Sound pressure reporting

Hi,
We are new (actually, still evaluating) LabView customers. I know nothing about Labview, and less about the technical intracacies of the project that I have to complete. Basically, what I need to do is acquire and capture up to three streams of sound pressure data using B&K 4944-A sound pressure mics attached to an NI 9233. I think we can do this with SignalExpress and the S&V Assistant. In essence we are replacing three B&K 2209 sound meters with B&K 4136 pressure mics capturing one or more precussive sound events. The new hardware is already purchased. We are trying to get the capture and reporting going.
My need is:
1. Capture a continuous stream of log data beginning with starting to record and ending with stopping the recording. We need to trigger the start of the actual logging based on a rising slope over 110 dB, and capture data for .25 seconds after the trigger is met (I have been unable to capture any data yet - probably has to do with I don't know where the values to set in the trigger). We need the trigger to be activated continuously until the recording is stopped.
2. Reporting. For each triggered log event above, need to extract and report the peak sound pressure level (dB) for the three data sources logged. Tabled data is OK; graphs and charts are not necessary.
3. Would like to (automatically, if possible) store the results from #2 above in a database to evalute potential product changes. For now we can use Visual Pen and Paper.
Can anyone help with a starting point?

Hello,
Triggering based on the level of the signal itself is known as an analog start trigger. Since the NI 9233 does not have analog or digital triggering, we could use another module (like a NI 9205) to start the task that the NI 9233 is in.
How Can I Trigger a NI 9233 or 9234?
http://digital.ni.com/public.nsf/allkb/4859504F14AF68DB8625721100640F26
If you do not want to buy another module, you could try post-processing the data to ignore any data that comes in before this level.
I hope this helps.

Similar Messages

Sound pressure spectra with SVT

Hey all,
I have a project which I have been having a bit of trouble implementing. I'm performing several measurements at 48kHz/24Bit of some sound events. My requirement are as follows for sound pressure spectra analysis.
500 - 20 kHz frequency bandwidth with a 25 Hz frequency resolution.
Peak Hold averaging with 50% overlap processing and Hanning windowing applied to the time records.
Anti-aliasing filtering.
A - weighting.
I have the sound and vibration toolkit and have implemented as follows:
1. Read wav
2. Convert from Volts to dB via SVL Scale Volts to EU.vi. (I used a calibrated signal of 94dB at 1kHz to find my mic sensitivity).
3. A-weighting using SVL A,B,C Weighting Filter.vi
4. Peform analysis using SVL Zoom Power Spectrum.vi.
Few Issues:
I dont know how to implement the Anti-aliasing filter.
Is the method I have used ok with measuring sound spectra? Or should I perform the FFT then convert to dB?
I have tried to implement the same analysis using Matlab and although I get the same FFT profile, my dB levels are different where the 16kHz noise is almost the same level as the 8kHz noise? Not sure what is happening there, but I want to concentrate on using LV for now.
I dont have any guidance available or experience with this sort of analysis. I have trialed my program out on a sample wav file which has two separate distinct noises events, one at 8kHz and another at 16kHz. My FFT is showing that the 16kHz is a much higher sound level (95dB) than the 8kHz (80dB) but when I listen to the wav via a pair of studio monitor speakers, the 8kHz is much more audible. Wouldn't the implementation of A-weighting filter tend to make the 8kHz noise more prevalent in the FFT?
Any suggestions on this topic are appreciated.
Thanks!

Hi David,
If you want to enable the enhanced alias filter, you can use a DAQmx channel property node. I would recommend placing a DAQmx Channel property node on the block diagram and then going to Analog Input » General Properties » Advanced » Enhanced Alias Rejection Enable. See if that works and let me know.
Jake H | Applications Engineer | National Instruments
Attachments:
Enhanced Alias Rejection.png ‏3 KB

Diagram or vi for Sound Pressure Level Meter (SPL)

Need help for implementing a Sound Pressure Level Meter (SPL) with LabVIEW 7.1 or 6.3....I must use a microfone and a laptop for my project.....Need diagram or ready vi ....!!!
How can I make an A,B,C Weighting Filter vi for my Sound Pressure Level Meter project....???
If there is one already made or you can give me any help please send it to me.......[email protected]
Thanks.....

