Frequency of a Voltage Waveform

I need to analyze the frequency response of a generator set to look for dips, rises and recovery times from load changes. We are acquiring the voltage and current waveforms at 10kHz and recording them directly to a TDMS file. We need to calculate the freguency of the voltage waveform based on the time domain and end up with a file with 10kHz data points that is equivalent to the voltage and current waveforms for analysis. How can I go about doing this in DIAdem?
Many thanks in advance for your help,
Chris Wildmann

Chris
To get frequency information out of voltage/current waveform, would depend on what the voltage or current waveforms looked like. The NI guys may have some other tricks up their sleeves, they usually do.
Options as I see it.
1) Time between Zero crossings and update the frequency from periods measured. (most accurate if is zero crossing.)
2) Look at the slope between points and calculate a very rough frequency estimate. (Very fast but not very accurate)
Would seem that any FFT done would not relate at all to time domain.
Paul

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3-Phase Voltage Waveform to Complex Numbers

I read this topic and and I do not think it worksfor my case. http://forums.ni.com/t5/LabVIEW/How-to-convert-waveform-into-complex-number/m-p/808360/highlight/true#M368430
I would like to simulate 3-Phase voltage Waveform at the sample rate of 1000 sample/s and get their numberic values as complex numbers. I could only get the Y-value but not the phase.
Could someone points out how to get the complex values (1000 values) in the attched example for VA, VB, and VC?

Hi muahang1234,
You could try something like the image I've attached. Basically, you generate the waveform and create another waveform that is 90 degrees out of phase. That way, you can track the Y position of each of the waveforms. The first waveform will be the real position while the other waveform's Y will be the imaginary position, or phase, of the waveform. You may have to convert this into polar form if you want it in that format, but that shouldn't be too hard.
It may be a little inconvenient to do this for 3 waveforms, but that should at least give you an idea of what to do about the issue.
Regards,
Andy F.
Applications Engineering
National Instruments

How can I multiply the frequency of an incoming waveform in Labview?

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Once you have the frequency measurement just wire it to the multiply function.

Ac voltage waveform generation

Hello Everyone,
I want build one Virtual panel which should generate a Sinusoidal wavefrom.
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With the waveform generation functions, this seems pretty trivial. Please attach what you have already done and explain where you are stuck.

Sampling frequency effects on convolving waveforms (need feedback from a signal processing GURU)

I have included my code as version 8.5 for those who have not yet upgraded to 8.6. I have also included screenshots so that you can reproduce the results which I have obtained. I hope that some signal processing guru can shed the light on what I discuss below.
This VI convolves the impulse response signal of a simulated servo motor which is essentially a damped sinusoid with the input stimulus that is a step function. The resulting convolved signal should be IDENTICAL to that of the step response of the motor which is RED in the 1st chart display. As you can see the resulting convolution in the 2nd chart has the same frequency structure but the magnitude is INCORRECT. As you can see in the 2 screenshots the magnitudes differ by a factor of 2 & so does the sampling frequency of the waveform. Why the sampling frequency impacts the magnitude is both puzzling & troubling.
Would appreciate any explanations & corrections so that I have confidence in convolving other input waveforms than just the step function.
Solved!
Go to Solution.
Attachments:
convolution.vi ‏55 KB
screenshot1.pdf ‏54 KB
screenshot2.pdf ‏57 KB

Hi tsunami,
Derivative x(t) VI uses the discrete differentiation to compute the derivative. Consider the simple central differentiation,
y[i] = (x[i+1]-x[i]) / (2*dt)
When you double the sampling rate from 500 Hz to 1k Hz, y[i] also doubles, since dt is only 1 ms now. So this means the second input of Convolution VI doubles the amplitude. Thus the output of Convolution VI also doubles.
To eliminate the effect of the sampling rate, you can divide the convolution result by the sampling rate.
Also please check your VI - I think you use "offset" control for both the sine signal generation and step function. I think "offset" should be 0 in this case. And generate step function separately. I am not sure if this is a problem. Please double check it.
After modifying the code, I can get the identical results.
Attachments:
convolution.vi ‏106 KB

Great discrepancies of frequencies of the arbitrary waveform using niFGEN 5401

Hi all!
I'm using niFGEN PCI 5401 to generate an arbitrary waveform defined previously in the Soft Front Panel.
The reference guide says, that I can produce a frequency up to 1 MHz. Does this value refer to the whole waveform or to each sample of the 16384 samples? And how can you explain, that at last the arbitrary waveform takes about 600 ms for one cycle even though the frequency was set to 1 MHz?!
Thanks in advance
Steffi

