Snd write waveform sample rate (so config.vi) questions

I need to use a "higher" sample rate then appears to be standard. From looking at the documentation, it appears the max rate is 44100. I know the more "modern" hardware will support a higher sample rate then this, so the question is, does Labview?
2nd question, When I attempted to modfiy "so config.vi", to add an additional case, I got an error "Selector Values Wrong Type", even if I select a value that is smaller. So I'm guessing it's not complaining about having an integer that is "out of range"

You may pull the max sample rate from the sound card. I have posted some example code here
http://forums.ni.com/ni/board/message?board.id=170&message.id=387823#M387823. It should work in labview 8.x. The code is very simple and crude.
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Similar Messages

Snd Write Waveform.vi not run on Suse platform

in lvsound.so, VI 'Snd Write Waveform.vi'
Configuration : LabVIEW 7.0
OS : Suse 9.3
Alsa v 1.0.8 and 1.0.9
This function makes LabVIEW to exit!
It works perfectly with RedHad 9.0, XP and MacOSX
The behaviour is different between source and executable :
- source : no trouble before executing this VI
- exe : the main application loads and runs as expected.
As soon as it launches a subpanel calling this subVI it makes LabVIEW to quit.

Hi,
I have reproduced this problem and have been trying to figure it out for
a few hours now. It seems to happen on LabVIEW 7.0 only for SuSE 9.2
and 9.3 (both kernel 2.6 based), when running the SO Config.vi.
I have something for you to try. Please replace the file lvsound.so in
the LabVIEW resource directory (usually /usr/local/lv70/resource) with
the attached version.
On my system, this avoids the crash, which appeared to be caused by a
problem in the system pthread library trying to unwind the stack during
pthread exit. (The sound library uses threads to asynchronously
read/write data to the sound device.)
Regards
David D.
AE - NI France
Attachments:
lvsound.zip ‏7 KB

Maximum audio sample rate and bit depth question

Anyone worked out what the maximum sample rates and bit depths AppleTV can output are?
I'm digitising some old LPs and while I suspect I can get away with 48kHz sample rate and 16 bit depth, I'm not sure about 96kHz sample rate or 24bit resolution.
If I import recordings as AIFFs or WAVs to iTunes it shows the recording parameters in iTunes, but my old Yamaha processor which accepts PCM doesn't show the source data values, though I know it can handle 96kHz 24bit from DVD audio.
It takes no more time recording at any available sample rates or bit depths, so I might as well maximise an album's recording quality for archiving to DVD/posterity as I only want to do each LP once!
If AppleTV downsamples however there wouldn't be much point streaming higher rates.
I wonder how many people out there stream uncompressed audio to AppleTV? With external drives which will hold several hundred uncompressed CD albums is there any good reason not to these days when you are playing back via your hi-fi? (I confess most of my music is in MP3 format just because i haven't got round to ripping again uncompressed for AppleTV).
No doubt there'll be a deluge of comments saying that recording LPs at high quality settings is a waste of time, but some of us still prefer the sound of vinyl over CD...
AC

I guess the answer to this question relies on someone having an external digital amp/decoder/processor that can display the source sample rate and bit depth during playback, together with some suitable 'demo' files.
AC

Is the sample rate half with the snd read waveform.vi?

Hi Guys,
I plan to use the PC soundcard for data acquiring with the standard LV "snd read waveform" vi. During programming I found that the sample rate is the half of the adjusted value. For example when I set
- the format to 8bit mono
- the buffer size to 11025 byte
- and the sample rate to 11025
then the operation takes 2000ms. I would expect 1sec.
Used system: LV7.0, WinXP SP2.
Does anybody know the reason? I have to use 44100Hz sample rate (44100 "individual" sample per secundum) in my application.
Thanks,
Tomi
Attachments:
Panel.jpg ‏18 KB
Block.jpg ‏15 KB

To be more precise, that's the time take for the below mentioned finite cycle
SI config>> SI start>> SI read >> SI clear.
Its this 'entire' process which takes 1997 milli Sec
Now, what you really have to check is wheather your 'SI read' function is living up to the sampling rate you have set!
Check that by putting your 'time elapsed' check for SI Read function alone!
Hope this helps
Regards
Dev

Can I acquire waveform at faster sampling rate using external clock?

I am trying to acquire waveform at a sampling rate of 1MHz. My 6036E card is maxed out at 200KHz. Is there a way to achieve a sampling rate of 1MHz using an external sample clock? Your PFI pin can take external clock as an AI sample clock, right?

