HAVE high sampling frequency of trigger
I am using a microscope to tunnel effect from here feeding two voltage signals on my map of acquisition of data USB-6212 (Labview 2013 SP1). A voltage signal is the voltage applied to a piezo in the microscope (AI0). This signal drifting slowly over time and it is noisy. The other voltage signal is a tunnel current is converted into a voltage (AI2) signal (see attached photo):
Ideally, I would like to record the two signals between the lines dotted in a .txt file whenever the event of tip in the top image rises. This should be about every second during a day.
So far, I've written a VI that calculates the moving average of the piezo signal and if the piezo voltage exceeds a certain percentage of the average running it fires a command 'Save as file'. The VI works well for a frequency of 100 Hz, but when I go to 20 kHz, the trigger does not work properly. I am also only watching a lot of a number (in this case, 1000) and if there is a trigger signal in these samples of 1000. So if there's a signal around 0 or 1000 I cut and split into two files that I want to avoid.
I don't have much experience with Labview and probably broke every rule of design in the book.
My question is if there is a smarter way to automatically back up the signal between the same dotted lines at high frequencies of sampling?
I thank very you much in advance!
I rewrote the portions of your VI to improve performance (we hope). No need to three queues. No inquiry unless there is a trigger occurs.
I'm confused by the outbreak that seems to detect the edges of the piezo signal high side, even if the tip is in the negative sense. I modified this logic (eventually) get a threshold top-side of the signal of tunneling.
It is unclear what might happen to 20 kHz. The example shows a constant 1 kHz sampling rate and 1 K samples treated by loop. If the sampling rate is changed to 20 kHz, then the loop will have to run to 20 Hz in order to keep up with the acquired data (@1 K samples per read).
I hope that the joint allows VI (not tested).
Tags: NI Software
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.
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.
I use a structure of producers and customers to measure and record the data. The sampling rate must be as high as 10 kHz. Given that so much data, it takes a long time before the data is saved. At first, I saved the data in an excel sheet. Then I tried to save it in binary, but it still takes a while to complete save. How can I make the time savings a short circuit?
First we will make some corrections to your DAQmx code. Since you are using continuous sampling, do NOT connect the samples per channel. Which is actually limiting your buffer. And there is really no need to define your buffer size either. It is default to very big, so this isn't a problem as long as you read your data quite often.
Now your data connection... You simply create a very wide range while acquiring data. Then, you save the data. It's actually not through the advantage of using the producer and the consumer. You should save your data in the loop of the consumer. It will be elinate need a lot of memory and you save the data to the file while you are buying.
But, in this case, I say that the producer and the consumer is not even necessary. Use the DAQmx Configure registration VI. With this VI, you can tell DAQmx to disseminate all data directly in a PDM file. You don't have to do anything. It is by far the best way to save on your DAQ data.
I have implemented a simple vi which curve data from incoming analog channels, and when you are prompted to save the data, it generates an arrary. When recording is complete, the data is written to a file. I use a local variable between while loops to add data to table. Everything works smoothly. However, when I increase my sampling rate in a few thousand s/s (> 5000 s/s) my final table is not as large as expected. Suggestions to optimize this? I do not undestand where is the neck of the bottle. In the end, I intend to be sampling 8 channels at least 8 000 s/s. The vi is attached in case someone wants to kick something. LabVIEW is fairly new to me.
Search for "producer/consumer" and how a queue can be used to transfer data between the loops.
What is happening is the loop of your is slows and not juggling with incoming data. The local is being written too.
Put a flag on the terminals of your index fo the two loops to see that a single loop is ahead of the other.
And if you do not belive me just wait a few minutes and crossrulz will tell you the same thing.
ENET-9213, WLS-9213 simulation and devices USB-9213, I'm able to correctly get the sampling frequency of I = 1351 samples/s using DAQmxGetDevAIMaxSingleChanRate, which is incidentally on the value of spec'ed of 1200 s/s.
However, when I create a task and add a voltage channel and then HAVE the sampling frequency of the task of query, I get a sampling rate of only 9 samples/s. I tried the same code with other devices and I get the sampling frequency corresponding to the device data sheet, it seems THST this problem is limited to 9213 devices.
Why sampling returned by the task using DAQmxGetSampClkMaxRate rate returns than 9 s/s.
And why the rate of conversion of DAQmxGetAIConvRate only 18 s/s.
I enclose the test code which may be used to reproduce this problem.
When I tried this with a USB-9213 simulation, I used the Sample clock Max Rate, as well as the Rate.vi of AIConvert:Max property node. I could see that for 1 channel, I could spend up to 675.67/s, and I couldn't for 16-channel get79.49S/s (which total is equal to 1271 S/s, which is in the specifications). The multichannel and single channel, I could get an AIConvert Max Rate of 1351.35.
Something that could happen is that you do not explicitly set this device runs in mode high speed. You'll want to set the property Get/Set/rest AI_ADCTimingMode channel at high speed, and you should see much better results in this way.
Something else to note - I use DAQmx 9.0
When I'm using Ableton Live, it allows me to choose 16, 24 or 32-bit, and then I can choose a up to 192000 sample rate. Is this possible in the hearing? I've been through all the preferences and all tabs and I can't find this option. Everything I find is a convert, or the option adjust. But this isn't what I want. I want to mix down this way.
