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.
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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

1DC- Sound Recording via Line input woes

Have the latest firmware update on my 1DC. There are 3 issues I am hoping someone can help with
#1 I assume you must set Sound recording to 'Line input' and NOT manual, but if I set the Rec level, whenever the camera is powered off, either automatically or manually, this Rec level is ALWAYS reset to zero. Also, the moment recording is startted, even though audio input levels are set correctly, no audio is recorded. This seems like a firmware bug to me. This does NOT happen when set to manual. Anyone know of a workaround?
When using Manual and plugging in an external audio source I assume this signal will override the audio from the camera's internal mic.
If this is correct and there are no issues when using 'Manual', then what is the difference between 'Manual' and "Line input'??
#2 When setting Sound input to Manual, the VU meters are visible, but NOT when set to Line input. Why?
#3 Can anyone recommend a specific connector to go from line out on a Zoom H6 to Mic in on the 1DC?
Solved!
Go to Solution.

No. Apparently a buggy firmware release. My understanding is that there is another release planned soon. Hopefully with several new (and working) features in addition to the line input.
Using Manual with a -25dB cable for now.

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.

Read pressure signals using signalExpress

Hi, I don't have Pressure option under DAQMx Acquire -> Analog Input menu when I add a step to acquire pressure signals in SignalExpress 4.0. Anybody knows what extra module do I need to do so? Thanks. Cong.

Please post your question to the SignalExpress board. Very few users of it here on the LabVIEW board.

How can I get my Logitech H600 microphone to be recognized in my Sound Preferences as an input device?

How can I get my Logitech H600 microphone to be recognized in my Sound Preferences as an input device?
Logitech replaced my H600 wireless headphones after they were unable to overcome the problem.
My MacAir internal microphone works, but when I switch to the Logitech device, there is no recognition of my voice.

My operating system is Yosemite 10.10.2

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).
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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

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

Im using an iMic to record but now it all of a sudden won't pick up anything played through it, yet if i turn on garageband it works. logic pro does acknowledge that the imic is plugged in but still won't pick up any sound, even with audio input : imic

Im using an iMic to record but now it all of a sudden won't pick up anything played through it, yet if i turn on garageband it works. logic pro does acknowledge that the imic is plugged in but still won't pick up any sound, even with audio input : imic and audio output : external speakers. is this just logic pro not working properly?

Same problem - happened after recent logic update.

No sound on optical audio input but audio signal is there; mid-2007 iMac running OS 10.6.8

I've connected a digital turntable to my iMac via Toslink RCA/optical and I know the cables and converter are working. In the Sound preferences I can see evidence of an audio signal being received but there is no sound. This setup used to work before I had my hard drive replaced.

Erase the drive then install OS X:
Drive Partition and Format
1. Boot from your OS X Installer Disc. After the installer loads select your language and click on the Continue button. When the menu bar appears select Disk Utility from the Utilities menu.
If you are preparing an external or a non-startup drive, then open Disk Utility in your Utilities folder.
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3. Under the Volume Scheme heading set the number of partitions from the drop down menu to one. Click on the Options button, set the partition scheme to GUID then click on the OK button. Set the format type to Mac OS Extended (Journaled.) Click on the Apply button and wait until the process has completed.
4. Select the volume you just created (this is the sub-entry under the drive entry) from the left side list. Click on the Erase tab in the DU main window.
5. Set the format type to Mac OS Extended (Journaled.) Click on the Security button, check the button for Zero Data and click on OK to return to the Erase window.
6. Click on the Erase button. The format process can take up to several hours depending upon the drive size.
If this doesn't work then the drive is probably bad or some of the new RAM you installed is marginally defective.

Playing sound from the mic input

I'm running my turntable through a mixer and then running the line level signal into the mic input on my macbook. It sounds fine when I record the sound, but I can't seem to get it to play sound live.
I used to do this on my pc by Muting and un-Muting the mic or line-in input, but on the macbook this doesn't seem to work.
What am I doing wrong?

I guess I should add that the mixer is showing normal volume levels and when I run Audacity, the volume levels are normal. I just want to convert my input from "recording only" to playback.

No sound when using HDMI input to TV

I connect Mac Book to HDMI input on TV with a mini DVI to DVI cable, to DVI to HDMI cable. Good picture through iDVD but no sound.

Yeah. It kind of *****! But it is true, HDMI does not transmit sound, just video. Which it does well I must say.
What I did is I have surround sound and I plugged a cord that has male connectors on both sides and I plugged it into my amp on one side and my mcbook on the other, so now I can get sound onto my surround sound and also get the video images on my HDTV. =) If you have a surround sound system with an amp, I would recomend doing this. Its easiest.

How to use voltage signals as input signals into USB-6211

Hello everyone,
I have a USB-6211 and a PDQ80A -quadrant photo detectot from thorlabs, i am trying to take three voltage signals from the PDQ80A into my PC using USB-6211 from NI. THese voltage signals are X,Y and sum (X+Y).
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Thanks in advance,
_Perseus

Technically, the question was in the title. But of course, the answer to that is... Connect them using WIRES. It's designed for voltage signals, just wire your cell outputs to whatever DAQ input channels you're going to read.
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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!

Sound pressure signal as input

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