Anyway to force a Digital Input sample rate?

I've tried three different A/D converters and all fail on the Mac Pro, but work with other equipment. All of the A/D converters have a fixed 48k output sample rate. I can set a 48k sample rate on the Mac and the sound's great. Unfortunately, it only holds this setting for a few seconds, then snaps back to a 44.1k sample rate and says 'locked'. The audio is then slow and distorted.. like playing a 45 rrmp record at 33 1/3 rpm. My question is, is there any way I can stop OSX from falling back to 44.1k; i.e. force it to hold the 48k sample rate?

Got an updated Core Audio driver from Digidesign - that fixed it.

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How to force digital input sample rate?

Does anyone know how to force a Digital Input sample rate and/or tell Core Audio which conversion clock to use... internal or external? Apparently both things can be done from Pro Tools, but I don't have it. My problem is locking onto an A/D converter's 48K Toslink output. I can set 48K in 'Audio - Midi' setup and the audio is perfect, but a few seconds later Core Audio returns to the default 44.1K. I need to force it to stay at 48K, or clock on the signal rather than internal clock.

Braver,
Let me clarify a few things. First, the 6733 supports correlated digital I/O. This means that the digital operation does not have its own clock. You can correlate the digital operation with the AO clock, which will allow a digital operation up to the max AO rate. You can generate a pulse train with an onboard counter and use that as your digital clock source. Also, you can use a completely external clock source to achieve the max digital rate of 10MHz. The knowledge base, linked below, points you to a couple of examples that ship with LabVIEW. Be sure and read the documentation associated with each example.
Knowlege Base
As for
the example CDI Single Point clocked by Counter (E).vi, notice how the output of...

Question on supported audio input sampling rate

Hi,
I have USB capture device which supports HDMI input and I wish to use it to connect to my digicam and stream it out with FMLE. While I can see that the video input is working properly, the audio is not working. The error message "Problem with audio capture device - Please verify that the audio device is working correctly and is not already in use." always appear when I try to select the device's audio device. Using the graphedit, I can see that the capture device's audio output pin only supports stereo 16-bit 48 kHz PCM output. As such, can I confirm that that FMLE only support 44.1kHz (as well as other 44.1kHz divisible) sampling rate?
Thanks
John

After researching and playing with the system, I found a solution. Uninstalling the "Realtek High Definition Audio Driver" under the Control Panel -> Hardware and Sound enables sampling rate up to 384KHz instead of 48KHz.
If there are other solutions without uninstalling the HD audio driver, please post. Thanks
Eddie

Sample rate for digital sampling (cDAQ-9172 & NI 9401)

Hi!
I have a cDAQ-9172 with a NI 9401 C-series module (digital). I would like to sample the digital inputs with a sample rate of e.g. 400 kHz or 200 kHz. My problem is that I can only select a the 100kHzTimebase clock, and therefore only get a 100 kHz sample rate. The 20MHzTimebase clock is too fast, since it gives me a sample rate of 20 MHz). Is it possible to get a user defined sample rate of e.g. 200 kHz, by e.g. dividing down the 20MHzTimebase clock?
Solved!
Go to Solution.

The cDAQ-9172 chassis does not have an internal timing engine for digital input however you can use one of the onboard counters to generate your clock. Set your pulse train generation counter to be one of the internal counters, such as "cDAQ1/_ctr0" and your digital input sample clock source to be /cDAQ1/Ctr0InternalOutput".

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!"

Can't change sample rate for digital input on Mac Pro

Hello all,
on my Mac Pro with 10.4.10 I can't change the sample rate for the digital input. Whenever I choose 48000 or 96000 Hz it returns to 44100 Hz after a few seconds. Feeding a 24 bit/96000 Hz signal from an external ADC into the optical input doesn't help. No input signal is available for digital recording software such as Sound Studio or Cubase. After changing the sample rate in audio midi configuration to 96000 Hz the sound can be heard for a few seconds but when the setup returns to 44100 automatically the signal is, of course, lost.
I've deleted all relevant preferences and restarted with resetting paramter RAM. Still the same. With external hardware such as M-Audio Firewire equipment setting the sample rates works properly.
Thanks to all for helpful clues.

Hi,
when E&M signaling is configured on digital interface like the VWIC is, 2 or 4 wires operation is not applicable because there are no wires at all, and reported only for compatibility with the analog E&M card.
Consequently, you cannot configure that and it will not make any difference to effect of the connection.
Please rate post if it helps!

"External sample clock" and "Rate" for digital input acquisition

Dear all,
I want to acquire digital input (21 bits with external clock = 50 kHz) with a PCIe-6343 NI board. Using the DAQ assistant under Labview, I selected the advanced timing with the sample clock time parametrized as External. However, it is also possible to select the Rate of the acquisition. In my case, i want to get the data at the rising edge of the external clock signal, so at a frequency of 50 kHz. How can I do that ? I just need to put a Rate of 50 kHz ?
thanks for your help.
Cedric

Cedric,
dddsdsds wrote:
[...]In my case, i want to get the data at the rising edge of the external clock signal, so at a frequency of 50 kHz. How can I do that ?[...]
You answered your question already. If you want to use an external clock, you have to configure the timing source of your task to be external. In order of proper buffer configuration, you should enter 50kHz as rate in addition to the external configuration, but this will not influence the speed of the acquisition (since it is "clocked" externally!)
hope this helps,
Norbert
CEO: What exactly is stopping us from doing this?
Expert: Geometry
Marketing Manager: Just ignore it.

