NI USB 6211 : sample rate and sample read

Bonjour,
je viens vers vous car je souhaite faire une acquisition continu d'un signal d'une entrée analogique. Sur cette entrée analogique est branchée un accéléromètre et un transducteur qui me renvoie donc une tension (120Mv/g) . J'ai donc réalisé à l'aide d'un exemple labview un programe. Cependant, celui-ci est très très long .. Lorsque je varie la tension d'entrée, cela prend plusieurs secondes avant d'afficher la valeur exact. Une erreur 200279 s'affiche lrosque les paramètres de sample read et sample rate ne correspondent pas... j'ai été voir sur le net sur ce lien :
http://digital.ni.com/public.nsf/allkb/AB7D4CA85967804586257380006F0E62
Mais rien n'y change. je vous laisse mon Vi en pièce jointe... cordialement
Attachments:
Pièces jointes :
Test 1.vi ‏28 KB

Doublon et résolu ici
Valentin
Certified TestStand Architect
Certified LabVIEW Developer
National Instruments France
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Travaux Pratiques d'initiation à LabVIEW et à la mesure
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Usage of DAQ USB-6211 with Sound and Vibration Measurement Suite

Hi,
Could I use DAQ USB-6211 with Sound and Vibration Measurement Suite ? All examples are based on 24-bit devices (usually with IEPE-on) such as NI 9234 or similar.
I don't need IEPE, 16 bit is sufficient for my dynamic range, could leave without simultaneous sampling - so 6211 (that I have) is OK for me.
Actually I'm interested in S&V Toolkit ( or Measurement Suite) as a tool with some specific Signal processing capabilities (for example, short-time FFT etc.)
Thanks in advance

Hi sgabr,
Yes, you can use the 6211 with some of the Sound and Vibration VIs, including most of the FFT functions. I know that's a little vague -- I don't have a list of which functions do or do not support 16-bit devices, but I did test it out on my system this afternoon. If there are specific functions or examples you'd like to know about, let me know, and I will try them out.
You also have the option to download the evaluation version of the Sound and Vibration Measurement Suite. That way, you can test drive it yourself!
Kyle B | Product Support Engineer | ni.com/support

Problème NI usb 6211, sample rate et sample to read

Bonjour,
je viens vers vous car je souhaite faire une acquisition continu d'un signal d'une entrée analogique. Sur cette entrée analogique est branchée un accéléromètre et un transducteur qui me renvoie donc une tension (120Mv/g) . J'ai donc réalisé à l'aide d'un exemple labview un programe. Cependant, celui-ci est très très long .. Lorsque je varie la tension d'entrée, cela prend plusieurs secondes avant d'afficher la valeur exact. Une erreur 200279 s'affiche lrosque les paramètres de sample read et sample rate ne correspondent pas... j'ai été voir sur le net sur ce lien :
http://digital.ni.com/public.nsf/allkb/AB7D4CA85967804586257380006F0E62
Mais rien n'y change. je vous laisse mon Vi en pièce jointe... cordialement
Attachments:
Test 1.vi ‏28 KB

Bonjour Geoff54,
Tu es sur le forum international donc si tu veux que quelqu'un te reponde tu devrais poser ta question en anglais.
Sinon il éxiste un forum francophone : French Forums
L'erreur que tu rencontres vient du fait que le buffer du PC se remplit avec les données que tu acquiers mais tu ne le vide pas assez vite et lorsque le buffer est complet il te renvoie l'erreur -200279.
Pour solutionner celà, il faut que tu viennes vider le buffer plus rapidement.
Tu peux faire celà en augmentant le "sample to read" à plus que 1000 ou bien en laissant la valeur par défaut qui est -1.
Bonne journée,
Valentin
Certified TestStand Architect
Certified LabVIEW Developer
National Instruments France
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Travaux Pratiques d'initiation à LabVIEW et à la mesure
Du 2 au 23 octobre, partout en France

