,5 KHz I need to sample and analyze at least 3 kHz, of course.

  Yes - the question is, how can I get the sample to say 3-5 kHz and work on the fundamental frequency of a channel in real time without causing any fall?

  All responses are greatly appreciated.

  Dave

  Dave,

  (1) I prefer to put all of the analyses (as TFF) in a separate loop.  The loop of the acquisition is acquisition.  He acquires the data and puts it into a queue or motor of action to be used elsewhere.  In this way the acquisition schedule is not dependent on the time required for analysis or display or save to file...

  (2) lines are built in features that allow the data to be passed to independent parties to the program efficiently and without risk of conditions of competition inherent in global or local variables.  They also have interesting features like the clusters of the error and wait times.  Motor action are the screws in the form of a while loop with a shift register uninitialized to retain this data.  They contain generally structure case to allow the selection of different actions, such as Initialize, writing and reading, or more complex things like subset of return or average accumulate data.  Research on the Forum for the nugget of Ben and many other messages on the subject.

  (3) convert the data type as a whole.  Obviously, you cannot acquire a fraction of a sample, by using a representation of data that supports the fractions is not necessary.  In this case, it is not a big deal, but constraint points can tell you that LV is doing additional work behind the scenes to change data types, maybe not the way you wanted changed them.

  (4) Yes and no.  The FFT should work even if the signal to noise ratio is reduced.  Think of the component continues as being the 'noise' in this calculation.  If you want zero crossings, so it is essential that the offset be withdrawn.  With the current shift zero crossings will be moved off the middle of the sine wave points or the signal may cross any zero if the oofset is greater than the peak of the periodic component value.

  (5) I suspected something like that.  Multiplying is a bit faster than dividing so it is best if the speed is important.  If it is more convenient for the user or the programmer to get dividers, let the programme calculate the reciprocal multipliers.  Do it once, outside the loop where it does not affect the time.

  (6) I missed that you move data between the loops.  Can't do it with the register shift.  See point 2) on the queues or the drivers of the Action.  Notifier could be used also for the stop.  My opinion: the only place where you need a local variable is if you need to write a value to a control, such that when you set a saved a file configuration.

  A lot to learn.

  Another question: in the original post you said that you needed the determination of the frequency in 'real time '.  It is a slippery term.  You use it for anything other than the number of samples to read?  How fast can change the frequency?  What are the consequences of a delay in obtaining the frequency?  How late can tolerate you before that consequences are unacceptable?

  Lynn

• How to acquire with NiScope at different sampling frequencies and lengths Records?

  I need to acquire the data of 2 channels of the NI PXI-5114 map two different sampling frequencies high, at the same time. Also, I put 2 different record length. Is this possible?

  I understand that 'Vertical' settings can be configured for individual chains because the function 'Vertical niScope Configure' has 'channels of entry with which we can assign the desired channel. But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Hi AJ_CS,

  Why do you have to be distinct from sampling frequencies on channels separated from the digitizer even?  What different sampling rate do you want?

  But for horizontal settings such as "min sampling rate" and the record min length, I could not find such an option to specify the channel. Would it not common to both channels?

  You do not have an option to configure the settings of hoirizontal on a channel by channel basis because this concept does not exist in the traditional sense of the use of a scope.  Compatible with the concept of IVI, an oscilloscope traditional benchtop will have only a button or a set of buttons for setting the parameters of synchronization of the unit.  There is therefore no horizontal configuration to separate channels on the scanners NOR.

  I hope that the device is capable of simultaneous sampling and therefore channels can be configured individually to different sampling rate.

  Similar to a traditional benchtop oscilloscpe, the device is capable of simultaneous sampling.  But as mentioned above, the channels can not be configured for different sampling frequencies high.

  However, you can ignore data that you think is not relevant.  For example, if you assign 100MS/s CH0 and CH1 to 50 MS/s, then you throw all other samples.

  Alternatively, you can use separate scanners (a channel on each digitizer) and configure them to taste at different rates.  You can set frequencies of sampling on scanners NOR separated and even synchronize them with TClk.

  -Andrew

• How to acquire the signal to very high sampling frequency

  Hello world

  My name is Luke Ho. I am trying to acquire the signal with Labview (Sthelescope). The signal comes from sensor acoustics, then filters and amplifiers to adapt to ADC rank (0 - 5V). Thus, the maximum frequency of the signal is 40 kHz.