How can you tell how fast you are sampling when using digital inputs to the PCI6014

i am currently sampling information via th edigital inputs of teh PCI6014. However in order to perform FFT and even to use the "Amplitude and Phase Spectrum vi" i need to know my sampling rate so that i can wire this constant to dt.i have attached the Vi that i created as well as some sample data. can you please help me to configure it such that i obtain a correct display of amplitude and phase. thanks
Attachments:
DISPLAY3 ‏40 KB
proper_sampling_at_1Hz.txt ‏1 KB

Hello,
Thanks for contacting National Instruments.
I found some information that I feel will help you to calculate the sampling rate, so that you'll be able to use this constant in your VI. Please see the link below.
http://digital.ni.com/public.nsf/websearch/5782F1B396474BAF86256A1D00572D6E?OpenDocument
I hope this helps you. Please let me know if you need any further support. Have a great day!
Regards,
Joe Des Rosier
Applications Engineering

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.

DIgital audio in sample rates, 441 48 96 but no 882... is there a way to ad

When feeding my macbookpro a digital audio signal via optical cable while viewing the audio midi setup utility, it shows that the mbp's internal clock automatically switches to the incoming sample rate. I got it to work with 441,48 and 96k. When I fead it the 882 sample rate it automatically switches back to the mbp's internal clock and reverts to line input instead of digital. So the audio midi setup utility shows when it is in digital input mode the three sample rates 441,48 and 96k in the format pop up window and not 882 so it must be just a software issue and I am wondering if there is a way to add the sample rate to the list. If it can take in 96k it should also be ale to take in 882. So does anyone know how you could ad a sample rate of 882 to the list in the audio midi utility format list?

One added note. The reason that I want to be able to do this is I like to record my audio projects at 882 and if the mac book pro could except the sample rate I could then use it as a 882 digital 2 track recorder. Enabling me to skip hitting the bounce to disk function in pro tools and actually be able to ride a fader as the mix is printing to the macbookpro, where as when you bounce to disk in pro tools you cannot access the mix as it bounces to disk.

Does the SB Live 24-Bit have a digital input, if so what sampling ra

I see the SB Li've 24-Bit has a miniplug coaxial digital I/O. I was wondering, if i were to connect a coaxial output carrying stereo audio to this jack, what sampling rate will it record at (if it is even possible)? Is it 24-bit/96kHz like the audigy series?
Thanks,
Chris

I am fairly confident it CAN record digital stereo audio through its minplug SPDIF jack. Since it supports the Creative Digital I/O Module (which supplies coax and toslink input/output) it seems like it must be capable of acting as an input. Now, anyone know the sampling rate?Message Edited by cmp223 on 04-04-2005 03:0 AM

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.

Different Sampling rates for different channels in Analog Input

Hi,
I would like to acquire data at different sampling rates on different channels say ACH 0, ACH 1 and so on. I have a PCI 6052E board and NI DAQ 6.9.2. Also is it possible to simultaneously perform Analog output on two different channels along with the Analog input? What will be the problems/consequences as far as the system resources are concerned. I am a beginner in this area and would greatly appreciate any help/pointers for my queries.
Thanking you in advance
Deepak

Search the eaxamples that ship with LV.
Theer is one called simultaneous input and output or something like that.
It will get you started.
re: multiple scan rates. This is acoomplished by sampling all channels at the highest rate and throw away the expttra samples you do not need.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

PXI-7831R analog input max sampling rate?

I'm using 5 of the analog inputs on the 7831R and seem to only be able to get a max sampling rate of 10K per channel. Looking at the specs it should do at least 10 times this per channel, also the time for the A/D loop is 228 ticks of the 40MHz clock so this would suggest a higher sampling rate than just 10K. I'm missing something here... Any ideas as to what the deal is?
pete

Hi Pete,
You are right in saying that the 7831R boards should acquire data much faster than 10KHz. Infact you can take a look at the actual specs from the data sheet http://sine.ni.com/nips/cds/view/p/lang/en/nid/14757 as well as the product manual. However, I am concerned that you may be measuring the rate of acquisition on the RT VI (or any host vi) you are running as opposed measuring it in the FPGA VI.
Your time critical loop may be running at 228 ticks of the 40 MHz clock but the data is buffered (at that rate) and has to be then transferred to the communication loop which then sends the data to the host vi (which may be your RT vi). Hence, the rate at which host VI receives the data will be much slower than the FPGA VI is acquiring data.
Hope this helps!
Prashanth

Wrong sampling rate from Sound Input Read VI on Dell Latitude E6510

Hi,
When I
receive data from the sound card with the Sound Input Read VI, the actual
sampling rate is not equal to the sampling rate set in the Sound Input
Configure VI. I estimate the sampling ratio by dividing the number of samples received
by the difference between the first and the last t0 from Sound Input Read. For
example, if the sampling rate is set to 7000 S/s, the actual sampling rate is
around 6900 S/s. If the sampling rate is set to 7050 S/s, the actual sampling
rate is around 7200.
This problem appears when the
code is run on my Dell Latitude E6510 machine. When I run the
same code on my Lenovo ThinkPad, the sampling rate is correct.
The Labview application I
am developing is to be used to acquire data synchronously from several sources,
including the sound card input. It is therefore important that the sampling rate
is correct.
If anybody has any idea
about what may cause this problem, and a way to fix it, I would be grateful.
Ole

It has been a trend the last years that sound card only support some sampling rates, not a range. You should refer to your sound card manual for this kind of information. You should also be aware of that the NI sound interface is junk, and they do care about it either. Use the WaveIO interface instead See the link. But I am quite sure your problem is related to the sound card hardware not some NI driver issue
http://www.zeitnitz.de/Christian/waveio
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

Anyway to force a Digital Input sample rate?

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