Search the archives. A similar question about the weighting filters was posted within the past month or two, if I recall correctly. I am not aware of any ready-built VIs, but the filter specs are published (Google A-weighting).
Also be careful with the frequency response of inexpensive microphones. They can skew the results substantially if you do not have some way to measure and compenste for the response.
Lynn

I'm looking for a .vi for sound pressure level meter (SPL)

Hi....
I'm looking for help regarding the implementation of a SPL (sound pressure level)meter using LabVIEW.....If anyone has a vi or diagram or know how to proceed with this project please tell me about it...
According to the project , I need a microfone and my laptop with LabVIEW to make the vi.
Please help
Thanks

Here's an example program:
This VI simulates the functionality of a Radio Shack Analog Sound Pressure Level (SPL) Meter. This VI requires a data acquisition board, Radio Shack SPL Meter and an RCA cable. Connect one end of the RCA cable to the SPL Meter's output, and then splice off the other end and connect it to channel 0 of the DAQ board. The connection to the DAQ board must be made differentially.
The output of the SPL meter is a raw waveform that is converted by the program into a decibel (dB) reading. The front panel shows a dB offset based on a range, just like the actual SPL meter. The front panel also displays the overall dB reading.
Attachments:
SPL.llb ‏88 KB

Sound pressure signal as input

I want to use a function SV Time-Varying Loudness in the Sound and vibration suite. The output of this function is exactly what I require for my speech analysis.
I have a speech sample recorded from the microphone obviously a voltage signal. This function requires a sound pressure signal.
The voltage signal would be proportional to the sound pressure signal right? How do I convert this voltage signal to the sound pressure signal (Pa).
I'm not even sure if I can use this function. I trying to extract the features from the speech signal. The algorithms used for this are linear predictive coding , perceptual linear predictive coding and mel frequency cepstral coefficients.
I already implemented Linear predictive coding in labVIEW. I wanted to try out Perceptual linear predictive coding because it is considered to be robust in noisy conditions.
This SV time varying loudness function does the exact thing I want to do but I don't know if it is applicable to speech signals.
In the help I says this VI process 2ms blocks of sound pressure data. But speech signals are to be processed only at 10 to 15ms. Please help

Your microphone should have calibration tables to relate the output voltage to an audio quantity like dB or perhaps even Pa.
Mike...
Certified Professional Instructor
Certified LabVIEW Architect
LabVIEW Champion
"... after all, He's not a tame lion..."
Be thinking ahead and mark your dance card for NI Week 2015 now: TS 6139 - Object Oriented First Steps

How to generate .wav files from sound pressure data

Hi,
I have sound pressure data (as dB values) vs frequency data.
How can I create a playable .wav file based on above?
Note: I do NOT have sound pressure vs time data
I am working in LV 7.1 FDS on Win2000.
If really required, I can incorporate VIs from sound and vibration toolkit (version 1.0).
However, I would prefer a solution that does not need me to do that.
Thanks in anticipation,
Gurdas
Gurdas Singh
PhD. Candidate | Civil Engineering | NCSU.edu