Hi Jerry!
I generally understood how DDS works, but I'm bemused why the output frequencies do not correspond to the input frequencies?
1.) Yesterday I made a test with the GIVEN Soft Front Panel and even this does not work as it should: If I want a standard square function of 1 MHz, I get a square function of 2,5 Hz! (The same with 100 kHz = 0,25 Hz and 10 kHz = 0,025 Hz.) Where does this factor come from?
2.) This factor also corresponds more or less to the generated arbitrary function in my program: with 1 MHz I get 1,6 Hz, with 100 kHz = 0,16 Hz ...
3.) I thought changing the ratio between pulse length and length of the whole signal period (it's first a square and then a long time zero) I could generate a short pulse anyway accepting the mentioned f
requency discrepancies. E.g.: One period takes 600 ms (although my input frequency is 1 MHz). If I have a ration of 1 (square) to 100 (whole signal) I should get at least a pulse of 6 ms. But the shortest pulse I ever got was about 20 ms.
The IMPORTANT QUESTION: is there a physical limit for generating pulses (e.g. due to the inertia of the DDS method)?
I hope I could explain my problem more precisely. Thank you anyway!
Steffi

Clipped frequency, but not clipped waveform?

Not sure what this means or how it happened, but as you can see in the image below, the waveform volume does not appear to be clipping, rather is is the frequency that appears to be hard clipping. Who cares, sure, but the audio definatly has that characteristic "clipped" sound, so this is a problem. Incidently, when I perform a Amplitude Statistics scan, is reports only 2 "Possibly Clipped Samples." Running the DeClipper effect under Diagnostics, doesn't detect any clipped audio and thus doesn't repair it either. Any suggestions would be great.

sorry, I didn't think about those numbers being helpful. But, yes it ends right in the middle of 15k Hz and 16k Hz. So probably just something alongs the lines of what you descriped. I suspected that this frequency range is probably to high to even be audible, and since the "clipping" distortion happens when the voice gets louder, the distortion is probably unrelated to the frequency graph. Ok, fine.
But still, the audio is audibly clipped. Why is audition not recognizing this? I know the DeClicker effect hardley perfroms miracles, but I would like to use it. Here is a close up of my waveform, as you can see it's clipping at -3dB, but its not exactly hard clipping, as some times the level goes above -3. How can I make auditon see this as clipped audio?

Change range or sampling rate whilst generating a voltage waveform continuously

Dear Sir/Madam,
I would like to be able to change the sampling rate and range "on-the-fly" (whilst generating a continuous analogue waveform with a DAQmx). How to do this?
Thank you.
Regards
Albert

Hey Albert,
It is possible to perform a simultaneous AI and AO in DAQmx. You simply have to set up the AI task in parallel with the AO task. Since you are doing continuous, this will probably work best in two parallel loops.
To change the sampling rate "on-the-fly", please see the forum post below.
Possible to change sample rate dynamically?
http://forums.ni.com/ni/board/message?board.id=250&message.id=35581&requireLogin=False
Regards,
Chris Delvizis
National Instruments

How to read frequency channel & voltage channels at the same time with different rates?

I am using a DAQCARD 6036E with a SCXI-1000 chassis. I have a SCXI-1100 & a SCXI-1126 modules, along with a SCXI-1124 AO module for voltage output. I am acquiring several voltage input channels with the SCXI-1100 & one frequency channel (frequency signal upto 12000Hz) with the SCXI-1126. I want to configure all the channels in the Measurement & Automation so that I can use the Scale function to setup all the channels. I am using a higher acquisition rate for the frequency channel than the voltage channels. I had the block diagram as attached. When I run the vi, I had the resource problem. The err. msg. is: the specified resource is reserved.. Any suggestions as how to acquire a frequency signal & several voltage signals with this system setup using the DAQ-mx?
Attachments:
Document.rtf ‏4727 KB

Since a common ADC is shared with all your SCXI analog input channels , you will have to sample all channels in a scan list at same sampling rate.
However, you can set up frequency and voltage virtual channels using different instances of "create DAQmx Virtual channel" function in the same task as shown in attached pic, set scan rate and read them using a single DAQmx read
Attachments:
vi.PNG ‏6 KB

I am trying to configure my SCXI-1540 card for excitation voltage and frequency using MAX - but it is not giving me that option when I click on the card and then properties, am I looking in the wrong place?