Conflict between the saved data and the sampling rate and samples to read using PXI 6070e

Hello, I am using PXI 6070e to read an analog voltage. I was sampling at 6.6 MHz and the samples to read were 10. So, that means it should sample 10 points every 1.5 um. The x-axis of the graph on the control panel was showing ns and us scale, which I think because of the fast sampling and acquiring data. I use "write to measurement file" block to save the data. However, the data was saved every 0.4 second and as 35 points data at the beginning of each cycle (e.g. 35 points at 0.4 sec and 35 at 0.8 sec, and so on) and there was no data in between. Can anyone help me how there are 35 reading points every cycle? I could not find the relation between the sampling rate and samples to read, to 35 points every 0.4 second!
Another thing, do I need to add a filter after acquiring the data (after the DAQ assistant block)? Is there anti-aliasing filter is built in PXI 6070e?
Thanks for the help in advance,
Alaeddin

I'm not seeing anything that points to this issue. Your DAQ is set to continuous acquire. I'm not sure if this is really what you want because your DAQ buffer will keep overwriting. You probably just want to set to Read N Samples.
I'm not a fan of using the express VIs. And since you are writing to a TDMS file, I would use the Stream to TDMS option in DAQmx. If you use the LabVIEW Example Finder, search for "TDMS Log" for a list of some good examples.
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"current encoder settings for bit rate and sample rate are invalid" message

I have some files I am working with. Details and what happened:
Had some music files where I was doing some trimming/splitting. Files worked fine in itunes.
Open the files in Quicktime Pro 7 to trim them. Exported to aif.
Opened the files with itunes. They open and play just fine.
In Itunes I click"create AAC version" to convert them.
I receive the message "An error occurred while trying to import the file. The current encoder settings for bit rate and sample rate are not valid for this file."
I don't know what to do from here. Suggestions?
Thanks.

Make sure that the AIFFs that you export are set for 16 bit PCM.
Also, assuming that you are using 44.1 kHZ sampling in your AIFFs, make sure that you are not trying to use 48 kHz in the AAC.

In Finder how do I get iTunes to show bit rate and sample rate of an entry?

In Finder how do I get iTunes to show bit rate and sample rate of an entry so I can set my DAC?

TonyEyes wrote:
In Finder how do I get iTunes to show bit rate and sample rate of an entry so I can set my DAC?
Tony, Make sure you are in Songs view. Then View > Show View Options, and check the boxes for "Bit Rate" and "Sample Rate." See picture:

DASYLAB QUERIES on Sampling Rate and Block Size

HELP!!!! I have been dwelling on DASYLAB for a few weeks regarding certain problems faced, yet hasn't come to any conclusion. Hope that someone would be able to help.Lots of thanks!
1. I need to have more data points, thus I increase the sampling rate(SR). When sampling rate is increased, Block size(BS) will increase correspondingly.
For low sampling rate (SR<100Hz) and Block size of 1, the recorded time in dasy and the real experimental time is the same. But problem starts when SR>100Hz for BS=1. I realized that the recorded time in dasylab differs from the real time. To solve the time difference problem, I've decided to use "AUTO" block size.
Qn1: Is there any way to solve the time difference problem for high SR?
Qn2: For Auto Block Size, Is the recorded result in dasylab at one time moment the actual value or has it been overwritten by the value from the previous block when AUTO BS is chosen.
2. I've tried getting the result for both BS=1 and when BS is auto. Regardless of the sampling rate, the values gotten when BS=1 is always larger than that of Auto Block size. Qn1: Which is the actual result of the test?
Qn2: Is there any best combination of the block size and sampling rate that can be used?
Hope someone is able to help me with the above problem.
Thanks-a-million!!!!!
Message Edited by JasTan on 03-24-2008 05:37 AM

