Measurement mics

Sure. That’s why they build acoustic labs. And why dB(A)?
Our marketing team hired a group of acousticians who went around to measure the real conditions in people's rooms where our product was supposed to be used. 10dBA was their answer. It became known later that other companies came to the same conclusions, so the number itself is not confidential. I am still under NDA and can not provide any further details.
 
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You'll be laughing. Very likely - if you turn your TV off.

Most people turn their TVs on at the very moment they come back home. First, it masks annoying road noise from outside (40dBA is a reasonable guesstimate). Then, TV becomes blah-blah-blah noise itself. If you turn it off... you suddenly realize that you can hear people walking their dogs half a mile away.

It all depends on the age of your building (50s ... 80s are no good) and how busy your nearby streets are. Alas, if you live near a highway, you are screwed.
 
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As to measuring below 10dB(A), one needs special microphones with like preamps. Gras offers those, B&K likely too. Normal sound measuring equipment from these brands isn't sufficient. Most pro integrated sound level meters hardly reach 20dB, the top models from B&K start at 16dB. So I wonder, how did those acousticians measure below 10dB(A)?
 

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Honestly, I do not know.

If I were asked to do it, I would ...
  • take a Rode NT1 mic (which is 4dBA)
  • use a good battery-powered audio interface
  • calibrate it to, say, 94dB as full-scale
  • make a long recording without anyone in the room
  • cut out the longest piece without unavoidable accidental noise
  • filter it as A-weighting says
  • take RMS

There are even better -2 dBA mics from B&K, but I would hesitate to take them into the field. You also can play with freezing the mic's PFET , etc...

BTW, there is a problem with the declared -20dBA Microsoft anechoic chamber and alike. There is no microphone capable of measuring such values honestly. So Guinness people take measurements by a low-noise mic in a known noiseless environment (say r1), then measure the chamber (r2), then calculate the noise as sqrt(r2^2-r1^2). The confidence intervals of such estimates would not fly by an experimental physicist but are good enough for mass-media publications.
 
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ScottG, you are absolutely right! Once upon a time, I played with previous analog versions of NT1. Do you know how to use NT1's digital out with other clocked DAC? It shall be very nice to have ADC so close to the mic, and get rid of hum 60Hz*N and other artifacts but the lack of clock synchronization would make it unusable.
 
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I use a dual-mic setup instead. A primitive ECM8000 (34 dBA, 105 dB at 10%) and AT4040 with 12 dBA self-noise and 145 dB at 1% THD (shall have 0.01% at 125)
(shall have 0.01% at 125)

For many years I have wanted to know at what sound pressure a microphone offers 0.01% (or 0.1% ...) THD.

In the data sheet there is either no information about the sound pressure, or 3%THD @e.g. 140dB SPL.
Or it says 1%THD @ XXXdB SPL @1 khz. And so on, I think it's clear what I want to know.

And:
What is frequency dependent here?

Best regards
Bernd
 
Hi Bernd,
AFAIK, microphone non-linearities are "soft", i.e. representable as infinitely differentiable functions (Volterra series, to be more precise) (loudspeakers are not, alas). Thus, if 1% THD is due to 2nd-order non-linearity, then you have 6dB of improvement in the relative level per each 6dB of lower SPL. If 3rd -> 12 dB of improvement, etc.
(x + a*x^2+b*x^3...) if x-> x/10 then (0.1x + a* 0.01x^2 + 0.001 x^3...)= 0.1*(x+a*0.1*x^2+a*.01*x^3...)

That means: to get 0.01% of THD on the mic running into 1% THD at 145 dB, you need to lower SPL by no more than 40dB, i.e. down to 105dB, so you calibrate your mic at -11 dBFS when using a typical calibrator. And, what I wrote above: 125 dB??? OOPS... Thank you for noticing and letting me know!

B&K used to have a long two-part white paper on microphones on their website, where they explained a lot of second-order effects.

The frequency-dependent is AT4040. Also, AT4040 is serial-number dependent, i.e. you need to correct this specific microphone, another one may be +-2dB on certain frequencies. Also, AT4040 is distance-dependent due to the cardio proximity effect. ECM8000 is (believed to) not - which error we consider acceptable.
 
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Do you know how to use NT1's digital out with other clocked DAC? It shall be very nice to have ADC so close to the mic, and get rid of hum 60Hz*N and other artifacts but the lack of clock synchronization would make it unusable.
No you would still be using the analog section for testing - the “loop-back” with ”dual channel” (and a audio interface) should “subtract-out” everything but the mic./mic.-preamp. (..it’s not solely about achieving a timing reference with dual-channel).

The USB (digital) portion would essentially be useless. :blush: Though perhaps not if you have other non-testing uses, it is after all a nice vocal mic.. :)

My guess is that the spec.s are also based on analog use rather than digital/USB (..which probably can’t handle such a high Spl).
 
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@b_force would you share the names of those companies (ideally around central europe)? Thanks!
Over the years, I have seen at least those cheap Dayton mics being completely off.
Already the ripple in the calibration response is a dead give away, since that's not how an electret capsule works.

I have owned two myself (still own one as a backup) and they were completely off as well.
It seems this problem happened after a certain moment in time.
No idea what happened in their factory.

I also have my strong doubts about these IK multimedia mics.

Just go on google and there are plenty of stories and reviews that can be found. :)

There has been a pretty good article about this on https://hifi-selbstbau.de/.
I will look it up.

They also have an excellent and very affordable calibration service btw.
 
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