I am trying to configure my SCXI-1540 w/1315 card for excitation voltage and frequency using MAX - but it is not giving me that option when I click on the card and then properties, am I looking in the wrong place?
I have LabVIEW Express 7.0 Professional - SCXI-1000 Chassiss - along with SCXI-1112, and SCXI 1520/1314 - cable to the PCI-6052E is attached to the 1520 card.
MAX says it recognizes all the cards in the chassiss.

Jeremy, Thanks for the reply.
I tried to set up the system as DAQmx.
I had MAX auto-detect the set-up - and it recognized all the cards - but did not select the 1315 accessory on the 1540 card - I had to do that manually - but in the configuration section I would select the accessory and it would deselect itself. I had to get out of the set-up and then right click on the 1540 card and click on the "Accessories" tab and select the 1315 terminal block that way.
It didn't seem right that the system should behave that way, but I pressed onward.
I then tried to create a new DAQmx Task
When I create new DAQmx task and then click on Analog Input no "Position" Tab appears for me to select, all I get are:
Analog>>Voltage
Analog>>Temperature
Ana
log>>Strain
Analog>>Current
Analog>>Resistance
Analog>>Frequency
Analog>>Custom Voltage with Excitation
Do you think there might be something wrong with either my hardware or perhaps my configuration that is not allowing it to recognize the 1540/1315 in DAQmx?
I had included the system report of what programs I am using, etc. as attached file nireport.txt
After the DAQmx did not work I tried installing everything as Traditional DAQ and it did let me set-up the 1540 Card with Excitation Voltage and Frequency and gain - I then went in to the VI that I had built and tried to use the Express DAQ Assisant to read the channel that I am connected to on the 1315/1540 - when I try to use DAQ Assistant it gives me an error saying that "Device not available in NI-DAQmx. It is possible that the device is being use by traditional NI-DAQ"
Which I guess it is since both configurations are now set-up in MAX.
I would PREFER to use the DAQmx set-up, but I am confused as to WHY it won
't come up with
Analog input>>POSITION>>LVDT
When I try to create a new DAQmx task.
Thanks for the help so far! As you can probably tell I am new at this.
Ryan
Attachments:
nireport.txt ‏2 KB

Can I continuous update a waveform in a 6713 Card? I need to inc and dec the frequency continuously without stoping the generation. Can I do so? how?

Hi.
I`m using labview 7 and a analog output board, the 6713. I`m building a system that generates sine wave forms and I need to change the frequency of the generated waveform without stoping the generation. Can I do so? And how?
Thanks.
Ismail.

You can invoke the "AO Clock Config" VI after invoking the "AO Start" VI to change the update rate without stopping the generation. The "Function Generator" example does exactly this.
Geoffrey Schmit
Fermi National Accelerator Laborary

Frequency, cadence, velocity

Hello
I am measuring the rpm of a bike wheel. I have a standard bicycle
speedometer with magnets on the spokes passing a coil and sending a
voltage pulse. The counter reads anywhere from 5 to 12 pulses per
pass of the magnet so it is not usefull. The analogue wave form
is fairly clear however. How can I measure the frequency of the
voltage pulse?
Thanks

Hi Zowb,
the waveform is produced by a oscillating circuit consisting of the coil and the wires to the coil and perhaps the capacity of the input amplifier. There are several ways to abate the waveform, e.g. by decreasing the quality of the oscillating circuit (I hope I found the right vocabulary since english is not my native language). You can do this with a resistor in row with the coil (little values shoult be good enough). But this also decreases the voltage of the pulse.
In electronics you would solve the problem with a retriggerable monoflop, wich is triggered with the first edge and kept in the triggered state when there are edges before the timeout. So if you want to do this in LV, it depends on the kind of signals you get. If you make an analogue aquisition or a timed polling of digital input you would use the waveform peak detection.vi to find the lokations of pulses. If yopu use a buffered counter then the hardware gives you this locations directly. Then in a loop you throw all pulses with time differences less than the least expected pulse time at highest wheel speed and collect all passed times in a shift register. This should do it.
Greets, Dave
Message Edited by daveTW on 02-28-2007 10:34 AM
Greets, Dave

