MyRIO aggregate sample rate

Hi Forums,
Quick question, the myRIO AI sample rate is 500kS/s aggregate on the MXP connectors (A and B).
My current thoughts are that connecting to inputs on AI0-AI3 on MXP connector A, would give me 125kS/s per channel, therefore ~ 62.5kHz bandwidth (please correct me if I'm wrong).
What I'm trying to work out, is whether the 500kS/s gets split between the total inputs shared between MXP A & MXP B; or is it 500kS/s aggregate per connector? I.e. If I split my inputs to AI0-AI1 on MXP A; and AI0-AI1 on MXP B, would I be able to increase my sample rate per channel?
Kind thanks,
Tori
Tori
Student
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Hey Tori,
Here is a thread on a similar question: http://forums.ni.com/t5/Academic-Hardware-Products-ELVIS/Question-about-myRio-Hardware-ADC-DAC/td-p/...
Any I think my answer in that thread (copied below) should answer your question:
The best way to think about the myRIO hardware is as if it has two ports the MSP and MXP (don't distinguish between A and B on the MXP connectors).
The MSP AI, MXP AI, and Audio In (ie all the AI) share the same ADC.
The MSP AO and Audio Out each have their own DAC but share an SPI line.
The MXP AO has its own DAC and has its own SPI line
Let us know if you have additional questions!
-Sam K
LabVIEW Hacker
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Similar Messages

Labview myRIO Analog data sampling rate

I have been using the myRIO to acquire audio input from a microphone.
In LabView I'm using the Analog Input Express VI to acquire the data.
I wanted to know if there was a way to increase the sampling rate of this data from this VI, or any VI at that?
Solved!
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Hi ykhali,
Try putting the VI into a timed loop and set the loop rate to your sampling rate.
Regards,
Joel I.
Applications Engineer
National Instruments

High sampling rate is not accurate

Dear alls,
Any one know why the time loop is not accurate at defined period at the RealTime Trace Viewer ? In figure of VI, I set the period for the time loop is 250 us, but when I analyze in Real time Trace Viewer, it shows the real period time about 676 us. That means my data acquisition will be sampled in wrong sampling rate.
Could you give me some explane, please? And how can I solve this problem? I tested this code with MyRIO board.
Best Regards,
Kien

crossrulz wrote:
Why do you have a wait in your timed loop? The point of the Timed Loop is to state the loop rate.
??? Bigger problems than that! Why do you have any timing other that the TASK timing? And really, The task timing is most likely the loop rate.
Can you post your actual VI? I suspect something else is happening here that we cannot see (configurations, channels used, etc).
And a snip of the Task needs posting

Optical digital input (S/PDIF) problem (sampling rates)

I have a huge problem with my 2009 iMac's digital input that I didn't run into on my older MacBook running the same OS (10.6.x), and it's driving me nuts!
Basically, I'm trying to record a digital feed from an ADC hooked up between my hand-built tube amps/turntable and my iMac, with the goal being the transfer of some of my LP records to 48/24 LPCM for archiving and editing. The ADC's output rate is fixed at 48000Hz and cannot be changed. My iMac, however, refuses to lock on to the device rate, instead forcing 44.1kHz in the audio/midi setup. In fact, I can't get it to do anything but 44.1kHz on anything I plug in (even my DAC's passthrough) despite the fact that I need to use 48kHz. The audio is correct for a few seconds, then forces back to the wrong rate again. No matter how many times I try, the computer insists on using the wrong sampling rate, which has rendered impossible any hope of using my iMac as an audio recorder.
I've tried setting up an aggregate device, but to no avail. Nothing I do seems to work, and since the problem is with Core Audio, any program used to catch the audio records gibberish. Is there a way to force Core Audio to use 48kHz in the terminal? It's not like the computer can't do it, as I get about 2 or 3 seconds of correct audio when changing the sampling rate manually. I just can't get it to stick!

