Which microphone to use for speaker analysis?

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I'm looking at testing some speakers with speakerworkshop or some of the other software programs available. Is there a particular recommended microphone to use?
I'm not wanting to spend much, but am under the impression that common electret condenser microphone elements can be very good.
 
Any mic can be calibrated. Finding someone to do it will be tricky + costly. Note: that when they use the term "calibrate" they do not flatten the response of the mic, but give you a list of corrections (in dB) relative to a flat line from xHz to xkHz to be entered into your measurement data. E
 
I already have an Akai electret mic - can any mic be calibrated to be used for measurement? If so, is there a way to calibrate a mic within Speaker Workshop? Thanks.

Measurement mics need to be omnidirectional. A directional (unidirectional, cardioid, anything but omni) mic changes frequency response depending on distance from the source.

You can check mic frequency response by measuring a tweeter that is fairly flat when mounted flush to a large baffle. Most electret mics have decent low frequency response, at least to 30Hz and some go lower, and are flat above that until their second resonance, which will perhaps be around 10-15k, depending on mic size. Larger diameter mics have lower resonances.
 
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I recently bought a Superlux ECM-999 measurement mic on ebay, new, for $74 + $10 shipping. It's a 1/2-inch condenser type and is omni-directional and supposedly very flat from 20 Hz to 20 kHz, according to the plots in the ad on ebay. It came with a hard case, windscreen, and stand mount hardware. It has a 3-pin XLR female connector jack in the base.

It does need a mic pre-amp, or some other way to supply the phantom power. I bought a new TubeMP mic preamp at the same time, for about $30, plus an XLR-male-to-XLR-male mic cable for input from mic to pre, and a guitar cable (1/4" male to 1/4" male) and a 1/4"-male-to-3.5mm-mono-male adapter for pre out to soundcard input, and it all seems to work very well. I've been able to start dabbling with ARTA.

Just FYI, the TubeMP mic preamp has XLR female and 1/4" female input connectors and XLR male and 1/4" female output connectors. It seems like a nice little unit, for the price. It does have a vacuum tube. And it has a lighted VU meter, and different gain range options and knobs for both input and output, and 48V phantom power that can be switched on or off.
 
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Thanks for all the above info. The Akai mic is in a storage locker so I'll have to try to find it and check the spec sheet to see if its omnidirectional-got a feeling its cardioid. Already got a M-audio Duo mic pre, gootee but that TubeMP pre sounds like a huge bargain.
 
For measurements, you want a mic pre that is as straight as possible. Based on tests I saw on other M-audio equipment, I suspect it will be 20-20KHz +/- a fraction of a dB. Still, it would be a good idea to measure its FR before you use it. Easy to do with the same software you use for measuring speakers. Just feed the attenuated output signal generated by your measuring software into your mic input and do a sweep.

vac
 
Measurement mics need to be omnidirectional. A directional (unidirectional, cardioid, anything but omni) mic changes frequency response depending on distance from the source.
Erm, can you please explain how microphone directivity would cause the measured frequency response to change with distance from a sound source differently than it would with an omni ? I can't see any mechanism where this would be the case.

Now, if you're talking about measuring an un-gated steady-state response in a room then yes a directional mic will give a different result than an omni at high frequencies because it's collecting less of the room response, but neither result is a useful or valid measurement of a speaker, nor of how the room response will be perceived at high frequencies.

An on axis gated measurement to a single driver will work just as well with a directional mic as an omni, with no difference in response (between omni and directional) at different distances.

On a large speaker measured too close the more directional mic might show a slight difference due to being slightly off axis to some of the drivers, but even with an omni mic in that situation you still have significant errors due to being off-axis from the drivers.

There's nothing inherently wrong with some microphone directivity for speaker measurements, the real reason for wanting an omni mic for most measurements is to get the capsule size as small as possible to push resonances up out of the audible range and to minimize diffraction - most cardioid and directional mic's are physically large and therefore have problems with in band resonances and diffraction.
 