  According to the Nyquist theorem, I sampled at least 80 Khz signal.

  Is there a sampling frequency devices like that? or y at - it another way of better? I used the Arduino before, but it was about 10 kHz.

  I need your advice.
  Thank you all and have a nice day.

  holucbme wrote:

  Thanks for your recommendation

  But is it possible without USB Data Acquisition, it is quite expensive for me.

  This is the cheapest option to NEITHER. I tried to look for options to other companies, but more I found in the same price range, or not answering is not your condition of sample rate.

• Xincrement is not agree with sampling frequency

  I use two PXI-5114 scanners that are being synchronized. I am taking 5 seconds worth of data.  I have = 1e3 sample rate and record length = 5e3. I get only about 1.3 seconds worth of data.  I looked at the actual length of the record and the actual sampling frequency outputs and they said the same thing as the façade made. I looked at and then the info out of the Cluster.vi of Fetch NiScope Multi Wfm and he had the same record length, but the value of Xincrement is 262e-6. This fits with what I get.  Any ideas?

  Thanks in advance.

  Hi AT1,.

  Thanks for posting. What you see, this due to the fact that the minimum sampling rate of the PXI-5114 is 3.81 kech. / s. The digitizer will force any value less than 3.81 kech. / s up to this value, so a record of 5000 samples take 1.31 seconds to acquire. If you are looking to acquire 5 seconds worth of data, I recommend to increase the length of record about 19 050 samples, which should be about 5 seconds worth of data. Let me know if this would work for your application.

  Kind regards

  Joe S.

• limited on PCI-6052E sampling frequency

  I'm trying to measure with two channels with a PC containing a NI PCI-6052E card that is capable of 333KS/s on LabView 8.2. With two channels each channel must be able to 333KS/s / 2 = 166.5KS / s. everytime I try to set the sampling frequency above 94339 Hz I the following warning and the vi no longer works.

  WARNING 200012 occurred DAQmx start Task.vi:1
  Possible reasons:
  Clock speed specified is greater than the maximum ADC conversion rate. ADC invaded errors are likely.

  If I place a similar task in the Measument and Automation Explorer and I can easily make two channels measures to 160kHz without error. Above 166kHz I start getting the same caveat (200012). The code that I use calls first DAQmx create Channel.vi, then DAQmx Timing.vi and finally DAQmx start Task.vi. Everything works fine and I did measuements, but I can't rise above 94339 Hz sampling rate even if it should be possible.

  I just looked and I can't understand the issue here.

  Dear Voyn,

  I thank very you much for your message on our forum. If you create a task for a single channel, you can get a higher sampling rate? Are you able to reproduce the same problem with an example from the finder example? Go to hardware to go end tab selected and output-DAQmx online-online-online-online Acq Cont voltage analog measures & chart voltage-Int Clk.vi

  You can select several channels as well. If it shows the same behavior; which driver DAQmx do you use?

  Best regards

• How to specify the sampling frequency? Must use "measurement & Automation Explorer '?

  I use to measure the input current analog OR cDAQ-9171 (chassis only location USB) and NOR-9207. I have 2010 NOR-installed DAQmx and LabVIEW.

  How can I specify the sampling frequency?

  If I use M & A Explorer to create the task, I can specify the flow rate (Hz) on the Configuration tab-> sync settings.

  For the acquisition of data NOR, it is mandatory to use M & A exploring?

  If I don't want to use M & A Explorer, how can I specify the rate (Hz)?

  Hello

  You can specify the sampling frequency with "DAQmx Timing.vi" located in the function palette DAQmx (read context-sensitive help on how to use wisely).

  You do not have to use M & A exploring (MAX) to create a task.

  A simple and quick way is to use DAQ Assistant (same configuration as in MAX) to configure your measurement.

  Another is to use blocks of DAQmx function to manually build your application code.

  In my experience Assistant DAQ is ideal for simple tasks (one measure), with regard to the more complex measures (synchronized the analog and digital inputs).

  I tend to use function blocks because they give you more freedom about code execution.

  Note: You can also build DAQmx code from a wizard configured DAQ task.