Generally, the DASYLab sampling rate to block size ratio should be between 2:1 and 10:1.
If your sample rate is 1000, the block size should be 500 to no smaller than 100.
Very large block sizes that encompass more than 1 second worth of data often cause display delays that frustrate users.
Very small block sizes that have less than 10 ms of data cause DASYLab to bog down.
Sample rate of 100 samples / second and a block size of 1 is going to cause DASYLab to bog down.
There are many factors that contribute to performance, or lack there of - the speed and on-board buffers of the data acquisition device, the speed, memory, and video capabilities of the computer, and the complexity of the worksheet. As a result, we cannot be more specific, other than to provide you with the rule of thumb above, and suggest that you experiment with various settings, as you have done.
Usually the only reason that you want a small block size is for closed loop control applications. My usual advice is that DASYLab control is around 1 to 10 samples/second. Much faster, and delays start to set in. If you need fast, tight control loops, there are better solutions that don't involve Microsoft Windows and DASYLab.
Q1 - without knowing more about your hardware, I cannot answer the question, but, see above. Keep the block size ratio between 2:1 and 10:1.
Q2 - without knowing more about your hardware, and the driver, I'm not sure that I can fully answer the question. In general, the DASYLab driver instructs the DAQ device driver to program the DAQ device to a certain sampling rate and buffer size. The DASYLab driver then retrieves the data from the intermediate buffers, and feeds it to the DASYLab A/D Input module. If the intermediate buffers are too small, or the sample rate exceeds the capability of the built-in buffers on the hardwar, then data might be overwritten. You should have receive warning or error messages from the driver.
Q3 - See above.
It may be that your hardware driver is not configured correctly. What DAQ device, driver, DASYLab version, and operating system are you using? How much memory do you have? How complex is your worksheet? Are you doing control?
Have you contacted your DASYLab reseller for more help? They should know your hardware better than I do.
- cj
Measurement Computing (MCC) has free technical support. Visit www.mccdaq.com and click on the "Support" tab for all support options, including DASYLab.

HT5848 What is the sampling rate and codec for iTunes Radio. Is it lossless encoded?

What is the sample rate and codec for iTunes Radio? Is it lossless encoded?

I have to agree with you. There are several forum discussions on bit rate being as high as 256 kbps but I don't see how it could be more than 96 kbps based on the poor sound quality I'm hearing. I'm comparing it to an internet radio station that is 128 kbps and sounds much better.
Am I missing something?

TS1717 when trying to import cd the following message appears "the current encoder settings for bit rate and sample rate are not valid for this file"?

Trying to import cd when this message appears "the current encoder settings for bit rate and sample rate are not valid for this file". Any suggestions?

Thanks so very much. I chose MP3 and Download and it is working beautifully. Have a wonderful day and it is so nice of you to get right back to me. I'm trying to cut a DVD for my grandson's grad party and I got most of the photos and didn't have any music since I lost everything in a clean install. Could hug you!

Sample rate and / or Bit depth probl

I am in the middle of mastering a tune, but when I come to play the tune in Soundforge 8 I get the error message: One or more playback devices do not support the current Sample rate and / or Bit depth. I am using a Audigy2 Platinum with the ASIO A400 driver and I'm sure it should be A9000, I can't find the driver update for this, does anyone have a link to this, or is it something else I should be looking at?

A400 doesn't have anything to do with a version number. It's related to the ressource allocated to your card.
What kind of source are you trying to play ?

The selected combination of bit rate and sample rate is not allowed.

I've always burned my purchased CDs into iTunes using a AAC 320 kbps and 48.000kHz. Now when I select that option in iTunes 9 I'm getting a "The selected combination of bit rate and sample rate is not allowed." pop up screen. So is it impossible to burn my albums in a 320kbps/48.000 AAC format?. I had no problems in previous versions of iTunes using this setting. I guess I might have to switch over using a MP3 import setting instead.