Hi,
I remember I asked the similar question before in the NI forum, they reply me:
the max sample rate (scan/s/channel) cannot be greater than your board max sample rate, even you use external scan clcok. So the answer is no for your case.

Adjusting Sampling Rate to write continuous data to excel?

Dear colleagues,
I am reading data from a DAQ from 4 channels and am wanting to write the data continuously to an excel sheet. Problem is, I cannot get the sampling rate and # of samples to appear "constant". instead, labview is only sampling x number of samples in a row and storing those to excel. What I want is this:
Beginning at time t=0, and every 0.2 seconds after that (5 Hz), read the values from all 4 channels and append them to an excel file. What I am using is a 1D to 2D array converter and the Write-to-Spreadsheet function outside my while loop. If I set it to only do 1 sample every 0.2 seconds, then it just stays at t=0, just refreshing the value and thus overwriting my excel data. How can I make it sample continuously from t=0 to t=when STOP button is pressed, and then log all samples into excel? I have attached my sample vi below. Thank you.
Attachments:
working_final.vi ‏547 KB

Hi Mike, Thanks for replying.
The latter of the two methods you mentioned, " you write your values every iteration to the file, therefore you have to place the "Write vi" inside the while loop.", this would be ideal. What sort of conversions need to be done in the while loop before the data is written? In my file, you can see that I am trying to use a converter to go from Dynamic Data to a 1D array. Would this be the appropriate method for doing this? What I am trying to accomplish can't possibly be this complex, I'm sure lots of people require the use of a similar write-to-spreadsheet function.

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

Is it possible to read scalled data in 2D Waveform (DBL) at 2M Hz Sampling rate?

I am using PXI Real time for my acquisition.
NI PXIe-6368
Simultaneous X Series Data Acquisition.
It is supporing 2MHz sampling rate; When I am reading unscaled data, there is no problem.
But when I am reading scaled data, I am getting,
execution get stop or 'Run Out of memory' error.
Can you please help me on this?
I am not able to share the code, right now. I will try to share in personal.
Thank you,
Yogesh Redemptor
Regards,
Yogesh Redemptor

Hi Allan,
If you provided some numbers, we may be able to nail this issue for you. Without them I will just guess.
The error codes you reported indicate that the app is not keeping up with the incoming data. If you are sampling 4 channels at 50kS/s then the hardware is transfering 200 kS/s into its on board FIFO. Page 344 of the NI catalog indicates that the FIFO can hold 2048 samples before it over-flows. The -10846 error indicates that these over-flows are occuring. To prevent this, the app must completely empty the FIFO more frequently. Doing the math this works out to one empty operation every 10ms maximum.
This is a rather demanding rate. This only addresses the -10846. The -10845 error is troubling. The hardware you are working with may not do the j
ob.
Suggestions:
1) make sure your app is reading at least this often.
2) slow down you Acq rate (I know, not an option)
3) DO NOT TRUST ME! Call NI and talk to their hardware support group and ask them;
a) "Can I do continuous acq from X channels at Y rate using a DAQCard 6024E?"
b)If yes, "Could you point me at an example?"
c)If no, "Can I get credit on this device toward a purchase of a device that can?"
I hope this get you started.
Reply to this answer if you have further inquiries,
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

How can I write waveform data from a while loop?

Alright, I have a NI-5122 high speed digitizer that I need to acquire 10,000 waveforms. I am currently using the NI example code "niScope EX Multi Record Fetch More Than Available Memory.vi" to do this and then onto that later. I set the Number of Records to 10,000, min record length to 8192, and the min sample rate to 100M (S/s). On the Fetch vi, I set it to get a 1D Dbl array and then send it to the while loop edge for auto indexing. What I want is to take each waveform and write it to a file next to the previous waveform not append it to the bottom of the previous one. I can do 100 and sometimes 1000 waveforms with my current setup and then use "write measurement to file.vi" to save the data, but at 10,000 it says that the memory is full. I have 256 MB on the NI-5122 and 4G of memory on my host computer. My reasoning is that if the NI-5122 can hold all that data then why can't my host computer. I have tried to fetch using 2D Dbl and transpose that matrix and send the data to "write measurement to file.vi" inside the loop and append the data that way, but when I do this the card does not acquire any data. Which seems like a software problem. I know this is probabaly worded poorly, so just post a response if further explanation is needed.
Thank you.