  Best regards

  Matej

• effects of sampling frequency on the convolution of waveforms (need feedback from a GURU of signal processing)

  I have included my code as version 8.5 for those who have not yet upgraded to 8.6.  I have also included some screenshots so that you can replicate the results I got.  I hope that some signal processing guru can shed light on what I mention it further.

  This VI convolves the signal of impulse response of a simulated servomotor which is essentially a damped sine the input pulse which is a step function.  The signal resulting convolved should be IDENTICAL to that of the step response of the engine which is RED on the display 1.  As you can see the convolution that results in table 2 shows the same structure of frequency, but its magnitude is INCORRECT.  As you can see in the catches of 2 screen sizes differ by a factor of 2 & done the sampling frequency of the wave.  Why the sampling frequency, impact on the scale is also very strange & disturbing.

  Would appreciate any corrections & explanations so that I trust the convolution of the other wave forms of entry than just the step function.

  OK, I think I have it working now.  Your premise on the effect of sampling on the derivative is not the issue.  Does it affect what the FREQ of levy is the basis of time of convolution.  As the convolution product is not continuous but discrete the length of the array should be taken into account & the sampling frequency must be consistent with this length of array as well as 1 second corresponds to 1 second.  If sampling freq is 2 kHz & the length of the array is 1000 then to get the correct time base by a factor of 2 must be taken into account.  In addition, to take account of the DC, shift of the ZERO gain factor must be added to the convolved signal to get the correct size.

  Thanks for making me think more deeply.

• PCI-6111 valid sampling frequencies

  We have old card PCI-6111 that we still use. A question was posed regarding the applicable sampling frequency setting.

  According to the manual, this thing must have base clock of 20 MHz and 100 kHz. There also a divisor of 24-bit and (not sure about this) multiplier of 2 bits. I guess that these can be used for the base of appropriate clock frequency scaling.

  What sampling rate can be set on this thing? This is a spec 5 of average size, then that would be the upper limit. Can I put any arbitrary frequency (integer) under this? Or what I need to work explicitly what it's capable of? With a 24-bit divider, this would imply that there is a wide range of frequencies that doesn't "work" on this map.

  What happens when I select a frequency that is "bad"? Round it inflicts, turn down?

  Hello Isaac, it's Paul with engineering Applications to the OR.

  This device offeres 2 at simultaneous sampling of the analog inputs, 5 MS/s per channel - The 2 AI can enjoy up to 5 Mhz per channel each.  You can only have 1 frequency clock for your tasks to HAVE, but this clock frequency can be adjusted from 1 kHz to 5 Mhz, set by you.  Note that most of our maps allow GOT down to 1 HZ, but this ADC is different because it uses a pipeline ADC convereter.  In addition, since this card can taste at the same time, it means that two analog channels will be read at the same exact time (in parallel).  In addition, the value of I is a 12-bit number.

• Synchronization of the inputs and outputs with different sampling frequencies

  I'm relatively new to LabView. I have a NOR-myDAQ, and I am trying to accomplish the following:

  Square wave output 10 kHz, duty cycle 50%.

  Input sampling frequency of 200 kHz, synchronized with the output that I get 20 analog input samples by square wave, and I know what samples align with the high and low output of my square wave.

  So far, I used a counter to create the square wave of 10 kHz, display on a digital output line. I tried to pull the document according to (http://www.ni.com/white-paper/4322/en), but I'm not sure how sample at a different rate than my clock pulse. It seems that this example is intended rather to taste one entry by analog clock pulse. There may be a way to create a faster clock (200 kHz) in the software and use that to synchronize the analog input collection as well as a slower 10 kHz output generation square wave?

  I eventually have to use the analog inputs to obtain data and an analog output to write the data channel, so I need the impetus of the square wave at the exit on a digital PIN.

  How could anyone do this in LabView?

  Hi Eric,.

  All subsystems (, AO, CTR) derive from the STC3 clocks so they don't drift, but in order to align your sample clock HAVE with pulse train that you generate on the counter, you'll want to trigger a task out of the other. I would like to start by a few examples taken from the example Finder > Input and Output material > DAQmx. You can trigger GOT off the train of impulses, start by Gen digital Pulse Train-keep -you probably already use a VI like this to generate 10 k pulse train. AI, start with an example like Acq Cont & chart voltage-Ext Clk - Dig Start.vi-you'll want to use the internal clock so just remove the control of the "Source of the clock" and it uses the internal clock. From there, simply set the "Source of the command" either be the PFI line generates the meter, or ' //Ctr0InternalOutput '-assuming that you are using the counter 0. You'll want to make sure that the start of the task HAVE faced the task of counter I is ready to trigger off the first impulse. They should be aligned at this point.

