Analog to Digital Converters

No need for that, just a2d the incoming waveform wideband and sort the channels out after doing the FFT.

Zafdor: It’s been years since any one has posted the term FFT. Years ago when this and other Forums were more technically oriented it was quite common to discuss Mr. Fourier’s Fast Transform along with A/D bit info. Ed :slight_smile:

There is a lot of wrong and misleading information in this thread.

Dynamic range is given in dB, not dB SPL.

Sampling resolution needs to be sufficient to represent the wide range of sound pressures and this range is much wider than the range of sound pressure levels (SPL).

Music has high peak-to-average ratio so high dynamic range is necessary for coding the waveform without distortion.

Music can have very high peak amplitudes at moderate level, it is not necessary to go to a rock concert to benefit from wider dynamic range in sampling.

Decreased hearing dynamic range does not imply that less sampling dynamic range is needed.

Out of all hearing aids I have used, Widex (music program) is closest in quality to analog hearing aids. But not close.

Sorry, at least one of those was my bad. The inference of the reduced dynamic range question was based on a recollection of relative and absolute sequential samples - as you say though, it doesn’t have any bearing as you still need to be able to code all parts of a HF signal irrespective of the amplitude.

Point taken about sampling rates: they still have to reflect Nyquist: however if your mic doesn’t do anything above 10Khz you won’t need to sample at much more than 20KHz

I was only talking about sampling depth, not sampling rate.

As for the Nyquist rate, to my understanding it only applies to stationary signals but not transients. For example, for a very short impulse, the sampling period must be shorter than the duration of the impulse. Therefore it is possible that higher sampling rate can improve sound quality for highly dynamic music.

Wow this is cool! I must say that Hearing Aid Helper understands what I’m talking about! Thank you. Also I must thank um bongo too. I have tried bongos thoughts already, and getting rid of the compression and setting the mpo has put me in the best spot i can be. It still doesn’t solve the problem of the mic being distorted. It is amazing how loud a crowd is when your playing a a honky tonk dance!! So without even playing they already distort the microphone. The only average of spl I have is that my spldb meter (which is a cheap radio shack digital) when I have it on it stays right around 100. it doesn’t go up or down a whole lot when we’re playing. I will see if I can get into a unitron with musician microphones. Also hearing aid helper would you email me direct? honkytonkjohn@live.com

Thank you all for the interesting debate! To clearer sound may we all be happy! One last thing. I use a starkey Davinci PXP. With a silver oxide battery. its the best combo I have found to date. If you have some thoughts on that it would be interesting to hear them. Thanks again!

As far as a know, microphone noise is only a problem at very low input levels. It is not a problem in music. Sampling depth limitation introduces noise at high levels and are therefore much more audible.

Arni i am not a technician so forgive my ignorance! Ok so are you saying that a microphone in a hearing ai doesn’t distort in loud music scenes, but it has a sampling issue? I do know that in quiter inputs, that the floor noise is louder. is this what your talking about? I guess what it boils down to is I’m not sure what your saying. A more lamens response may help my ignorance. thank you!!

Hi

Microphone noise is low and independent of the incoming sound, so it can only be heard when the incoming sound is low. For loud input level, it is simply masked and cannot be heard.

The sampling issue is a problem at very high levels, when the sound pressure is above the sampling scale. This kind of distortion can only be heard at very high input levels.

This is a microphone input distortion issue. This is a common issue for anyone who works in lots of noise, but most specifically a musician.

What Arni is speaking about, is the noise that you hear when there is very little sound around you (it sounds like a quiet hissing noise). This is no different than when you have a quiet stage full of live microphones and the PA system is live. There is an audible ‘hiss’ noise.

That said, microphone noise and even sample rate have nothing to do with what HonkyTonkJohn is asking about. His primary issue is microphone distortion when he plays live. Therefore he needs a microphone setup capable of higher SPL inputs. Similar to a kick-drum microphone. It is designed to withstand high SPL.

I am not debating the A2D limitations. However my experiences of setting up PA systems of all sizes and hearing aids for many many years, I’ve learned a thing or three about reading between the lines and getting to the real essence of what people are getting at.

I hope this helps all readers.

Cheers.

