Can I set an arbritrary sampling rate in Biobench (need 60/sec to sync to video)?

Is there a way to set a sampling rate that is not in the drop-down menu in Biobench? Thanks!

Hi George,
BioBench comes with a set of predefined sampling rates. I encourage you to try using LabVIEW to trigger/sync your DAQ with video. The sample freq. in BB are std. for low freq. biomedical signals.
Please let me know if you have any further questions.
Regards,
Morten Jensen
National Instruments

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Can I export project at sample rate 48.000khz & up, to then be Mastered??

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How do you set sequence audio sample rate?

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Yes, I get this error when i am capturing. From reading a previous post about the error, I thought the solution was to check the audio rate of the captured clip in the browser, and then make sure it matched the audio rate of the sequence. Like I said, everywhere you get to change the sequence setting, it SAYS it is 48/16, but yet, whe I scroll over in the browser, it says the audio rate is 48 KHz and the audio format is 32-bit floating point. Am I looking at the right places?
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My VoltageRead.vi ‏30 KB

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Go to Solution.
Attachments:
canopen_block-diagram.JPG ‏26 KB
canopen_error_1ms.JPG ‏16 KB

Hello DirkW,
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What 's the meaning of "sample rate =-1" when to use the CAN init.vi?

ALL,
I saw a kind of use to CAN init.vi where The mode = "output recent" and the sample rate = -1. I check the NI CAN help document, it seems that the sample rate must be large than 0 when the mode = output recent. So,here,why the sample rate can be -1? What does it mean?
Thanks!
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I found that it is different whether to use "-1" or ''0" to sample rate. In CAN init.vi, there is an input parameter filepath which is used to load the ncd file or dbc file. The ncd file can't set the sample rate of CAN message, but dbc file can. When I use sample rate = -1 and dbc file , the period(=1/sample rate) of CAN transmission equals the Sample time defined in dbc file and is not determined by loop time in LabVIEW programm. But when I use sample rate = 0, the period of CAN transmission equals the loop time in LabVIEW programm. So I guess the period of CAN transmission is determined by the Sample time defined in dbc file when use sample rate = -1. Is that true?

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Hello;
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Vianney,
The following links will give you more information on FieldPoint and sampling rates.
http://zone.ni.com/devzone/conceptd.nsf/webmain/2124F6BBD29663D886256CB80054DBD7
http://zone.ni.com/devzone/conceptd.nsf/webmain/A14D3C3314A0E24A86256DBE00664943
http://forums.ni.com/ni/board/message?board.id=110&message.id=455&requireLogin=False
http://forums.ni.com/ni/board/message?board.id=110&message.id=1681&requireLogin=False
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Anyway to force a Digital Input sample rate?

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shyam wrote:
Thanks jutapa. I was indeed working with a creative card(SB A2Zs).
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I suppose it's not possible for ASIO drivers with Creative cards (if the modes are hard locked like the 6/48 and 24/96).
Most Creative cards supports sampling rates of 8, .025, 6, 22.05, 24, 32, 44., 48, 96 (and 92) kHz ... supposeingly, you can use lower rates only with DS/MME/WDM drivers.
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jutapaMessage Edited by jutapa on 0-25-2006 09:03 PM

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Why Audition CC don't show sample rate correctly?

Hi
Using Audition CC on Mac OS X with Mackie 1220, 12-channel premium analog mixer with integrated 24-bit/96kHz Onyx FireWire I/O (Quality is amazing!)
Need to record on mono a horrible old tape recording with voice only to "hero"'s repairing audio and configure the best sample rate for recording.
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Device Class: CoreAudio
Default Input: Onyx Firewire (0025)
Default Output: Onyx Firewire (0025)
Clock Source: Device
I/O Buffer size 512 Samples
Sample Rate: 96000 Hz
Question 1: Expert said do not use more than 48000 Hz for recording voice only and use mono. Humans can't listen more than 20000 Hz... well... My recording on 96000 Hz is better than 48000 Hz and I'm not Superman... and can hear better difference with 96000 Hz recording. Don't know why.
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Question 3: Don't understand the differences of my 24-bit/96kHz Onyx FireWire I/O and 32 bit (Float) of Audition. If I record using 24 Bits (Maximum bit depth of my hardware) why appears 32 bit (Float)? My recording are 24 Bits or 32 Bits?
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Tom

CS6 has a serious issue with saving files correctly. The program is asuming that 48kHz is the maximum you will be using and in my case it saved a 96kHz recording with a 48kHz internal header. The file size is consistent with all my previous 24/96 recordings and it sounds just fine interpreted correctly - but played an octave low in frequency and tempo it really sucks unless you are a Blue Whale..
I can play it "interpreted" as a 192kHz file just fine, and it now sounds 100% right, but I cannot save it correctly. I have yet to find out how to recover because when I use "convert Sample Type" it saves it with the same mistake - the wrong sample rate off by the same ratio again.
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Tom the reason you can tell the difference between sample rates is that your hearing has two dimensions, frequency and timing, I sure hope you can't hear the bats at night but I expect that you can tell a good drummer from a bad drummer. In addition there is also the issue (dimension) of bit-depth - instrument decay and acoustic space occupies the time space between notes and if it is not sampled at the right time and place or rounded off to the nearest digital digit you have a problem.
You all know that some humans have perfect pitch and others dont, this gives you some indication how much we each differ. Some people have even learned to use echolocation; the best know cases being blind people because they are not supposed to be able to find their way and know where they are. You can learn underlying concepts of this little discussed aspect of human hearing here and hit the university libraries for the rest. http://en.wikipedia.org/wiki/Human_echolocationhttp://
SteveG's comment are only true in one of the three dimensions - frequency range. We all hear about the same for starters anyway.
The second dimension, timing, is what provides spatial information. Coarse sampling effects this as well. My most recent experience was when transcribing some old casette tapes - when I experimented with FLAC and WMA lossless I found that they din NOT downsample well. Spatial information was significantly deminished be that aucoustic or studio work. Needless to say this surprised me because I was auming that I could drop the samplerate to 48kHz as soon I was done editing and save a lot of drive space. For now I made some excellent 48kHz/24 bit mp3s (320) because they altered the sound the least.
Now you know some additional reasons why older guys like me who have lost the high end and a lot of decibel as well can still tell the difference:we have good sense of timing.
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Can I set an arbritrary sampling rate in Biobench (need 60/sec to sync to video)?

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