The closest thing is when I go to "Export Audio Mix Down", I found, I can select 32-bit. Then, there is a box for the sampling rate, with all the different values. But it doesn't allow me to change of 44100.
Thanks for the responses guys. Hmm, well the reason why I ask this question is because I am preparing a cd to be sent to a mastering studio. The engineer told me to mix down to 24 bits instead of 16 bit. I asked him why, because everything happens to 16 bit cd anyway. He said even if my session has 16 bit files, if I have them running through the beaches with effects bus vst as Altiverb (which I do), it will improve the sound effects in mixdown, if treated according to a higher bitrate. Would poll better mix so he could work in the mastering session. I assumed that this meant that a higher sampling frequency would also be beneficial. Maybe not? I use a lot of high-quality vst effects, so I want to make sure I get the best possible results. I mean if you have this option and space disk hard isn't a problem, why wouldn't you use it?
It is interesting that the export Audio Mix Down allows me to mix a session from 16-bit to 32-bit files, but if doesn't let me change the sampling frequency. Maybe I'm not understanding these terms exactly. I always thought that, for two, more the number, the higher sound quality.
SteveG - tell you it would be a waste of more than 32 bit 44.1 kHz mixing? What about k of 48 or 96 k? There is not a noticeable difference?
The sample rate only affects the highest frequency that can be solved - it has no impact on the quality of all, once the rate is high enough to fix all that humans can hear. You determine the highest frequency that can be resolved in any case of the sample given by half to get the "Nyquist" frequency - so 44.1 k the highest frequency that can be fully resolved is 22.05 kHz. The human ear extends up to 20 kHz, but it's only in young children - by the time you reach your teenage years he began to fall, especially if you listen to a lot of loud music... then 44.1 k is already greater than any human condition to the response at high frequency. This overall been proved? You bet it has!
To discover the truth about the hype about the sound quality in general and more bits/sample rate are concerned, you should first read this thread of AudioMasters - and the links it contains. It's of high quality academic research, and all attempts to discredit have been completely ransacked.
Your mastering engineer is correct up to a point, but not really for the right reasons - mixtures of 32 bits are usually more specific, simply because the sums do more accurately and scale of signal is significantly improved. You must keep in mind that the only place where you will hear no appreciable difference with one more high bit depth would be in really quiet parts of reverb tails. The best way to do a mix in Audition is to do all this in 32-bit (which is floating point version of 24-bit anyway) and if he really wants a 24 bit file, you can convert the 32-bit mix after that you have to create a copy of 24-bit integer. As much as the 16-bit CD is concerned, if you (or engineer) procrastinate it properly when you do the master of the 16 bits of the final mix of 32-bit CD, then the effective resolution is higher than 16 - bit anyway - there is also a thread of AudioMasters explaining all this too (it's complicated).
What all the above implies, is that if you keep your mix 32 bit 44.1 k files as they are, if you remaster at a higher sample rate, all you have to do is up-conversion files in hearing. You don't win a single thing in doing that, but then again, you won't lose anything either - and person don't will be able to make a difference! This is not a case of "why don't use you it?" - this is really a case of ' why would you? "
Either way, it should be noted that due to some misinformation presented by people who should really have known better long, understanding of most of the people of sampling is completely and totally false. Once more, a search around AudioMasters will give you a better understanding of the present. If I get a chance later, I'll look on all appropriate threads.
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!
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.
In the code in the example attached, I create a task with a single channel of AI.
I get the maximum sampling frequency using DAQmxGetDevAIMaxSingleChanRate (or DAQmxGetDevAIMaxMultiChanRate), both return the same value of 1351 s/s.
When I try to configure the sample calendar using DAQmxCfgSampClkTiming at the maximum sampling frequency clock he does not accept the rate and returns the following error. Note that the error message shows 2 channels, even if only a channel has been added.
Sampling frequency is greater than the maximum sampling frequency for the number of specified channels.
Reduce the sampling frequency or the number of channels. The increase in the conversion rate or
reduce the time of the sample can also mitigate the problem, if you define one of them.
Number of channels: 2
Sampling rate: 1.351351e3
Maximum sampling frequency: 675.675676
Why the device driver thinks I have 2 channels in the task, when a channel has been added?
Please find the code to reproduce this problem attached.
By default, the ENET/WLS/USB-9213 in NOR-DAQmx module has the AI. AutoZeroMode the value of the DAQmx_Val_EverySample property. This causes NOR-DAQmx acquire the channel of the internal path of the unit (_aignd_vs_aignd) on each sample to return more specific measures, even if the operating temperature of the device moves over time. If you need the sampling frequencies higher than this allows, you can call DAQmxSetAIAutoZeroMode(..., DAQmx_Val_Once) (who acquires the formatting string when you start the task) or DAQmxSetAIAutoZeroMode(..., DAQmx_Val_None) (which disables the setting entirely).
Note that for measures by thermocouple with cold junction compensation sensor of the 9213 NOR, NOR-DAQmx acquires channel built-in CJC (_cjtemp) on each sample as well, for the same reason.
