Usage of sound level measurement

sir
i am using TES 1352A sound level meter to acquire sound signals.i have installed the sound and vibration toolkit. i am acquring the signal through NI USB 9234 module connected to the sound level meter.Now my problem is how to convert the voltage output from DAQ into decibles. if possible help me with an VI.
thank u

I have used this instrument once. The output is the sound level. It is scaled to a factor x Db per volt output
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

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Sound level measurement

I want to make a programm with labview in order to measure the sound level.
j'aimerai créer un programme faisant office de sonomètre sous labview
Merci pour votre aide

Bonjour Canigoo555,
National Instruments dispose d'un outil pour des applications de son et de vibration : "NI LabVIEW Sound and Vibration Toolset" :
http://sine.ni.com/apps/we/nioc.vp?cid=3122〈=US
C'est un outil que vous devriez installer en plus de LabVIEW et qui vous permettra d'avoir de nouvelles fonctions pour des applications de son et de vibration dans la palette des fonctions.
La fonction qui vous intéresse se trouve dans cette palette et s'appelle : "SVT peak sound Level.vi".
Je vous joins tout de même un exemple LV pour votre application.
Cordialement,
Sanaa TAZI
National Instruments France
Sanaa T.
National Instruments France
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Attachments:
Sound Level .llb ‏1294 KB

Hello All, I want to measure sound level and do octave analysis for Pumps pl suggest a good quality microphone

Hi all, please help me to choose microphone to measure sound level of pumps

I agree with DOS. a 2.8Ghz CPU should OC to 3Ghz easy and should reach 3.2Ghz without much trouble. If I where you I would go with Kingston, Corsair or Geil, probably in that order. If you don't plan on overclocking passed 3Ghz their is no reason to go above 3200 for the RAMM frequency.

Microphone calibration for a sound level meter

Hi I need an application for measure sound level by using the sound and vibration toolkit and severl microphones as input, but I don´t get the same values as a portable Phonic portable meter. I´m using a 60db gain preamplifier but when i use this value in the pregain scale voltaje to EU block I don´t get any value, but if I use -60 it works! is that correct? on the other hand, I´ve seen a calibrate microphone block for the traditional daqmx but I dont understand how to use it and I cant find examples. Do you have one? I have a cDAQ with 9205 A-input card. Also, if I use the PC microphone input to get the data do I have to put an pregain value for the scale voltaje to EU block?
I need dome help please. Thanks a lot.

¡Hi mario_rojas!
I checked the documentation and SVL Calibrate Microphone (DAQmx) VI if you run it with the proper configuration it will measure the sensitivity of the microphone connected to the channel you specify. Can you tell me what version of the Sound and Vibration Toolkit you are using? There are some specific inputs that you need to configure, but I need to know the version to tell you the values you need.
About the -60dB gain, it should work with +60. When I have the version of the toolkit I can check this out for you.
About the question for the PC microphone, you should not have to input a pregain value, since you do not have an intermediate amplifier between the microphone and the daq board, you just need to find the sensor sensitivity. We can use the SVL Calibrate Microphone (DAQmx) VI to find out if we don’t have that data in the specs.
I think we are dealing with the same problem in this forum:
http://forums.ni.com/ni/board/message?board.id=250&message.id=36845
I think it’s better if we continue working in one forum. Please let me know if this is the same issue, and if it is, in what forum would you prefer to keep posting.
Kind greetings,
Tania Lozoya | National Instruments
Mensaje editado por Tania Lozoya

LAFMax Sound Level

We are trying to get LAFMax (the maximum A-weighted noise level measured with a fast time weighting) from recorded sound pressure measurements. We sampled two different ways. One set was taken in one second blocks at 25.6KS/s and the other method was to get a single 20 second block at the same sample rate.
My thought was to run through a loop that picks off 3200 samples each time, gets Leq, and keeps track of the maximum value found until the entire block has been parsed. For the one second sample blocks I would have to use the maximum from all of the 20 blocks.
My problem is that I am very new to LabVIEW programming and have very little time to devote to learning before I need these results. So any pointers would be greatly appreciated.
Thanks
Solved!
Go to Solution.

Hi arathbun,
Just to make sure I understand, you are or have already acquired all of your data. You now want to set up a simple program that will go through all of it and track the max value found from it all. This will ultimately yield the max value of all the data, including all twenty blocks correct?
Also, I believe this is actually going to all be developed in LabVIEW? SignalExpress does not provide a whole lot of data control, it is great for simple signal acquisition but more complex applications it is a bit beyond its capabilities.
Though you are new to LabVIEW it would be worth some time reading through some of the simple tutorials to get you familiar with the environment and tools.
http://www.ni.com/academic/students/learnlabview/
From a high level perspective, it sounds like you need a for loop to run through your sets of data, and use a shift register to store an integer value for the max. Your internal loop would take a new number value and compare it to the current max, and select whichever is bigger as the new max. You could also store your information in an array and use the min/max properties to get similar information and faster results.
Hopefully that will get you pointed in the right direction. If you are indeed using LabVIEW I would recommend that you post in the LabVIEW forums as you will get better support more applicable to your questions.
Regards,
James W.
Applications Engineer
National Instruments

