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.

Similar Messages

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!

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...

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".

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.

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!"

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.

How to sync. 50Hz sample rate with 5Hz external digital trigger signal?

Hello there,
I want to sample an analog input at 50Hz, where I further want to use
the 5Hz PPS signal of a GPS receiver as an external trigger/clock
signal for the NI DAQ USB-6008 device. However, I do not want to read
10 samples every 200ms (= 50Hz avg. sample rate), but I want to read 1
sample every 20ms, i.e. with "real" 50Hz.
The "natural" way, it seems would be to use a task that waits for a
rising edge on the ext. trigger input, then reads 10 samples, then
waits for the next trigger edge, and so on.
However, it seems that with this approach I cannot read the data from
the DAQ with real 50Hz, further it seems that every 10 samples the task
needs to be restarted, which takes longer than the 20ms (corresponding
to the 50Hz target sample rate).
I hope the problem description is clear enough....
Thank you in advance for any advice.
Regards,
Oliver

You have to use the counter together with the analog input to achive this. You could configure the counter to generate 10 pulses with 50 Hz whenever the counter sees a rising edge on the inpput of the counter. The output of the counter would be used as the external clock for the analog input.

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.
-Alan A.
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
GB Ram
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

Audigy 4 pro vs. cubase sx sample rate - fight to the de

helloI recently bought an audigy 4 pro soundcard, heard it was probably the best for home recording etc.
I have encountered some problems... I have a load of backing tracks stored in cubase sx 2.2, but now when i open them up it shows the message "sample rate could not be set. This may be due to the sample clock being set to external sync."The files now run at 48kHz instead of 44.kHz, making them jittery, out of time, or chipmonk like. I've read through various messages around similar problems, i've tried everything & nothing has worked...in cubase - project/project set up/sample rate = 48kHz and can not be changed.
in cubase - device set up/vst multitrack/asio driver = creative asio + clock source = internalit only offers 5 asio drivers, asio direct x full duplex, asio mulitmedia, creative asio, SB audigy4 asio 24/96 [a400] and SB audigy4asio [a400] i do not get the offer of asio4all.While i have also tried going through the control panel...controlpanel/audio control panel/device settings/digital out samplerate and setting it to 44.kHz this doesn't seem to change anything. I have no problem with recording new songs at a higher sample rate, but has audigy 4 pro rendered all my old songs useless?please help!!!!!!!!

Doogs wrote:
helloI recently bought an audigy 4 pro soundcard, heard it was probably the best for home recording etc. I have encountered some problems... I have a load of backing tracks stored in cubase sx 2.2, but now when i open them up it shows the message "sample rate could not be set. This may be due to the sample clock being set to external sync."The files now run at 48kHz instead of 44.kHz, making them jittery, out of time, or chipmonk like. I've read through various messages around similar problems, i've tried everything & nothing has worked...in cubase - project/project set up/sample rate = 48kHz and can not be changed.in cubase - device set up/vst multitrack/asio driver = creative asio + clock source = internalit only offers 5 asio drivers, asio direct x full duplex, asio mulitmedia, creative asio, SB audigy4 asio 24/96 [a400] and SB audigy4asio [a400] i do not get the offer of asio4all.While i have also tried going through the control panel...controlpanel/audio control panel/device settings/digital out samplerate and setting it to 44.kHz this doesn't seem to change anything. I have no problem with recording new songs at a higher sample rate, but has audigy 4 pro rendered all my old songs useless?please help!!!!!!!!
If you RTFM, you'll find out, Audigy 4 is locked into 6-bit/48kHz and 24-bit/96kHz resolutions when ASIO driver is in use.
I suppose, you still can (if not saved @ 48kHz) load your projects into SX @ 44. kHz, by selecting MME drivers instead of any ASIO.
Perhaps installing Asio4All gets it popped into that list you have there.
There are also tools to convert from 48-->44., but the source has to be as wave format.
Here is one freeware SRC tool @ http://www.voxengo.com/product/r8brain/.
.jtp

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

How does Core Audio set sample rate?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

Well, here is more information:
- I reinstalled QuickTime 7 and did many other things (trashed preferences,etc)
- I set the audio hardware (Pro Tools/HD Native card) to 48k
- I checked the Digi CoreAudio Manager and it said "Connected at 44.1K", even though the hardware was set to 48K
- I played the problem videos, and all are now in sync
- I check the Digi CoreAudio Manager and it still says 44.1K
- I checked the hardware and it is now set to 44.1K
I know that the videos are 48K, because I created them in Final Cut Pro X.
Does this mean that QuickTime 7 always converts videos to 44.1K, regardless of their sample rate?

How can I find out the audio sampling rate of BetacamSP tape?

Hi guys
I'm trying to digitize BetacamSP tape. But I'm afraid if I might choose wrong setting...
This tape is from very long time ago so we don't know which audio sampling rate we recorded with..
How can I find out the audio sampling rate of this BetacamSP tape?
Thanks:)

The sampling rate is set by the Sony DVMCDA2 you are using, when the conversion is made from the analog input to the digital (DV) output. You should be outputting standard DV which is 16bit 48khz audio.
Assuming you are in the US, your Easy Setup for FCP should be DV-NTSC, and then open the Log and Capture Pane and set the Capture Settings Device Control to Non-Controlable Device and you should be good to go.
You will have to roll the deck manually and start and end your capture manually.
You can download a user manual for the DVMCDA2 by clicking here.
MtD

Low sample-rate measurements on the PCI-6115 DAQ card

I need to measure an analog signal at a sampling rate of a few tens to hundreds of Hz in sync with the rising edge of an external clock. I have a PCI-6115 DAQ card w/ Labview 6.1 and NI-DAQ 6.9.2. The PCI-6115 is a high speed card and has a minimum sample rate of 10 KS/s. Is there any way of implementing a low sampling rate measurement using the PCI-6115 in sync with an external clock?
Thanks in advance.

Kuldeep,
It is possible to do what you are describing above (in fact I don't think an external clock is required to do this), however, bear in mind that the reason for this minimum sampling frequency is due to the ADCs used on this high speed board. The ADCs used are pipelined ADCs, meaning that when a signal is digitized, it is digitized in distinct stages within the ADC (in the case of the 6115, I think there are 3 stages involved). Data is moved from one stage of the ADC to the next each time a sample clock pulse is recieved. If too much time elapses between these clock edges, the signal to be digitized can actually 'leak' off of the ADC. This can result in improper digitization, which can lead to less accurate measurements. So, while it is possible to mak
e the device sample below it's minimum rate, it may be advisable to sample faster than the rates required by your application, and either average multiple data points per measurement, or throw away extra points taken.
I hope this helps,
Dan

Digital sample rate

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