Signal Conditioning

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Octave Graph of a .WAV file and its calibration

I have a wav file consisting of a recorded calibration signal, the signal is a recorded tone with a sound pressure level of 93.9 dB (~0.99 pascal) at 1000 Hz. The equipment(microphone) I used to record the file "calibration.wav" has a sensitivity of around 53.49 mV/pascal. After performing 1/3 octave analyses on that .wav file it should show 94 db for 1000 hz but it is below that. Is there any way to fix it? Also tried to scale the time signal from V to Pa by using Scale to EU. But nothing was helpful. 

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Howdy Rathib,

 

Do you have any code snippets, graphs or hardware information you can share about your system?  Without having some additional context, it's hard to give more than an educated guess as to what could be happening.

Tyler O.
Applications Engineer
National Instruments
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Hi I have attached my code and the snippet of graph but I can't attached the .wav file which is a calibration tone. Octave should be 94db at 1000 Hz using 1/3 octave but it isn't. I have changed the engineering units to Pa. 

 

Please check the attached file. Thanks fro reply. 

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You have a mic signal  (with an attached pistophon?)  excited with a known dba level.

the scalar output of the mic (voltage, resp. numbers in a wav) now should represent a 94dba level.

the conversion (and it's inverse) from scalar (sine) numbers to dba is known 🙂 (up to you to g++gle it, hint: log conversions involved 😉 )

you look for a scaling (gain) factor (multiplication) for your scalar values to optain a 94dba result, or correct in the log scale (adding/substracting) .. up to you, just don't mix them 😉

 

(Sound and vib toolbox has some calibration vis...., however some knowledge on sound (pressure) measurement is still/always needed)

Greetings from Germany
Henrik

LV since v3.1

“ground” is a convenient fantasy

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Dear friends

 

 

I am starting a new challenge, analyzing noise from 2 microphones with labview.

I have a wave files with 3 channels. 2 for microphones and other (as microphone) as pulse counter from a car wheel.

 

I need to read the file, counting pulses (i have the code for that) and reading the 2 microphones channels and making a 1/3 octave analyzing, and save data into an excel. 

 

I use a very old and expensive equipment from Bruel to acquire the signals and at the end create a wav file.

After that I would try convert the 3rd signal into a distance with a car and analizing the other 2 channels (microphones) as a noise (1/3 octave) and joint it into a an excel file 

 

Seems this example from this post should be a begin for my project, but it is in a version greater that mine (LB 2013). I have SV module too.

 

I appreciate a start help.

By the way, I use the Audacity soft to cut modify the wav file, but by exporting it transform the 3 ch file into a mono file, why?

 

Best regards

cpalka

 

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