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cancellation of background noise in an acoustic chamber

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Hi everyone,

 

I have this project which will make use of an acoustic chamber, we will test a loudspeaker to display its frequency response.

 

I'm generating a chirp signal via my DAQ to be fed to a speaker - and the microphone will acquire this signal.

The VI I created will display the frequency response of the speaker (SPL dB vs. Frequency).

 

 

However, we want to make sure we are getting the least possible noise present in our chamber. We want to cancel these frequencies (background noise) in order to have an output of the frequency response which is only produced by the speaker. How do I cancel these frequencies to obtain my objective.

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Hi gijude,

 

put more acoustic insulation around your acoustic chamber.

Fix all wires going to your speaker and mic…

Best regards,
GerdW


using LV2016/2019/2021 on Win10/11+cRIO, TestStand2016/2019
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Solution
Accepted by jamison.suade

I agree with GerdW. If the noise is truly random, it is almost impossible to remove after acquiring the signal.  If your noise is an interfering signal, such as hum from power devices or whine from a blower, it might be feasible to subtract the background. However, and variation or drift in the frequency or amplitude of the interference over time makes cancellation much  more difficult.

 

Acoustic absorbing materials will also reduce the effects of any reflection of sound from your loudspeaker inside the chamber. Such materials work quite well, are not too expensive, and last a long time as long as they are not misused.

 

Lynn

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Hi sir John

 

My noise is an interfering signal, how do i basically remove this or subtract this using only LabVIEW's capability, and not adding any external absorbing materials. I just wan't to minimize the background noise

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Record the interfering signal, possibly outside the chamber. Then compare that to the signal measured insde the chamber (with the test loudspeaker not driven). Determine how much filtering, phase shift, and attenuation the chamber provides between the two versions of the signal. Create a matching filter in software such that its output is nearly identical to the interfering signal measured inside the chamber. Then subtract that from the signal measured while the loudspeaker is being tested.

 

Better is to prevent the interfering signal from getting into the chamber in the first place.

 

Lynn

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Usually you repeat these measurements n times.  signal adds by n  and random noise by sqrt(n) ......

 

As smaller the bandwidth as smaller the noise .....   but you get blind to nonlinearities....

 

 

Greetings from Germany
Henrik

LV since v3.1

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