cwdsp. Sine wave is where the sampling rate?
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?
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.
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.
Tags: NI Software
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?
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.
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.
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?
Thanks for the quick response.
Where it picks up the exchange rate between two currencies for this period (month, week, day) when I want to see everything by 1 world currency?
Your help is greatly appreciated.
All Exchange rates are in the W_EXCH_RATE_G table.
Set you what rate you want to use in the DAC, the default is the corporate rate. to change it, you must change the setting of dac
$$ GLOBAL1_RATE_TYPE, $$ GLOBAL2_RATE_TYPE, $GLOBAL3_RATE_TYPE $
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.
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?
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?
Hi, engineers, is anyway in Labview that I can express an alternating voltage (sine wave) in the form of Phaser? Thank you. Please see this link for background information:
I have not used it myself, but try this: http://zone.ni.com/devzone/cda/epd/p/id/2982
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
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!
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.
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.
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,
Sorry I have a very simple problem. But it seems that I am too new to LabVIEW... and / or have no idea. I've tried a few things but nothing worked propperly.
I need to generate a sine wave with the following of the flexible parameters.
f = 0.02 Hz... 10 Hz
Range = 0... 500
Offset of 1500
I would like to see a cursor moving and get off after the sine wave. The parameters should be changed at any time that the generation of the sine wave inside a While loop. If that would work I intend to integrate that in the code I wrote for an Arduino, Makerhub, slavery... The values already referring...
I tried allmost all the generation of singnal live that I could find but nothing has worked. Calendar completely proven on a waiting insid the while...
Frequencies below 1 Hz is the fast cursor again... WTF...
A sinusoidal 1 Hz signal should make the cursor up and down 1 times per second... or am I totally wrong.
Ah... Perhaps another question. If possible, I want to count the period. For example, after 4 times of the sine wave stop all programs.
Any help apprechiated... I work with LabVIEW 2012SP1
First of all, please understand this waveform generating function works. You specify a waveform (amplitude, frequency, etc.) and whenever you call the function, it returns all the wave specified points.
You can set the number of points with the #s of the info of sampling pole control. As you put the function in a loop, each itaration gives you #s number of points (1000 in your case). You always get a sine wave on your cursor because each iteration returns a different set of 1000 points (this is because of the method, the function calculates the waveform).
The easiest way to create a sine wave is using the sinus (mathematics, primary, Tigonometric, sine) function. You must use the iteration of the I of the loop counter so that the entry of the sine function. Note that the entry is in radians.
With a little math, you can easily produce and display a sinusoid at 1 Hz.
I have a question on the treatment of a PIC16F84 output signals. It seems that the simulation of Multisim does not work properly - but before I blame Multisim, I ask the community NOR or software engineers or a solution. Because I'm German, you are invited to continue this thread in German if it is allowed by the rules of the forum. If you need additional information to analyze my problem, I'll be happy to provide.
The circuit itself has to convert "composition by pulse" signals "tone" (DTMF tones). So you can get old, classic phones work on new devices that do not support the "composition of pulse" more.
The circuit is powered by the analog telephone line current loop line. The PIC is provided by a rudimentary voltage regulation and count pulse signals (voltage failures / power interruption on the telephone line). After that the captain means the series of impulses in their equal number (e.g. 3 pulses = number 3). The captain gives finally two signals with different frequencies to generate a DTMF tone (e.g. number 3 here is 697 and 1477Hz). As you can see in my PDF file attached, it works very well.
Now I have to convert the rectangle wave given by the captain to an at least similar to a sine wave form - otherwise the device that receives the DTMF tones won't understand them.
So I connected a low-pass filter at the output of the PIC. Now, expect the rectangle signal to be smooth in a way as the 'e-function' will (loading / discharging a capacitor through a resistor). But the results are very far from that - as you can see I have very strange curves.
When I implemented a frequency generator with the same output signal as the PEAK and the low pass filter even I get curves as expected.
So we can say that the output of the PIC works like a frequency generator in my circuit. But why does the filter not behave as it should?
I've tried a lot of different values for the parameters of my RC-filter and simulation - this does not solve the problem.
It would be nice if someone has any idea how to solve this problem.
The output impedance of the PEAK may be too high. May be that my car 50 output? Try scaling of impedance of the filter. Do the 10000 ohms resistance and capacitor 10 nF.
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