Is it just a case of correcting the mic's frequency response? Are they essentially perfect apart from a few bumps in the frequency response?

I'm very excited by the idea that it might be possible to go further than correcting a speaker's frequency response and effectively convolve the signal with a correcting FIR filter or similar to completely linearise it. Are these mics up to the job of providing the measurements? (given suitably anechoic surroundings etc.)

Beyond that, it might also be possible to create self-learning multidimensional FIRs that correct for non-linear nasties as well...

I have in mind the idea of being able to build a speaker using any drivers in any box and once the thing has been physically tweaked to the best it can be, the DSP can refine it to as close to 'transparent' as is possible. I would never be bored again.
 
In the past I have wanted to do live compensation for speakers response, but now its a more humble measure and equalize. Things like the 'Convolve' filter for ffdshow sound amazing, I can't wait to try it, but first I'm going to build great speakers to start with.

The microphone I ended up getting is a FM-6B from www.hy-q.com.au (found this in the forum). $6 ea + postage in Australia and flat supposedly to about 1db from 20-20K.

Some Ideas I had in the past-
1) Use a microphone directly in front of the woofer for closed loop control of the sound. This may be possible with sealed boxes only. I have read an article on a design a long time ago, but it had a loop gain of 4 which is woefully low - not much control at all.

2) Use a microphone as part of a control loop continually adjusting a FIR filter. Essentially doing live adjustments to the equalisation based on microphone input while you are using the stereo. This is very similar to the way noise cancelling headphones/speakers work, except what is being cancelled is the difference between your speakers behaviour and a perfect sound. (Only acts on linear effects though, you need the previous method for non-linear).
 
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, can you please explain how microphone directivity would cause the measured frequency response to change with distance from a sound source differently than it would with an omni ?

Proximity effect: Proximity effect (audio) - Wikipedia, the free encyclopedia

Every unidirectional pattern microphone (figure of 8, hyper,super,cardio,subcardio, hypocardio) exhibit this behavior. This is the reason for use of omni mic for analysis.

The best i've heard are Bruel & Kjaer (now DPA) and some Earthworks but they are clearly off budget... The little behringer is ok as long you don't go over 90/94 DB Spl. Beyond that level you can't trust them.
 
Could you do something like bolt the speakers together, facing each other, driving one with a test signal while measuring the output of the other?

I haven't quite worked out what it would tell you. Maybe nothing!

It is called reciprocity. You don't need to bolt the speakers, just face them at each other about 1 meter apart. Drive one and use the other as a microphone, reverse the hook up and do it again.

That is how microphones get calibrated!

One of these days I will do an article on it!
 
It is called reciprocity. You don't need to bolt the speakers, just face them at each other about 1 meter apart. Drive one and use the other as a microphone, reverse the hook up and do it again.

That is how microphones get calibrated!

One of these days I will do an article on it!

This is very interesting!

Are you saying that if a speaker has, say, a particular group delay for a range of frequencies, that by using a matching speaker as the 'microphone' this will be doubled? Similarly, I imagine that resonances in the playback speaker would also enhance resonances in the 'microphone'.

Not quite sure about the significance of the 1m... Are you then introducing a need for an anechoic chamber etc.? Or do you need a gap for a more representative free field response as opposed to the pressure response anyway?
 
It is called reciprocity. You don't need to bolt the speakers, just face them at each other about 1 meter apart. Drive one and use the other as a microphone, reverse the hook up and do it again.

Actually you need an auxiliary (third) transducer. You can use two speakers and your microphone.

You can do self reciprocity with a single microphone and a perfectly reflective wall. But it is tricky and mics don't make very strong signals.

It is difficult to get good data with good signal to noise. I did a trial reciprocity calibration using my behringer and an omni dynamic mic and a tweeter. I got plausible results, but I can't know for sure if they were correct.
 
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