Seeing the exact same thing here. External digital source is 48kHz, and this Mac Pro Desktop will only stay on that setting very temporarily (for about .10 or .15 seconds) then it reverts back to 44.1kHz. Even if I don't have a signal or even a Toslink cable plugged in to the input.
(I'm surprised you are getting 2 or 3 seconds of correct audio, when I'm getting about 1/10th that before it reverts back to the wrong sample rate. Maybe related to a buffer size option somewhere?)
I have tried every known option to reset or re-configure this, and even to the point of creating a new user, resetting PRAM, resetting SMC, etc. I was going to try upgrading to 10.6, thinking maybe it was a 10.5.x problem, but now I'm not sure that will help.
Had I the choice to set the setting to be 48kHz and lock it in, it should stay on that setting - (even if the chance exists that what I'm getting is not correct.)
More often than not, I know what I'm doing more than Mac OS does, and I want to force it to do what it needs to.
There really needs to be a "do it because I told you so" option for most of the Mac audio configuration system. This automatic "the hardware decided to do something for you" stuff is absolutely horrid for a system that is marketed as "Pro". A real "Pro" system would let me edit the configuration once, with the tool (text editor hopefully) of my choice, rather than attempting (and failing) to automate things.
+1 for another instance of this being "a huge problem" and "it's driving me nuts!"

Audio sample rate does not match (HDcam to dvcam)

I'm trying to import clips and keep getting this message:
"The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape."
Footage was originally shot on HDCAM and transferred to DVCAM elsewhere. Using FCP 5, am importing via firewire from a Sony DSR-11 deck. Using DV NTSC 48kHz Anamorphic as capture settings (though I've tried everything that I thought might possibly work with no success). The audio does not seem to drift over the course of several 5 minute or so clips. Clip settings show audio at 48 kHz (don't know if that's from capture settings or from actual data). Seems to me all audio should be 48 kHz 16 bits, so can't figure out what's going on. I have to export an EDL for the project to be finished in HD. Read some similar threads that ended in December, seemingly without much resolution. My broader concern is why this is happening; my immediate concern is do I need to worry about this right now since the media files will need to be recaptured in HD anyway. Any thoughts?
Thanks

A little more info. I'm having this problem on 4 tapes (from different cameras) that were transferred to DVCAM in a squished format to appear full screen on a 4x3 monitor. Video that was letterboxed and I can bring in with the standard DV NTSC capture settings does not have this problem. Still have the problem if I try to import the clips from the squished video with standard settings. Any thoughts?

Sample rate off the audio input and out devices do not match - what to do?

This is fundamental, I know, but nevertheless I can't find my way around it. I get this error message when trying to recor:
Sample rate off the audio input and out devices do not match
and am asked to do this:
Use the appropriate operating system or audio device control panel to adjust the sample rates of the input and output devices to use the same settingt.
I have defined the sample rate to 44.1/16 bit in accordance with my inbuild soundcardt.
I am trying to record from LineIn.
When running on a M-Audio sound card I don't face any problems.
HP 8560W, sound card IDT/High definition audio Codec
Any suggestions?
Knud
Copenhagen

You're sure you have set BOTH the input and output settings to 44.1 16 bit?
Which version of Windows are you running? There are a number of posts on this forum about how to fully access both the Windows Mixer and the Mixer for your soundcard. Especially, you need to ensure that all "Windows Sounds" are turned OFF.

Separate sampling rate for two different channels for a USB-6009 daq

Hi,
I am using a USB-6009 and incorporating the 'daq assistant' to change the sample rate. I am trying to find a way to set the sampling rate to two unique values for two separate channels. I've tried setting up two daq assistants and adjusting the sample rate different for each channel, though this does not work. Is there any way to set the sample rate high for all channels then reduce the rate for a different channel - or an alternative? I would appreciate any input on this, thank you!
- Anthony
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All tasks on a DAQ board that use a sampling clock must use the same clock. Therefore, you cannot have two tasks on the same DAQ board sample at different rates.
Alternatives would be:
1. to combine all of the channels into a single task and just accept the extra data
2. get an extra DAQ board to use in parallel
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
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How to acquire data from 2 chs of the same DAQ card at different sampling rate

I am using single DAQ card (either 6013 or 6014) in my system i want to acquire data from 2 (or more) channels with following requirements
1. sampling rate of each channel should be independant of each other (say one is 20 Hz and other is 15 kHz)
2. data from all the channels should be acquired simultaneously.
3. coding must be done using DAQmx VIs
I have tried out following things
1. I created separate task for each channel: i found out that two tasks can not run simultaneously even though the channels are different
2. I tried out single task with two channels included in it. and i used 'channels to Read' property to determine from which ch. i want to acquire data: this method works fine if the sampling rates are same. but if i change the sampling rate of one channel it gets reflected in other channels as well.
can somebody help me out to solve this problem.
i will appreciate if somebody can post the sample code as my deadline is approaching
Tushar Jambhekar
[email protected]
Jambhekar Automation Solutions
LabVIEW Consultancy, LabVIEW Training
Rent a LabVIEW Developer, My Blog