All,
With some help from NI-India, we managed to build a nice sound player and .wav saving utility.
I have attached the VI sent by NI-India. The key component which was missing in my VI was "resample waveform".
Note: For some reason, I could not use the attached VI to generate sound but did use the concept to succesfully read my sound pressure data and output it to the speaker (and also save it as .wav file tha can be played on Windows Media Player). Also note that I am now generating sound from sound pressure vs time data. I have not tried generating sound from sound pressure vs frequency data.
I would love to have the following answered:
1) What data does the sound write VI expect? Is it Sound Pressure values? What units (N/m2 or dB or dBA)? I think it expects sound pressure data and the units are not relevant. Reason being my next question.
2) I found that until I mutliply the sound pressure data by a very large number say 10e7, the sound is hardly audible (even when volume was programmatically set to max. i.e. 65535). The sound QUALITY does not change when I change this multiplier.
2.a) Does that mean its only the relative difference between wave data points that affects sound quality? If yes, then I believe the sound pressure data can be in any units.
2.b) Is it expected to use this huge multiplier OR did I do something without really knowing what it meant?
Thanks,
Gurdas
Gurdas Singh
PhD. Candidate | Civil Engineering | NCSU.edu
Attachments:
Sound from FFT (LV 8).vi ‏165 KB

Log 8 hours sound pressure data

Hi all,
I want to log an 8 hour signal from a pcb mic. with the sound and vibration assistant(signal express). I want to acquire at 50KHz, but i think there wi'll be too much data. Is there a WAY to reduce that data?
thank you
pollon
Solved!
Go to Solution.

HI Pollon,
what about averaging the signal in a certain amount of time, to reduce the number of points you have to save?
If for example you're acquiring at 50 Hz, you can save the mean value for every 50 points, and in that case, you're saving one point each second.
Hope it helps,
have a nice day!

FFT Amplitude into Sound Pressure Level (SPL)

Hi everyone,
I have a problem with my program. Basically, i have a chirp signal to be outputted into my DAQ and being fed to my speaker.
The microphone will sense anything coming out from the speaker, and acquire it using my DAQ.
I want to display a frequency response vs. SPL.
My method is that, I'm using FFT on the received Chirp Signal from the microphone and display its frequency response.
My questions are?
1. Is by doing mere FFT will give me an output of SPL vs Frequency?
2. If not, how do I convert this value of the amplitude of the FFT to SPL db?
3. Why is the output amplitude of my FFT is negative?

The FFT is a mathematical tool that outputs complex values that contains frequency and phase information. If you want to obtain the amplitude of your signal as a function of frequency, it is easier to use the "Power and Phase Spectrum" vi in the -> Signal processing -> Spectral palette.
Marc Dubois
HaroTek LLC
www.harotek.com

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
~~~~~~~~~~~~~~~~~~~~~~~~~~
"It’s the questions that drive us.”
~~~~~~~~~~~~~~~~~~~~~~~~~~

IOS 6.1 still sound bug

Dear comm, AND DEAR APPLE SUPPORT!
It is really annoying!
Today I have updated my iPad2-16GB-3G to iOS 6.1 and that bloody sound bug is still there.
[Shortly the sound works and randomly after a short period of time it chrashes. No hardware key and no soft key works anymore. The soft key is gone.
I saw several chats regarding the different iPhone / iPad versions dealing with the same prob.]
I would appreaciate an opportunity to put more pressure on Apple to solve these problems. These wonderful gadgets like iPhone and iPad are that expensive!
And one of the key functions does not work. A bug that is not fixed since month, now!
and double it!
To all persons concerend: Stand up and put pressure on the Apple support to solve that problem!
THX and sorry for venting my anger!
joapp

First, Apple Support is NOT reading these forums. These are user to user technical help forums hosted by Apple. There are only other users here.
You may have a hardware problem. Visit an Apple store for evaluation or contact Apple support.
There is a chance that you have a bad Application on your iPad which is crashing the sound driver but, fron your description, this seem unlikely, however, give that possibility some consideration.
Note: the majority of sound problems reported here are caused by the user having mute turned on and not realizing it. You don't seem to have that from your symptoms.

