Sampling frequency Fs and number of sample #s in a waveform

Hi there,
It is probably a stupid question but I am still going to ask it:
I do not completely understand why in the "Basic Function Generator.vi" there is a distinction between the number of samples #s and the sampling frequency Fs..?
What is the purpose to have more or less samples in the generated waveform then specified by the sampling frequency of the signal?
I hope this question make sense (or am I missing something obvious)?
thanks,
Best,
Renaud

rihns wrote:
I am sending those values to an AO (to a mechanical system that reproduce the sended waveform) and record the signal with an AI. My problem is that when I reduce the number of sample as explained in the previous message, I enhance the number of cycles i.e the time of the signal seen by the AO but it does not seem to apply to the AI... see attached image (Fs = 305000 #s=30500 AO is in white AI is in red)
Why does the #s only affect the time of AO and not the one of AI? (the number of cycle is the same though, only the time base is different)
I think we need to see some code in order to get the full picture of what you are doing. The number of samples of the AO should have nothing to do with the AI.
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What is the effect of output signal with the same sampling frequency but different number of samples?

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The Sampling Rate determines the maximum frequency that you can see (the rate/2 according to Nyquist).
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Hi kdm,
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Hi,
I'm a new labview user and I need some help in trasposing an old traditional NI-DAQ acquisition software in a NI-DAQ-mx one.
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I'm trying to acquire 8 voltage signals from a NI-USB-6341 device. When the Number Of Samples Per Channel I set is below the number of channels (8), the software acquires only a number of channels equal to the number of samples per channel I set.
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This is my code. The problem is the same with or without the code for the scan backlog indexing.
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Attachments:
acq code.jpg ‏119 KB

Sampling sine wave and saving the samples in an array.

helo..
I am new to Labview and i need some help regarding sampling the sine wave.
I have generated the sinewave in labview and now i want to sample it.(do i need an ADC for sampling?) and the samples should be saved in an array.
I dont know how to do this.
Please tell me how to do this or post some related examples..
Thanks in advance

All of the functions that generate a sine wave in software do save all of the samples in an array. You just need to know how to get them. Since you did not mention the function you used to create the sine wave, lets assume it's the Express VI Simulate Signal. On the signal output, wire a Convert From Dynamic Data type function and select the default 1d Array of Scalars. Right click on the output of this and select Create Indicator. You now have the array. If you are using Sine Waveform which outputs a waveform data type, just wire up a Get Waveform Components function and select the item 'Y'.
For learning the basics, please see the free tutorials
Message Edited by Dennis Knutson on 10-28-2008 08:23 PM
Attachments:
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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.
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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).
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-Kevin P.

DAQmxWriteAnalogF64() - array & number of samples

I plan to use DAQmxWriteAnalogF64() to output AO to my device. This is part of the code that generate the AO:
// DAQmx Configure Code
DAQmxErrChk (DAQmxCreateTask ("",&taskHandle) );
DAQmxErrChk (DAQmxCreateAOVoltageChan (taskHandle,"Dev1/ao0","",-10.0,10.0,DAQmx_Val_Volts,NULL) );
DAQmxErrChk (DAQmxCfgSampClkTiming (taskHandle,"",1000.0,DAQmx_Val_Rising,DAQmx_Val_ContSamps,1000) );
DAQmxErrChk (DAQmxRegisterDoneEvent (taskHandle,0,DoneCallback,NULL) );
// DAQmx Write Code
DAQmxErrChk (DAQmxWriteAnalogF64 (taskHandle,1000,0,10.0,DAQmx_Val_GroupByChannel,data,NULL,NULL) );
// DAQmx Start Code
DAQmxErrChk (DAQmxStartTask(taskHandle));
Questions:
1. DAQmxCfgSampClkTiming() specifies a number of 1000 samples with a rate of 1000 samples/second. dataLayout is DAQmx_Val_ContSamps, which is generate samples until task is stopped. Does this mean that the input array that I pass into DAQmxWriteAnalogF64() must have a size of 1000? This is single channel.
2. Say if I use a total of 4 channels, each has 1000 samples, then my input array must have a size of 4000, while my sampling rate remains at 1000 samples/second, correct?