Frequency sweep with peak frequency

Hi guys, I need some help here. I´m using a NI 9215 to do a frequency sweep from 30 to 90 Hz.
So far so good. Now I need a resulting resonance frequency at highest voltage.
I tried a max peak frequency VI but couldn´t get it to work.
Any idea what to do?
mfg, Andi
ok, here is my sweep.vi for better understanding
Message Edited by andinew on 08-11-2009 04:14 AM
Attachments:
sweep.vi ‏216 KB
create_log_frequencies.vi ‏15 KB

Hi Andi,
one thought first off: looking at your "create_log_frequencies.vi", I realized that it generates linear frequency hops, not logarithmic ones...
But on to your question: to find the frequency with the highest amplitude (this will be your resonace, I guess), you will have to look at the entire signal - your "sweep.vi" code iterates the measurement and interpretation five times per second, independantly of the speed of the sine sweep.
My advice would be to generate one waveform containing the whole sweep run, and writing this sweep block for block with a constant sample rate onto the output. You could then read back the response with blocks of the same size as your output, do the single tone measurement on this block and store the resulting frequency and amplitude in an array preserved trough a shift register or simply into two auto-indexing tunnels. The latter is okay as you don't need a "while loop" in the measurement loop (you know how long the analog output loop runs - why not use this information to configure a "for loop").
After your analog input "for loop" has finished, search your amplitude array for the maximum - then use the found element's index to index the frequencies array and return the corresponding resonance frequency.
A good starting point could be the example I wrote a couple of month ago: it runs continuously the same "sweep" (you'd have to adjust the waveform, I know) and reads back the stimulus and the response (wire the output to your probe and parallely to ai0, your response signal will be connected to ai1). You can then insert tone measurements etc. at the marked section...
When trying to open the example, you will be prompted to find a VI called "_Get Terminal Name....vi" - you can find this VI at the following location: C:\Programme\National Instruments\LabVIEW 8.5\examples\DAQmx\_Utility\_Utility.llb
As an afterthought: the "phase difference" you are calculating between every other sample of the analog measurement array is a bit confusing: if oyu need the phase difference between the original signal and the stimulus response, you will need to wire the stimulus to a second input and compare the phases of the two separate signals...
Best regards,
Sebastian
Message Edited by SFK on 08-12-2009 04:12 PM
Attachments:
Multi-Function-Synch AI-AO mit Signalgeneration.vi ‏91 KB

Ramped waveform generation daqmx

Hello all,
I am trying to generate a finite voltage waveform 500ms long with my USB 6221. I'd like to be able to ramp the amplitude of the waveform up from zero over a few cycles to the full amplitude and then also do the reverse at the end of the waveform. Is this feasible?
Thanks in advance,
Sam

I guess I should have stated that I will be making waveforms with frequencies up to about 30kHz. So far I have been just loading a few cycles into the buffer, and repeatin the generation for a fixed number of samples. With frequencies this high, if I write the entire waveform to the buffer I run into space issues. I guess what I'd need to do is write the increasing cycles into the buffer, then as those are generating write the full amplitude cycles for to the buffer, and repeat until just under 500ms, at which point I'd write the decreasing ramp to the buffer. Can I write new samples to the buffer while generating? And if so is there any issue with dropping samples?
Thanks,
Sam

Measure frequency using 6023E

Hi,
I would like to measure the frequency and magnitude of the input voltage signals around 60HZ. Can I use the onborad counter of 6023E and labview 7.0 to do frequency measurement function per cycle , if yes, how to configure the hardware connection between input source and connector box? then can I use the analog input function to find magnitude at the same time? Welcome any suggestion.
Fred

Hello Fred,
Gus is correct: generally you shouldn�t use the onboard counters on your 6023E to read an analog signal. These counters are made to input and output TTL signals.
Here are a couple of options that may be good for you:
1) Use external signal conditioning to read your input signal and convert the frequency to voltage. For this you could use the SCC-FV01 module or if you have a higher channel count you could use the SCXI-1126. Both of these modules convert an input signal�s frequency to a voltage which can then be read by one of the analog inputs on your 6023E. Here are links to the product pages for these two modules:
SCC-FV01
SCXI-1126
2) Read in the signal on an available analog input from your 6023E and use post-processing to measure magnitude and frequency, as gus suggested. In LabVIEW there are some helpful VIs to analyze acquired waveforms. These VIs are located on the function palette at the following location: All Functions -> Analyze -> Waveform Measurements<\strong>
I hope this helps!
Ken S.
Applications Engineering
National Instruments

Frequency of a Voltage Waveform

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