  For debugging, you can use DAQmx export Signal to export the sample clock - you can then brought the train line and the PFI pulse to make sure that they are aligned.

  Hope this helps,

  Andrew S

• High speed continuous measurement of encoder with sampling frequency of 1 kHz

  I am able at all times the position of a linear encoder using a PCI-6602 counter card, and I need to know how to set up so that the counter rotating at high speed, but the data is inserted into the buffer at a frequency of 1 kHz.  I am able suddenly to a hydraulic cylinder, and I am not concerned about the event recording to high frequency except to the extent where they throw off the number considerably if the equipment does not run fast enough to detect all the impulses of the encoder.

  Now, I think is that the external sample clock signal control (routed internal pulse output counter) time rate whereby the equipment detects the impulses of the encoder and the rate at which it inserts data into the buffer.  With a pulse 100 per inch encoder and a sampling frequency of 1 kHz, the extended final position of the cylinder is turned off by +/-0.15 inches, which is unacceptable.

  I need calculate a speed of this information, so I prefer not to use software timed sampling to control this (it's more difficult programming for other reasons as well - several asynchronous measures).  Any ideas on how to configure the hardware to count faster than the speed at which she inserts counties in the buffer?

  OK, you're clearly on the right track here, so I will focus on some details.

  1. How do you know that the +/-0.15 "differences are * measurement error rather than * error of movement?  Why wouldn't be an accurate measure and a proposal which can vary slightly from the nominal value?

  2. I wonder some all electric noise and defects that may produce false edges.  The fact that the behavior was better by using a sampling rate limited (200 kHz) in the digital inputs may be that some of these flaws were so short that they were never captured.

  I did a ton of work with the Commission to 6602 encoder and I can certainly confirm that count equipment is sensitive to the edges in a few tens of MHz.  (I know its 80 MHz for edge counting, but I think I remember that it can be of the order of 20 to 40 MHz to accommodate the time of signal propagation extra of the quadrature decoding circuit).

  A small point of clarification.  You're talking about the speed at which the meter "works to.  The value of count is a register whose value is changed completely by the circuit, * independent * of the sampling frequency.  If you enjoy with material-clocked County in memory buffer or interrogation of software without buffer not a bit for circuits that increments / decrements the value of the counter register.  (In other words, I am completely convinced that you would get commensurate with position end even if you took only 1 sample software-polled after the end of the move instead of sampling at 1 kHz all the way through.)

  So, if the value of the counter is disabled, it is because the circuit detects producers of County of the edges that shouldn't be there.  Something you can try is to set up digital debounce filter for input lines of the PFI corresponding to the encoder Source inputs and to the.

  -Kevin P.

• Change the value to trigger record data for 1 s sampling frequency of 50 KHz

  Hello

  I have a VI with NI9215 and cDAQ-9178 chassis hardware. The function of the VI came out an instruction to RS232 interface and record 1 second of data every time that the set point is changed.

  The procedure is

  (1) modify the policy to the flow regulator

  (2) wait 2 seconds.

  (3) record of 4 channels for one second to the sampling frequency of 50 KHz.

  At present, the problem is for the first edition of this program, two seconds (rather than) data was saved and corn, the error message 200279.

  

  II. I revised for the second edition of the structure of the producer and the consumer who can increase the speed of the buffer.

  The question is how to configure the trigger to start the backup of data and limit data save for one second whenever the set point value changes.

  

  (1) which edition is best for my application?

  (2) how to trigger the data record?

  (3) how to record only a second of data?

  I also checked this announcement and the elapsed time seems not to work for this case.

  Any help would be greatly appreciated!

  Melody

  Hello

  you have not used properly the nodes property.

  1. replace the case structure in the first loop, with DAQmx features, with a structure of the event. Change the event fires for a worth of control of the setpoint change.

  Edit: as stated in your first post, use the structure of the event, but put inside the while loop.

  2. DO NOT connect error output from the stop command property node. Replace it with a local variable for the stop button.

  Try these and let me know.