sorry, I guess I wasn't clear. I'm excluding sound produced by regular loudspeaker drivers - their distortion levels are generally higher than microphones, making them useless for distortion testing, afaik. So I'm looking for an acoustic source of non-distorted sine waves, not generated by loudspeakers...
I think B&K calibrates microphones using another microphone. They are reciprocal devices. There's a ton of great stuff on the B&K site and I think they cover what you're asking. Been a long time since I trolled this stuff.
http://www.bksv.com/Library/Technical Reviews.aspx
Technical Review 1998-1 Danish Primary Laboratory of Acoustics, Microphone Reciprocity Calibration, Calculation Program for Reciprocity, Calibration
A tuning fork has two vibrating elements and will produce very strange results depending on the accuracy of the two arms frequency and the angle of measurement - not good at all.
Of all the instruments, I think the flute is the purest so maybe Helmholtz was onto something
http://www.bksv.com/Library/Technical Reviews.aspx
Technical Review 1998-1 Danish Primary Laboratory of Acoustics, Microphone Reciprocity Calibration, Calculation Program for Reciprocity, Calibration
A tuning fork has two vibrating elements and will produce very strange results depending on the accuracy of the two arms frequency and the angle of measurement - not good at all.
Of all the instruments, I think the flute is the purest so maybe Helmholtz was onto something
Omnidirectional condenser microphones (like all those famous BKs) are usually calibrated via an electric field (consisting of the AC test signal plus a bias voltage) applied by an electrode in front of the diaphragm. So you do in fact have a directly coupled ESL/microphone combination. There won't be any method (at least not in the universe as I know it) that is more accurate than this one. Due to the usual directionality issues - that are cancelled out this way - there must be some calcualtions done on the measurement results in order to get the actual free-field response afterwards.
Regards
Charles
Regards
Charles
From Wikipedia:
A tuning fork is an acoustic resonator in the form of a two-pronged fork with the tines formed from a U-shaped bar of elastic metal (usually steel). It resonates at a specific constant pitch when set vibrating by striking it against a surface or with an object, and emits a pure musical tone after waiting a moment to allow some high overtones to die out. The pitch that a particular tuning fork generates depends on the length of the two prongs.
Not sure how "pure" the waveform would be. Might be a good idea to record the waveform produced by a tuning fork and looking at it on a scope or on audio editing (Adobe Audition) software.
A tuning fork is an acoustic resonator in the form of a two-pronged fork with the tines formed from a U-shaped bar of elastic metal (usually steel). It resonates at a specific constant pitch when set vibrating by striking it against a surface or with an object, and emits a pure musical tone after waiting a moment to allow some high overtones to die out. The pitch that a particular tuning fork generates depends on the length of the two prongs.
Not sure how "pure" the waveform would be. Might be a good idea to record the waveform produced by a tuning fork and looking at it on a scope or on audio editing (Adobe Audition) software.
phase_accurate said:
Thanks, I just saw that. I guess it works only for pressure sensitive mics (omnis), and that cardioid mics would then have to reference these mics.
I've been looking into distortion levels for mics, and found an interesting article on Linkwitz's modified wm61a electret capsule, published by audioXpress - pdf is here.
The panasonic capsule is compared with a B&K reference mic, with pretty decent results, about .5% THD at 125dB SPL. My next question is how this compared with studio mics, and apparently its about average, with small diaphragm condensers having somewhat less distortion, and large diaphragms a little more. Not bad, sounds like the panasonic capsules can be used for reference.
The next thing I wonder is what the THD levels are at lower SPLs. I read something saying you could extrapolate, for each 6dB drop in SPL, you'd expect half the THD. For the modified panasonic capsule, at 95dB, that implies about .02 %THD, or about -74dB down. Not bad, but I wish it were better - some drivers have distortion around this level, which might make measuring them problematic. Sure show the primary limitations on fidelity are the transducers (loudspeaker drivers, headphones, microphones)....
At low levels the limiting factor is noise: room ambient, FET 1/f, resistor etc. Its hard to get more than 60dB SNR in an acoustic environment.
That's what you pay for with an expensive mic - good sensitivity and low noise. The Panasonics have very good response but generate more noise than your lab mic.
That's what you pay for with an expensive mic - good sensitivity and low noise. The Panasonics have very good response but generate more noise than your lab mic.
loudspeakers could be acoustic filtered as mentioned above, indirect measurement of microphone IMD products in the sound feilds of 2 speakers driven with different frequencies could avoid the harmonics in each speaker's output
for a single sine drive the 2nd harmonic would decrease ~ with level, n-th higher order harmonic percentage are expected to decline as n-1 power
and as usual, 2nd harmonic distortion isn't expected to be audible at even 3% of a sine wave, but 2nd order nonlinearity generates IMD difference products with complex waveforms that can be lower in frequency than the excitation and therefore lie on the steep "front" slope of the masking curve and should be much more readily audible
this points to higher harmonics but shows a masking curve:
http://www.listeninc.com/files/pdf/Rub&Buzz.PDF
of course Klippel is the expert on dynamic speaker distortion:
http://www.klippel.de/pubs/papers.asp
and Geddes (who is active here) has studied nonlinear distortion audibility and speakers for a while now and his current position seems to be that loudspeaker driver nonlinear distortion is seldom a problem and most effort shoud be put into the "linear" part of loudspeaker/room design such as frequency response/directivity/early reflection/diffraction control
http://www.gedlee.com/
for a single sine drive the 2nd harmonic would decrease ~ with level, n-th higher order harmonic percentage are expected to decline as n-1 power
and as usual, 2nd harmonic distortion isn't expected to be audible at even 3% of a sine wave, but 2nd order nonlinearity generates IMD difference products with complex waveforms that can be lower in frequency than the excitation and therefore lie on the steep "front" slope of the masking curve and should be much more readily audible
this points to higher harmonics but shows a masking curve:
http://www.listeninc.com/files/pdf/Rub&Buzz.PDF
of course Klippel is the expert on dynamic speaker distortion:
http://www.klippel.de/pubs/papers.asp
and Geddes (who is active here) has studied nonlinear distortion audibility and speakers for a while now and his current position seems to be that loudspeaker driver nonlinear distortion is seldom a problem and most effort shoud be put into the "linear" part of loudspeaker/room design such as frequency response/directivity/early reflection/diffraction control
http://www.gedlee.com/
jcx said:
Shocked again, I was working out the best way to do this just last night. I was going to try to exercise the diaphram with a large lowish frequency tone (B&K mentions 90Hz in their handbook) and then using a lower level one at a higher frequency look for the IMD tones on the high frequency one with FFT's. I figure with enough averaging one could resolve pretty far down in the noise. Just a couple of power levels about 20dB apart should get some kind fit to theory for say a transducer with small 2nd and 3rd order non-linearity. Sort of indirectly trace the transfer function.
Has this ever been written up anywhere? Going up to 140dB on
the 90Hz might require the rest of the households cooperation.
a question occurs to me:
How does Doppler effect enter in this setup? - any high frequency sound reflected from the low frequency driver will be modulated - can the Doppler products be separated by phase? or do you just have to make them small (driver directivity?) and accept the residual as a resolution limit?
How does Doppler effect enter in this setup? - any high frequency sound reflected from the low frequency driver will be modulated - can the Doppler products be separated by phase? or do you just have to make them small (driver directivity?) and accept the residual as a resolution limit?
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