Sound Level Pressure with several channel

Hello,
I met a problem with the VI "SVL Exp Avg Sound Level.vi". It works perfectly if I make a measurement with a single microphone, but with more microphones I have some computation error with discontinuity depending on the buffer size.
The following message is given in Labview Help :
"The SVL Exp Avg Sound Level VI requires a continuous input data stream. An
internal mechanism automatically detects any sampling discontinuity and resets
the filters. "
I suppose that there is a link with my problem, but I have no idea to correct it.
Here is my VI which is composed of a producer/consumer pattern. The consumer loop computes the sound level pressure during the measurement and stores the measured data in a TDMS file.
I am not expert with enqueue and dequeue function, so maybe it is bad programmed and I have some discontinuity in my data stream coming from the dequeue function.
Thanks for your help,
Mathieu

Based on your screenshot, it looks as though your code may be computing the weighted sound level for each channel in a single For Loop. The Weighting VIs and the Sound Level VIs maintain state, so it is not correct to use it in a For Loop to process multiple channels. Instead, wire the entire array of sound pressure channels to the input of the weighting and sound level VIs. The polymorphic VIs will adapt to the input type, and if the input data is continuous, the warning will not be returned.
Interestingly, you may be able to simplify your code if you specify the read order to group measurement types. Here is some code that I was looking at when evaluating performance for some different approaches to grouping waveforms. The Group by measurement type case was the fastest.
Doug
NI Sound and Vibration
Attachments:
performance_read by meas type.vi ‏79 KB
Get Channel Names by Measurement Type.vi ‏23 KB

IEPE Mic Sound Level Error

Hi All,
I have recently purchase the Sound and Vibration tool kit for LV8.6 for making (surprise, surprise) sound pressure and vibration measurements on an enclosure that houses a variable speed pump. I use a cDAQ chassis with the 9233 IEPE module and an ACO mic/pre (model 7052SYS). The mic is IEPE, of course, with a specsheet sensitivity of 20mv/Pa.
I am getting started using the S&V example vi named DAQmx Sound Level Meter. When I set up the AI mic channel to the 20mV/Pa sensitivity, I am still seeing large error in my measurement as compared to calibrated handheld sound meter. In specific, the room ambient noise is ~32.9dBA as measured with the Extech handheld meter. The VI and IEPE mic set-up reports this as ~39dBA. The error when noise is introduce sometimes gets a little closer, but the two do not agree.
What have I overlooked? The IEPE mic is new and has factory cal certs so let's assume this is not the issue.
Thanks for the help!
-JD

JD:
That indeed could be a potential source of the difference between the readings. To know for sure, you would need to do a spectal analysis fo find the energy content across the frequency bands.
If you have a quiet chamber, you could possibly generate a single tone frequency (at multiple freqeuncies within the range of both meters) and then compare the meter readings.
-AK2DM
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"It’s the questions that drive us.”
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Sound level meter -Save Leq

I hope someone can answer me fast as I need to meet a deadline
I made the sample Sound Level Meter DAQmx for Sound and Vibration 2013.
My concern with this is that I need to save the Leq per second after we collect all data.
What I did is attached but I do not know where to attached the write measurement file express.
Please help me...
Attachments:
NI how to save.PNG ‏57 KB

Poke the Data acquired into a Queue. (making this the producer loop)
De-Q the Data in another loop (consumer loop)
Update the file in the consumer loop - this means that File IO overhead will not affect your sampling rate.
Open the File reference before the Consumer Loop starts and close it when complete to minimise File Opening and closing as data is saved (or save the file every n samples/seconds if you are paranoid)
This will get you a faster more flexible architecture - File IO has a much larger overhead than DAQ normally so moving this to a seperate Q and thread allows parallel processing.
James

Relative Level Measurement using PCI-4462

Hi, everyone.
I would like to create a Labview program to test the my radio volume using PCI-4462 and Sound/Vibration Measurement Suite 2011.
Before this, I always use Panasonic VP-7722A to test the radio's volume steps.
There are 31 (0-30) volume steps and I always set the relative level at step 30, at which my reference is 0dB.
Then the volume will be tuned down one step by one step until step 0 for check its attenuation (dB) at each step.
However, I coundn't find the way to perform the same steps using PCI-4462 and Sound/Vibration Measurement Suite 2011 so far.
Kindly need any expert to help and advise on this.
Thanks.