Hello,
It sound like you might have received a LabVIEW: Memory is full message. Are you also graphing this data as it is being acquired? Are you building an array inside the loop? At the 100 MS/s rate, you are using a lot of memory to acquire, graph, build and transpose array data, and then write to file. If you are saving to file and do not necessary need to see the data, I would suggest taking the graphical indicator out of the code. Please review some of the KnowledgeBases linked below about the LabVIEW: Memory is full.
KB 36QD14V3: Why Do I Receive a "Memory is Full, VI Stopped at Loop Tunnel 0xXXXX" Error?
KB 3ZNDGRS9: LabVIEW "Memory is Full" Error
Samantha
National Instruments
Applications Engineer

Best way to write waveforms to file

I'm reading 7 waveforms from DAQ at 20kHz sample rate and wish to write the data to a binary file [e.g. timeval ch1val ch2val....etc, in format timestamp sgl sgl sgl etc].
What is the easiest way to do this (without using the easy VIs)? The format of the file must be readable by other software - this is why I can't use the easy waveform VIs. The sgl and timestamp formats should be no trouble to read using e.g. MATLAB code.
The other consideration is that this writing to file must not interfere with the data acquisition.

To convert a floating-point value to it's binary equivalent all you have to do is use the flatten to string function. The tricky part is getting the application on the other end to recognize the format. As soon as you start writing binary data you have to start worrying about what processor the end user's machine is using because it makes a difference in how the bitstream is interpretted (bid-endian vs little-endian).
Remember also that to represent LV timestamps (which in the wider world are very nonstandard) you have to use either a double-precision float or a U32 number. In the Microsoft world a slightly more standard representation of time is the number of days since Midnight Jan 1 1900--a value which can be represented in a single-precision float.
T
here are also datalogging examples that demonstrate writing binary files--though those tend to use very low-level binary representations. They aren't even scaled to floats. This would provide the most compact storage since no number would be more than 2 bytes (16-bits), and the fastest too since you aren't taking the time to scale it in LV.
Mike...
Certified Professional Instructor
Certified LabVIEW Architect
LabVIEW Champion
"... after all, He's not a tame lion..."
Be thinking ahead and mark your dance card for NI Week 2015 now: TS 6139 - Object Oriented First Steps

Parallel port sample rate

Hi,
I`m desperately in need of your help.
For my school project, I`m trying to write a labview code in order to acquire air pressure data by using 8-bit analog digital converter via parallel port. Although my signal frequency very low -100 Hz-I think I experience sampling rate problem. I want to put sample rate and frequency data in my code but how can I put it in my code?I `m giving square waveform via function generator which I need to see it on my graph in Labview. But there is something wrong in my signal in waveform chart. You can see screenshot and code attached. . Also there are some noises and amplitude is not stable . How can I fix this?
Would you please have a look at to my code and tell me how can I set sample rate and frequency?
Attachments:
CAN1.VI ‏21 KB

Hi LW-s
There isn't really a way to set the sampling rate for these VIs. You will have to use software timing. The only way to do this is with 'wait' functions inside your loop. You can also implement some logic to smooth/adjust your waveforms. For a square wave the logic is pretty simple. If the input value is above a threshold store it as the max and if it is below the threshold store it as the min. (See the example below). For sine waves this is a bit more difficult and if I were you I would probably try to implement some sort of averaging.
Best Regards,
Chris J
Message Edited by ChrisJ on 08-23-2005 05:31 PM
Attachments:
example.JPG ‏86 KB

Digital sample rate

Is there a way to set the sampling rate for digital inputs?
In am using an 6032E Daq board. I use two VI's: Port Config and Port Read, but there I cannot set the sampling rate.
The only thing I found with rate is for handshaking.
I still have an other question then: in the same VI, I want to read from an Analog input and from a Digital input. I already wrote that VI, and it runs, but is there something I must consider when reading from both analog and digital inputs at a time?

You cannot set a sampling rate for digital operations on an E-eries DAQ board. The port is static, meaning that there is no timing circuitry controlling the port. It is updated or read by software command only.
You configure the port once, but you must call port read each time you want read from the port, and port write each time you want to update the port.

How to store the sampling rate into the header of the data file?

I want to store the data parameters such as sampling rate to the data file. Now I am using the Write Labview Measurement File VI, the data is saved, but like the sampling rate this kind of parameter is not in the file. How can I store this parameter?