Sounds like this horse has already been beaten to death, but why not just physically dampen the input to the microphones and increase the gain appropriately to compensate for this? This is far from trivial to accomplish, but should resolve any input saturation issues.

Several hearing aid manufacturers include features in their software that map out the input being received by the aid, which could be used in determining how much each frequency band is actually being dampened (important to know if you want the end result to sound at all acceptable). Honkytonk had mentioned putting tape over the microphones, but this doesn’t sound like a very stable solution in practice.

They wouldn’t be horribly effective to use outside of work, but should allow you to work with a larger range of brands and models.

Most people don’t need dampening or high SPL microphones. Therefore, the manufacturers don’t build them as such. These high SPL microphones are placed in hearing aids by special request.

The software controlled dampening you speak of, is after the microphone input stage. It is controlled by the expansion, threshold kneepoints, compression ratios and output limiting controls. All of which would be ineffective to use if your microphone diaphragm is being warped by the impact of sound pressure.

As a relatively simple but questionably effective concept, putting tape over top of the microphones is actually the right thing to do. This is similar to using an ER musician’s filter in a normal ear to help the eardrum not go into physical distortion thereby making the sound clearer to the listener.

Cheers.

Correct me if I am wrong, but I do believe that everybody would benefit from a higher spl microphone!

No, not exactly. Most people don’t need to have this type of microphone.

Generally, the high SPL microphones are a little ‘tinny’ sounding compared to regular microphones, thus you have to correct the sound within the software by adding more low frequencies.

As mentioned before, most people don’t spend much time in 100+ dB sound environments. Most of them would get as far away from it as possible.

I think you may have misread my post. I was suggesting a workaround for converting a “standard” hearing aid into one that would work for Honkytonk’s purposes. If you take a 110 dB input and physically dampen it by 20 dB before it actually reaches the microphone then you’re effectively giving the hearing aid a 90 dB input, which it should be able to handle just fine.

Again this is far from trivial (you essentially need to create your own gain matrix through testing and properly map it to one that already exists in the hearing aid), but should work in principle.

Agreed. This is far from trivial, and far from easy to do. As of right now, the only tested and practical method that I know of currently, is to use high gain/high slope microphones to withstand the high spl levels for musicians.

Cheers.

Will you perhaps consider here that you are talking about a different type of technology.

Conventional mics use a ferrous core which has a given mass and passes through the coil wrapped around it. At high input levels the mass starts to lag behind the sinewave of the input signal, thereby creating distortion.

Hearing Aids use electret condenser mics with an incredibly low mass diaphragm made of gold coated mylar.

THEY DO NOT SUFFER INPUT DISTORTION EVEN AT EXPLOSIVE LEVELS.

All clear, good. :smiley:

Carry-on.

I can hear the distortion If I am subjected to high levels of sound, just turn down the volume and I can hear the distortion perfectly clear!!

Um Bongo, you missed the point. I will explain myself again. For all who are reading, I am speaking of hearing aid microphones, not conventional for stage use.

Since you brought it up, I should point out that any microphone, electret condenser or otherwise, will be subject to distortion when exceeding its build characteristics and specifications. Everything has operational tolerances.

You are correct, a hearing aid microphone is designed to withstand extreme conditions. These conditions the microphone manufacturers speak of, are related to the construction of the mic to withstand severe environmental conditions, high resistance to mechanical shock, RFI and EMI immunity, and overall good sound quality.

However, the microphones in most standard issue hearing aids were never designed to operate with low distortion at very loud input levels.
Therefore, a musician will likely get input distortion when using a standard hearing aid while playing an instrument on stage with his band, as he stands near numerous floor wedges and the drum set, while also being subjected to the front of house PA reflections from room acoustics and the sound of the crowd all going at the same time.

HonkyTonkJohn said it YET AGAIN that his hearing aids are producing an audible distortion, even after going through all your suggested tricks and tips.

What I am suggesting (though I am having a hard time understanding why you won’t accept it), is that there are all sorts of microphones that are used for hearing aids. Each has its unique characteristics. If you put the right combination of microphones into the right instrument, you will have a winning combination for HonkyTonkJohn’s application as was presented at the original post.

Now it is clear and all good.