Audio CDs are encoded with a sample rate of 44.1 kHz. Ripping them at 48 kHz requires resampling and does not improve the quality. Unless you have a special need for 48, use 44.1.

Restore Old LP'S AT 96000 Hz Sample Rate and 24 bit Resolution?

I restore LP vinyl albums using the above Sample Rate and Resolution.
Do I have to convert to 44100 Hz, and 14-bit before adding to my iTunes Library?

Ringmaster wrote:
Thanks for your help!
I guess I had better keep all of them in this format. It might make a difference if I decide to burn CD's; also, all of my back ups will be of the same format.
Do what works for you, but neither of those two things is an advantage. If you burn audio CDs, they will be downsampled to 44.1 kHz anyway. And as far as backups, iTunes does not care if different songs are in different formats, nor does any other music player that I know of.
Unless you plan on further mastering or remixing, or you really just like the higher fidelity, there is no obvious reason not to use 44/16.

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

Sample Rate and "Smart Encoding Adjustments"

Wondering if someone could help me out with this...
Is there a reason to choose a higher sample rate over a lower one when importing? Does it improve the audio quality? Or should I just put it in the auto setting?
Also, what does the "smart encoding adjustments" option mean? (In the "custom" settings for mp3 format)
I'm basically trying to get my music onto my HDD at the top quality possible, so I'm trying to figure all this out.
Thanks.

The info below should give you a start on the concepts. Google can find many more facts, opinions, and misconceptions about Lossy vs. Lossless music formats. Way too much information to be listed here. Do several searches with various keywords.
Song file size is a factor of bit rate and song length. Audio quality is a factor of bit rate and encoding format. AAC and MP3 formats are considered Lossy, as they sample the target music file and reduce the total size with some reduction of audio quality. Lossless files are considered CD replicants as they contain all the digital data on the original audio CD. They can be fairly large in comparison to the traditional Lossy file.
Encoding a music file into a Lossy compression format will strip details from the file. Transcoding from one Lossy compression format to another Lossy format will compound the loss of details from the file. (eg: transcoding a sound file from: AAC to MP3; or MP3 to AAC). The audio degradation becomes more apparent when transcoding files ripped at lower bit rates (less than 192kbps).
When you burn an AAC file to CD and then re-rip the CD as AAC or MP3, the sound you end up listening to will have gone through a lossy compression process twice. Those losses can add up, taking what were only mild or even unnoticeable deviations from the original sound after the first phase of compression and making those deviations much more noticeable and objectionable. This is especially true if you try to take music at a low bit rate like 128 kbps (what Apple uses for iTMS) and try to compress back down to the same low bit rate.
The preferred method is to save all audio "masters" in a Lossless audio format such as Apple Lossless, WAV, AIFF or FLAC (or the original CD), and then transcode directly from the Lossless source file to your preferred Lossy format such as MP3 or AAC. This procedure preserves as much of the original audio signal as possible and prevents the compound loss of audio details from the file.
The generally accepted theory is that AAC/128 sounds as good as, or better than MP3/160 (and possibly even MP3/192). Transcoding your AACs/MP3s will most likely result in noticeable audio quality degradation. But -- test it out for yourself. If you cannot hear the difference, then it may be acceptable. Bear in mind that any improvements &/or upgrades in equipment (iPods, headphones, your ears, etc.) may uncover the additional audio limitations you created at a later date.
See: Choosing an Audio Format

PXI-4462 actual sample rate and delay...

Hi All!
Is the DAQmx properties, which give me actual sample rate and filter delay? Or I must manualy calculate this values, as described in DSA manual?
Jury

Hi Jury,
I am not sure if I fully understand your question, but there are DAQmx properties that do return the actual sample rate, but there is not one for filter delay. If you place a DAQmx Timing property node down, the actual sample rate will be returned by selecting Sample Clock » Rate.
Regards,
Jason D
Applications Engineer
National Instruments

NI USB 6211 : sample rate and sample read

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