Hello hn_sofec,
In regards to your doubt:
No, the input array that you pass into DAQmxWriteAnalogF64() does not necesarily have to have the same value than the number of the sample rate. The number of samples is just the number of samples that compose the waveform that you want to generate. Then, the sample rate, will determine how fast will you transfer to the output buffer those samples. For example, for an array of 100 samples (lets say that those 100 samples represents a complete period of a sine wave), and configuring a sample rate of 1000 S/s, it would take 0.1 s to generate the first complete period of the sinewave (array of 100 samples). After1 second you would have generated 10 complete periods of the sinewave.
I suggest you to review more examples in order to understand better how the functions work. I share you one here:
"NI-DAQmx: Continuous Multi-Channel Analog Output Example in CVI"
http://www.ni.com/example/25867/en/
Hope this helps. Best regards!
Daniel C.

Configuring IQ / Number of Samples for USRP2920

Hello,
I've been trying to record HF-Signal using the USRP 2920. I got unexpected behavior for some configurations of IQ rate and Number of Samples.
When setting the IQ to 1M as example and Numbers of Samples to 60984, the recorded file remaind empty even after recording over a long period of time. It looks like for a given IQ there is an appropriate Number of sample applicable.
Do someone know the trade off to be made between bo parameters?
thanks for any suggestions
Solved!
Go to Solution.

Hi Bolivar,
There are a few considerations and facts in my knowledge:
The number of samples that you specify as input to the fetch VI is the number of samples that will be fetched by your host from the NI-USRP device per fetch call.
NI-USRP driver (at the host level) will try to acquire these configured number of samples within the timeout specified by you and if it is not able to fetch these from the device you would get a timeout error.
The maximum communication speed/data rate between the device and the host is governed by the communication medium used between the two.(in this case the ethernet connection 1Gbit per sec).
Once you call the initiate VI the NI-USRP device starts capturing data.
So when you configure the timeout, number of samples and the IQ rate of operation following pointers might help.
Is the amount of data that you want to fetch (number of samples * data width) fetchable within the timeout configured ?
Are you fetching out samples quickly enough so that the device buffers do not overflow due to a high IQ rate ?
That being said. let's say we fix IQ rate at 1M
Very small sample sizes will result in a lower actual data rate due to the overhead associated with every fetch call might even cause device buffer overflow.
Very large sample sizes might work if above considerations are made.
A good sample size for your application might be found by making a small benchmark and calculating the actual time taken by the fetch VI call to execute to get a fixed amount of data at various sample sizes and picking the best sample size for your application. I think multiples of 1024 might work better for sample size.
I am not sure if this solves your problem.
Thanks.

Number of sample required

Hi to all the expertise,
In my labview program 6i, I design a scan rate and number of scans to acquire. Now I want to read 100 samples, which mean on my graph, I want to see 100 samples of waveform. How should I configure it, so that I"m able to achieve this.
Kindly advise.
Thanks

Simple as it
If it is a finite acq, keep scan rate as 100 samples /sec and number of samples to read as 100 samples
does this answer your query ?
if not, get back with more details
regards
Dev

Error:the number of sample must be 0 please help.

dear genius.
I searched the same error in this board but I can not understand how can i fix my problem.
I am just a beginner in LV so I need your help. I made AM modulation and demodulation vi .
when I first used "simulated signal" as message signal, noting happened.
but, I changed simulated signal as "acquire sound", and add a play waveform, after that, problem occured.
actually, I can not understand about sample number or sampler rate .
how can i fix my problem?
and do you know some website to exercise digital communication using usrp and labview for trying ?
please let me know your idea.
thanks a lot.
Solved!
Go to Solution.
Attachments:
error.vi ‏164 KB

kimjihwan123 wrote:
thanks kingt of ni
thanks again. but can you tell me the differecne between sample rate and sample per second? (I thoungt it was same)
First of all Knight of NI is my title. Crossrulz is my alias.
Secondly, Sample Rate is the same as Samples Per Second. But you want to make sure your sample rate and number of samples gathered are the same from both sources. I would also turn off the Actual Time (or whatever it is called) in the Simulate Signal so that it just gives you the sine wave instead of using time to generate it.
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Here is what I want to do.I would like to show using an numeric indicator the expired time increase from 0 to 2 seconds. At the end of this two second period, I want a second numeric indicator to start incrementing from 0 to 3 seconds. I am using a USB 9237 DAQ. Sampling frequency of 2k Hz, and samples read of 500. So far, here is what I have put together.
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Sampling frequency Fs and number of sample #s in a waveform

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