Randomly hang in Crystal Reports for Visual Studio 2005

after a certain number of prints (PrintMyDoc) the application hangs on the line of code shown below
the void is PrintMyDoc
private ConnectionInfo CreateConnectionInfo(string User, string Password, string Server, string Database)
            ConnectionInfo objConn = new ConnectionInfo();
            objConn.UserID = User;
            objConn.Password = Password;
            objConn.ServerName = Server;
            objConn.DatabaseName = Database;
            return objConn;
private void SetDBLogonForReport(ConnectionInfo connectionInfo, ReportDocument reportDocument)
            try
                CrystalDecisions.CrystalReports.Engine.Database db = reportDocument.Database; // HUNGS HERE !!!!                CrystalDecisions.CrystalReports.Engine.Tables tables = db.Tables;
                foreach (CrystalDecisions.CrystalReports.Engine.Table table in tables)
                    TableLogOnInfo tblInfo = new TableLogOnInfo();
                    tblInfo.ConnectionInfo = connectionInfo;
                    table.ApplyLogOnInfo(tblInfo);
                foreach (CrystalDecisions.CrystalReports.Engine.ReportDocument subrep in reportDocument.Subreports)
                    foreach (CrystalDecisions.CrystalReports.Engine.Table table in subrep.Database.Tables)
                        TableLogOnInfo tblInfo = new TableLogOnInfo();
                        tblInfo.ConnectionInfo = connectionInfo;
                        table.ApplyLogOnInfo(tblInfo);
            catch (Exception ex)
                MessageBox.Show(ex.Message);
the objects are disposed as shown in the lines here below.
        protected override void Dispose(bool disposing)
            try
                if (!(crystalReportViewer.ReportSource == null))
                    ((IDisposable)(crystalReportViewer.ReportSource)).Dispose();
                    crystalReportViewer.ReportSource = null;
                try
                    objReport.Close();
                    objReport.Dispose();
                    objReport = null;
                catch (Exception)
                crystalReportViewer.Dispose();
                crystalReportViewer = null;
            catch
            if (disposing && (components != null))
                        components.Dispose();
            base.Dispose(disposing);
public void PrintMyDoc()
            SqlConnection conn = OpenConnection();
            if (conn != null)
                if (System.IO.File.Exists(reportName))
                    ReportDocument myReport = new ReportDocument();
                    myReport.FileName = reportName;
                    myReport.PrintOptions.PrinterName = Stampante;
                    ConnectionInfo objConn = CreateConnectionInfo (Utente, Password, conn.DataSource, conn.Database);
                    SetDBLogonForReport(objConn, myReport); // <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
                    this.crystalReportViewer.SelectionFormula = SelectionFormula;
                    if (Preview)
                        this.crystalReportViewer.ReportSource = myReport;
                        this.crystalReportViewer.RefreshReport();
                    else
                        myReport.RecordSelectionFormula = SelectionFormula;
                        myReport.Refresh();
                        myReport.PrintToPrinter(Numcopie , false, 0, 0);
                        objReport = myReport;
                        myReport = null;
                        this.Close();
                    conn.Dispose();
            else
                MessageBox.Show("Error", MessageBoxButtons.OK, MessageBoxIcon.Error);
Hope someone can help me asap
thanks

How did you come to the conclusion that it is a problem with the UFL? If it is a problem with the UFL then there is little that we can do to troubleshoot it because the UFL is developed outside of Crystal. The code can do anything. You'd need to follow up with the person who wrote the UFL.
If you are still unsure of the cause, then I suggest you start testing with other reports. Use ones with no database connection as a control group, then use reports that DON'T include the UFL.
After that, you could capture a stack dump of your application and look at the results. See if you can find where a problem library file is. One thing to note, we won't look at your stack dump here. That is beyond the help we offer on forums.
You could also look at restructuring the portion of your application that handles Crystal Reports. Instead of calling Crystal within your application, you could spawn a new process that does your reporting then shuts down. It will help to isolate the reporting to its own application space that is only run when a report is requested. It doesn't sound like reporting is a large portion of this application if you run 20 reports a day and it runs for 10 hours straight.

Simple Java Sound Question...

Can I draw waveforms while capturing data? If yes, how. If no, why.
Thanks. I need the answer ASAP, please help.