If you want the sampling rate to appear in a different location of the header, then it and it's subVIs can be modified to write anyway you want. Right click on the Express VI and slect Open Front Panel. Then you have a VI that can be modified. If you want to use an unmodified Write LabVIEW Measurement File, you could use that, read the whole file back in with one of the file read functions, insert a string with the sampling rate and then write the whole thing back out again. It might be simpler though, to use Write Characters to File to create your own header and then use Write to Spreadsheet File to write the data. There is also the function Export Waveforms to Spreadsheet File on the Waveform>Waveform File I/O. This uses a slightly different format than a .lvm file. It too can be modified if you don't like the default header.

Change sample rate on NI9477 on the fly

I want to generate a continuous square wave on NI9477 and need to change the frequency of the square wave on the fly. Can this be done by changing the output sample rate? Are there any examples how it cane be done?

E-team,
You really can't change your generation rate of your card without stopping the task and then restarting it. However, you can do things like generate at a higher rate and vary the waveform you are generating to change the appearance of the signal. For Instance:
Let's say you are generating at 1 kHz, and your waveform is 10101010, it would appear that you are generating a 500 Hz square wave. You could generate at 2 kHz and write 1100110011001100 and it would also appear as a 500 Hz square wave. Now, if you are generating at 2kHz and you have generated 1100110011001100 and you want to suddenly change the frequency of your output signal you can just start generating 1010101010 at the same 2kHz and it would now appear as though you have changed your output frequency, while in reality you have always been generating at 2 kHz.
As for examples, you could look at a non-regeneration example, where you are always writing new samples to the DAQ card.
Aaron W.
National Instruments
CLA, CTA and CPI

IIR Filtering and response .vi: Butterworth filter magnitude response depends on sampling rate -why?

Hi folks,
I am not expert in filter design, only someone applying them, so please can someone help me with an explanation?
I need to filter very low-frequent signals using a buttherwoth filter 2. or 3. order as bandpass 0.1 to 10 Hz .
Very relevant amplitudes are BELOW 1 Hz, often below 0.5 Hz but there will be as well relevant amplitudes above 5 Hz to be observed.
This is fixed and prescribed for the application.
However, the sampling rate of the measurement system is not prescribed. It may be between say between 30 and 2000 Hz. This will depend on whether the same data set is used for analysing higher frequencies up to 1000 Hz of the same measurement or this is not done by the user and he chooses a lower sampling rate to reduce the file sizes, especially when measuring for longer periods of several weeks.
To compare the 2nd and 3rd order's magnitude response of the filter I used the example IIR Filtering and response .vi:
I was very astonished when I the found that the magnitude response is significantly influenced by the SAMPLING RATE I tell the signal generator in this example vi.
Can you please tell me why - and especially why the 3rd order filter will be worse for the low frequency parts below 1 Hz of the signal. I was told by people experienced with filters that the 3rd oder will distort less the amplitudes which is not at all true for my relevant frequencies below 1 Hz.
In the attached png you see 4 screenshots for 2 or 3 order and sampling rate 300 or 1000 Hz to show you the varying magnitude responses without opening labview.
THANK YOU for your ANSWERS!!!
chris
Solved!
Go to Solution.
Attachments:
butterworth-filter-differences.png ‏285 KB

Hello Lynn,
thanks for the answer. You are right that there are few points "behind" the curve in the graph, see png.
However, this is the filter response which Labview (2009) provides to me directly out of the "IIR Filter for 1 Channel. vi" in the "filter information" output cluster. Where up to now I do not know how to influence it - apart from adjusting the input parameters "IIR filter specifications". OK, I assume I have to gain more knowledge of this. The curve of the magnitude resonse dies not change when I change the number of samples of the input signal of the signal generator, only wehn I change the sampling rate.
I used directly the example vi from Labview with the name indicated in my first post "IIR Filtering and Response.vi".
So I assumed that everybody has it in his/her examples shipped with LV and it is not necessary to post it.
I just adjusted the size of the diagram of magnitude response to see the curves better as you see in the attached vi.
So I did no changes to the vital parts of signal generation and filter of the example. The screenshots are like they come from the example when using the option "one waveform" where I as user assume that this which is behind is quality-controlled by NI.
I was also astonished that the filter magnitude response is different to the one I copied out of graphs 1 year ago - but I unfortunately cannot reconstruct which example I used there...
Thanks for any further comments
chris
Attachments:
IIR Filtering and Response_CH.vi ‏55 KB
butterworth2nd_order_bandpass_0p1to10Hz_mag_response.PNG ‏18 KB

Snd write waveform sample rate (so config.vi) questions

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