Hi, everyone.
I would like to create a Labview program to test the my radio volume using PCI-4462 and Sound/Vibration Measurement Suite 2011.
Before this, I always use Panasonic VP-7722A to test the radio's volume steps.
There are 31 (0-30) volume steps and I always set the relative level at step 30, at which my reference is 0dB.
Then the volume will be tuned down one step by one step until step 0 for check its attenuation (dB) at each step.
However, I coundn't find the way to perform the same steps using PCI-4462 and Sound/Vibration Measurement Suite 2011 so far.
Kindly need any expert to help and advise on this.
Thanks.

Error -1906 occured at SoundVib_SoundLevel.Lvlib:SVL Leq Sound LeveL (N Ch).vi:3

Hey all, I have this message error when I try to run my vi.
Possible reason,
Sound & Vibration: Multichannel measurements require that each signal has the same sample interval (identical dt or identical dr)

I am generating a analog signal and acquring the same signal. I want to visualise the generated and aquired signal on the sound level meter
as you can see on the attached vi. when I run the vi the error message ( error - 1906 occured at soundVib_SoundLevel.LvlibVL Leq sound level (N Ch).
Attachments:
Signal generation with sound level meter.vi ‏126 KB

HT5517 How can I adjust sound level when streaming radio from my Apple TV through Airplay speakers?

I have a 2nd Gen Apple TV with an airport express set up as airplay speakers. I can stream the new radio through using the Airplay speakers pretty well. The problem is that once I've got it going, I cannot adjust the sound level without laboriously backing out of the radio menu, going back to the top level menu then drilling down into the Airplay settings to raise or lower the speaker's sound level.
Is there an easier way to do this?

go to system preferences > Sound > Output. then change the output from "built-in" to HDMI.
Maher

How do I get a descent sound level and quality when exporting out of Logic ??

Hi guys,
I know this topic's been discussed many times before but I still can't fix my problem
I'm going crazy right now. It's been weeks I've been working on beats on Logic and I think they now sound pretty cool.
So I'm trying to export them out of Logic on mp3,or wav, or aiff but everytime it comes out very low. So when I'm applying different tips (limiter, adaptative limiter and different things I found on the internet... ) the sound level is a bit higher but still the quality is not as good as it is on Logic
So here's my questions : first, is it possible to get the exact same quality as the one you have when listening into logic ?
then how to get the best quality with a normal sound level so when you listening to the mp3 or wave or aiff on itunes just after a normal song you don't have to turn the sound up
Thank you very much for you help !!
Ben

And in addition to what octopi wrote ……
Related Articles:
Mixdown for Mastering - written by Holger Lagerfeldt
http://www.onlinemastering.dk/mastering-faq.html
Loudness when Producing and Mixing - written by Holger Lagerfeldt
http://www.gearslutz.com/board/music-computers/468170-loudness-when-producingmix ing-
tips.html
Overload in Signal Conversion - written by Søren H. Nielsen and Thomas Lund
http://www.tcelectronic.com/media/nielsen_lund_2003_overload.pdf
0 dBFS+ Levels in Digital Mastering - written by Søren H. Nielsen and Thomas Lund
http://www.tcelectronic.com/media/nielsen_lund_2000_0dbfs_le.pdf
Level Control in Digital Mastering - written by Søren H. Nielsen and Thomas Lund
http://www.tcelectronic.com/media/nielsen_lund_1999_level_co.pdf
Cheers

We have a new iPod Classic and would like some help in choosing "Import Settings" (under Peferences) when loading the iPod from the Mac. We want to be sure that we can maintain a constant sound level. Used with Bose sound dock. OS 10.6.8 iTunes 10.6.1.

Using an iPod Classic on a Bose sound dock....when loading the iPod from the Mac, what "Import Settings" (under iTune "Preferences"), should be used to maintain a constant sound level when playing music? Mac OS 10.6.8, iTunes 10.6.1.
There are many Import Settings available and it is somewhat confusing. Any suggestions?

Go to iTunes>Preferences>Playback. Tick "Sound Check."
iTunes: About Sound Check

How do I change the startup sound level on my desktop??

How do I change the startup sound level on my desktop?

use StartNinja ! worked for me.
it works under Mountain Lion and is even retina-ready:
http://www.allvu.com/index.php/sndownloadpage.html

My pc sound level is very low on playback from dvd and most websites even though its at 100 percent

the playback level on dvd and most video streams is very low.(youtube seems ok)This is my second pavillion g7 w/ windows 7 .The first had a defective fan but the sound level was ok and there was a graphic eq section for sound control with other output options.I've yet to find that area on this pc in the month I,ve had it even though their identical computers.I,m at a loss for the cause of the problem.Headphones work fine but are annoying.

i use professional studio monitors to monitor the audio
the issue , is if i turn up the audio it will clip on premieres main output. so i cant turn it up any more.
the specific moment i took the screenshot the audio volume was at -18db, but even if it was up at 0db it would not even reach half volume on my computers main output
this is the case regardless of the nature of the audio, be it stock music, dialogue, or sound effects maybe there is a hidden master output that im unaware of

Usage of sound level measurement

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