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.
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?
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,
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.
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.
I use NI PXI-4462. (204.8kS, input analog 4 / s sampling frequency)
I want to collect data from "load" (channel 1) and "acceleration sensor" (2nd, 3rd, 4th channel).
I also want to save data to a text file.
So I do a front pannel and block diagram. You can see in the attached file.
The program works well in a low sampling rate.
However, when I put up to 204800 s/s sample rate, the program gives me "error-200279".»
I don't know what means this error, and I know why this happened in the high sampling rate.
I want to know how I can fix it.
Is there any problem in my diagram?
Is it possible to save high sampling rate data?
I really want to samplling more than 200000 s/s rate.
I would appreciate if you can help me.
You have provided excellent documentation. So what has happened is that the amount of time it takes to run the other portion of the loop results in a number of samples to be taken is greater than the size of the buffer you provided (I don't know exactly what it is, but it will happen at high frequencies of sampling high) resulting in samples are crushed. You might be best served in this case to take a loop of producer-consumer - have the loop you have acquire the data but then have an additional loop that processes the data in parallel with the acquisition. The data would be shipped from the producer to the consumer via a queue. However, a caveat is that, if you have a queue that is infinitely deep and you start to fall behind, you will find at the sampling frequency, you specify that you will begin to use more and more memory. In this case, you will need to find a way to optimise your calculations or allow acquisition with loss.
I hope this helps. Matt
I have a rectangular steel beam that is affected with a weight of 100 kg and I would look for the modules able to sample the signal correctly.
The Nyquist theorem says that if half of the sampling frequency is higher than the input signal, it will be recorded correctly.
What I think about it before you buy a data acquisition module to find the signal of the rectangular steel beam? I will perform an analysis model by finite elements using the elastic properties or properties of plastic? Is the natural frequency of the associated structure of the input signal?
Some technical assistance is appropriate, determine that the higher frequency component is interesting to your signal. Set your frequency of sampling to twice this value. In addition, to protect data, to build a filter of antisliasing of material it alleviates any energy above the highest frequency of interest.
I hope everyone is doing well!
Well! I am a student in first year of Labview and would like an expert on this VI opinion I did. I'm learning by doing! This VI is to see the effects of sampling at different frequency. I have a LABVIEW 8.5 and uses an express VI to simulate signal, two assistant screws DAQ etc. I also play a little with the number of samples and sampling frequency.
Come to the points that I did not understand!
1. the present VI crashes and I am not able to understand what is the reason?
2. the time scale itself changes as I raise the number of samples, even if I keep the same frequency sampling.
3. in addition, the peak frequency changes with the number of samples! Why?
I hope to have your kind response!
Thanks for your time!
Sorry for the late reply. I made a few changes to your VI and it works very well.
You can start with the choice of the same sample rate and the number of samples of the three waveforms. In this way, all will synchronize initially.
After that, you can try to change the sampling rate and the number of samples for the waveforms. However, you should becarefull when setting very large number of samples. If you have a low sampling rate, say 100 Hz and a high number of samples, say 1000 samples it will take 10 sec to acquire all the samples. If the second DAQ assistant is running at the highest sampling at the same time, you will have an overflow of buffer of data acquisition and it may hang. Of course it is a means to avoid this by implementing different structures in LV, but for the purposes of this test, you should be ok if you just keep this in mind.
I have a simple configuration with a strain of channel 4 OR-9237 amp holds a carrier of series C of WLS - 9163 (wired ethernet mode) - Details probably does not matter.
I used MAX to create a DAQmx task associated with which all four gauges samples. The calendar setting is "Scan Loads" is continuous sampling, 2 k buffer (read samples) and 10 Hz rate. I guess that this task would generate 40 data values per second - 10 for each channel.
I have a simple loop of reading using DAQmx Read.vi that works always (without any stimulation time). Playback is set to read all available data and then pump it into a table.
In the attached example, I also added a few words of debugging to stop the loop after N iterations.
As the loop is programmed with a 0.2 second period, I expect each pass of the loop to read about 8 samples or 2 samples per sensor. Instead, I get hundreds each passage. It's like reading has substituted the sampling frequency specified in the task of the unit. I absolutely need data to be material to the rhythm.
Where have I lost?
I changed your example I selected 'Strain gage' entry analog and then lowered the minimum and maximum thresholds to +-1-2. What happens is that each other in the loop, I 2048 samples or zero samples. The display flashes a whole line and then it clears any other past.
In response to your second post, I understand that the loop cannot run quite right that I select. I think that, but at a sampling frequency of 10 Hz, I have to sleep on the software side for nearly a minute before I built 2 K samples.
I played with the frequency of sampling, assigning to various values from 0.1 to 10000Hz. The behavior is the same until I approach the high rates where available samples remains to 2048-4096 sometimes, the display becomes continuous.
Ahhh, Darn. Yet another search was this link that points to the root of my confusion. The 9237 can taste arbitrary rates using its internal clock. Duoh! I wish that the pilots are smart enough to warn you if there is a discrepancy between the selected sampling rate and capabilities of the device
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