Hi Dennis Knutson
Thanks for your suggestion.
Tushar Jambhekar
[email protected]
Jambhekar Automation Solutions
LabVIEW Consultancy, LabVIEW Training
Rent a LabVIEW Developer, My Blog

Multiple sample rate question

I have one project with edits of different files all over the place recorded at 44.1 kHz. I have another project with edits all over the place recorded at 88.2 kHz. I wish to combine all tracks from these two projects. I realize I will need to convert the sample rate of all the 88.2 kHz files, but I don't want to mix down any track. I want all of the edits to be cut at the same places in time, just with the referenced files being at a different sample rate.
I've tried everything. It seems that it is impossible. If I convert the sample rate of the referenced files, then open the project and change it's sample rate, the edits are now cut at differnt locations in time. It seems I have to mix down each 88.2 track, then sample rate convert these to 44.1.
But I need to have all the edited segments as they appear in both projects in the same project. Any ideas?

Don't record at 88.2! Why would you do that?!
I'd back up and archive your current session first as you are probably going to have to do some very destructive messing.
I haven't put this to the test but maybe you could try converting the route files in a third party software like ProTools. You'll get a bunch of prompts when you reopen Logic but this may work. But for me the simplest (albeit time consuming route) would be to convert all of your regions to new original sound files (you can do this from the audio pop down on the arrange page). Then in your audio pool select all unused audio, delete and go through each new file and convert the sample rate. It's a bit of a car crash situation needless to say next time preparation preparation preparation!
Finally when you have been through this kind of process make sure that you are operating at the correct new clock rate before continuing otherwise this stuff still aint gonna work.
Hope this is of help.
Good luck.

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
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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

Sampling Rate- How do I check and adjust the sampling rate on my Laptop?

For the purposes of matching the sampling rates on my laptop to my usb microphone, (recording in garageband) how do I find out what the sampling rates are on my 2009 macbook? Many thanks!

GarageBand '11: Set the audio resolution: http://support.apple.com/kb/PH1873

How can I programmatically determine the capabilities of a card under NI-DAQmx (e.g. max sample rate, number of AI/AO/CTR channels, etc.

Is there a DAQ_Get_Device_Info() equivalent for NIDAQmx? I need to iterate thru all the devices on my system, and build up a list of device capabilities. The system may include M-series and E-series cards.

Attached is a program I've used in the past to determine number of AI channels. It could be easily modified to check for AO or digital or counter. Also, there is a ton of properties that you have access to (i.e. max sample rate, max/min voltage inputs, etc.) that are accessed as properties of the type of channel, or timing properties, as opposed to properties of the board. Check out the DAQmx C Reference Help (Usually at Start>>All Programs>>National Instruments>>NI-DAQ). Expand the NI-DAQmx C Properties, and look at the List of Channel Properties, and Timing Properties, etc.
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Attachments:
Device_Info.vi ‏25 KB

Creative Audigy 2 NX Bit Depth / Sample Rate Prob

This is my first post to this form
Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
Here are my system specs:
HP Pavilion zd 7000
Intel Pentium 4 3.06 GHz
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Windows XP Prof. SP 2
Thnx.
-cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

High sample rate data acquisition using DAQ and saving data continuously. Also I would like to chunck data into a new file in every 32M

Hi:
I am very new to LabView, so I need some help to come up with an idea that can help me save data continuously in real time. Also I don't want the file to be too big, so I would like to crete a new file in every 32 mega bytes, and clear the previous buffer. Now I have this code can save voltage data to TDMS file, and the sample rate is 2m Hz, so the volume of data increase very fast, and my computer only have 2G ram, so the computer will freeze after 10 seconds I start to collect data. I need some advise from you briliant people.
Thanks very much I really appreciate that.
Solved!
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Attachments:
hispeedisplayandstorage.vi ‏33 KB

I am a huge proponent of the Producer/Consumer architecture. But this is the place I advise against it. The DAQmx Configure Logging does all of it for you!
Note: You will want to use a Chart instead of a graph here.
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines
Attachments:
hispeedisplayandstorage_BD.png ‏36 KB

Error message trying to synchronise audio and midi sample rate 42463

I have a Rode Podcaster mic that is terrific. It has been working fine in GB. suddenly I am getting this when I try to record my podcast:
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Has anyone experienced this before? How do I resolve it? I'm still using my Version 6.0.5.
Apple, is this a way of forcing me to upgrade to V10?

Just click the icon on each track that stops it acting as a live input. It looks like a sound wave. Only have them clicked orange when using the track to record! I don't know if that will solve your problem!

MyRIO aggregate sample rate

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