Hi nfactorial,
Can I draw waveforms while capturing data?If you would like to do that in Java you would need to know about how to use the Java language in general and especially about the Java Sound and Java2D APIs.
If yes, how.It would be too much to explain to answer such a general question. A general answer is, you need to use the Java Sound API to capture the data for which a waveform should be drawn. The Sound API delivers data as a stream of bytes typically encoded in pulse coded modulation (PCM) format. The data stream has digital samples each representing a level of sound pressure for a certain point in time. The stream of samples in the amplitude/time domain need to be transformed to a spectrum of samples in the amplitude/frequency domain, i.e. a number of PCM samples need to be mapped to the frequencies that should be displayed. This is done with the fast fourier transformation algorithm. Each set of amplitude/frequency values can then be displayed as a waveform by using some line drawing logic. The entire process would need to run constantly, i.e. as bytes are received from the sound data stream transformation and drawing is triggered.
Related readings:
Java Tutorial
http://java.sun.com/docs/books/tutorial/
Java Sound Documentation
http://java.sun.com/j2se/1.5.0/docs/guide/sound/index.html
Java Sound API Programmer's Guide
http://java.sun.com/j2se/1.5.0/docs/guide/sound/programmer_guide/contents.html
Java Sound Resources
http.//www.jsresources.org
Java 2D Graphics Tutorial
http://java.sun.com/docs/books/tutorial/2d/index.html
Wikipedia on fast fourier transformation
http://en.wikipedia.org/wiki/Fast_fourier_transform
HTH
Ulrich

Help with an EtreCheck Report

I have an older iMac running Yosemite and I just ran the EtreCheck and this the report. I am not sure what it means by "memory pressure" or what the solution is to the issue?
Thanks
EtreCheck version: 2.1.8 (121)
Report generated April 3, 2015 at 11:48:31 AM EDT
Download EtreCheck from http://etresoft.com/etrecheck
Click the [Click for support] links for help with non-Apple products.
Click the [Click for details] links for more information about that line.
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        BANK 0/DIMM0
            4 GB DDR3 1067 MHz ok
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            4 GB DDR3 1067 MHz ok
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Problem System Launch Agents: ℹ️
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    4 processes killed due to memory pressure
Problem System Launch Daemons: ℹ️
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    6 processes killed due to memory pressure
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    127.75 GB    Page-ins
    4.32 GB    Page-outs

Memory pressure indicates something is stressing your RAM and It appears to be CleanMyMac. It is unneeded and generally causes problems. Completely remove it. Also you don't need Memory Clean.
I suspect the "memory pressure" report will disappear once you evict those apps.
Also this:
System Software: ℹ️
    OS X 10.10.2 (14C109) - Time since boot: 55 days 17:23:37
Restarting your system once a week or so will automatically clean out a bunch of temp stuff that might be adding to the problems.
Your 8GB if RAM is plenty as long as you don't cripple your Mac with unneeded third-party utilities. You have a Mac, not a Windoze computer.

AW7 sound in button won't play

I have an AW 7 piece where I have imported graphics and
SOUNDS into a button. The packaged piece shows the button graphic,
but when the cursor goes over the button, I get an error prompt
stating: "... while trying to play the sound. The internal sound
driver reports the following error. Xtra not found." What am I
missing. Help please.

> If it is a .wav file, you'll need WAVREAD.X32, and, I
think, either
> MIXVIEW.X32 or MIX32.X32.
>
> If you use OBP, it should include the needed xtras for
you.
>
Actually there is a bug in OBP that does not find Xtras if
sound is used
only in buttons. The workaround is to include one of the
sound files in a
sound icon somewhere on the flow - or else learn which Xtras
are required
and be sure to add them manually.
Steve
ACE - Adobe Community Expert
My blog -
http://stevehoward.blogspot.com/
Authorware tips -
http://www.tomorrows-key.com

Sound pressure reporting

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