Who makes the lowest distortion speaker drivers

2 grams, that's hideously heavy compared to air. Electrostatic speakers are a tiny fraction of that and much better approach matching the weight of air.

Some of the most prized drivers in the sub forum clock in near half a pound*. Think I'm kidding?

Silly to make make sound in air by shaking heavy cardboard (or even lighter aluminum attached to heavy coils of wire). What kind of absence of distortion can you expect when you are trying to accelerate (and decelerate) a half-pound object? C'mon. Think RePhase is going to help you?

Ridiculous to debate distortion in heavy-cone speakers. They are all dinosaurs compared to true horns (not too feasible, eh) or electrostatic speakers or all the concepts yet to be widely commercialized (such as motional feedback, plasma drivers........).

B.
*check it out: 227 grams

Oh dear.

Ben, you seem to have forgotten all about Physics. Air weighs 1.2kg per cubic metre, or 1.2g per litre. Saying something is heavy compared to air is just crazy. How much air?

How much distortion do you expect when you try to accelerate something weighing a few tonnes?
Like a car, or maybe a newspaper printing press. To accelerate anything, you need force. Heavier things need more force (Newton's 2nd). So long as the driver's motor is enough to control a heavy cone, what's the problem?

It's interesting that you say they're dinosaurs. Can you show me an old-fashioned light-coned driver that'll do what this can? Data-Bass
The moving mass is a kilo, by-the-way, but the output density (noise per cubic foot) is like nothing else. It'll also sink a couple of kilowatts with 1dB of compression.
Distortion measurements are there, too, and I suspect any of your preferred drivers would've turned themselves inside-out before putting up the SPLs that driver can in a sealed box.

Chris
 
Does a mem array will suffer from spatial aliasing ?

We sold our mic business so I have no one to ask anymore. We did a 31 element array and I assume it was compact enough to still be smaller than the popular 1" mics. True story (as I was told) - Sales arranged a test along with one of the large European mic manufacturers at a recording session with Rihanna. She asked to keep the only prototype because she thought it was "perfect" for her voice and they were too embarrassed to say no.
 
We sold our mic business so I have no one to ask anymore. We did a 31 element array and I assume it was compact enough to still be smaller than the popular 1" mics. True story (as I was told) - Sales arranged a test along with one of the large European mic manufacturers at a recording session with Rihanna. She asked to keep the only prototype because she thought it was "perfect" for her voice and they were too embarrassed to say no.

I'm perhaps too curious but, could you name your company please ?
I'm also often victim of this strong cognitive bias, my preference goes on prototypes because this is an adventure.

PS : 5 years of apple "think different" ads on the TV... our poor brains.
 
I would say that performance wise that would be true, but cost wise, not even close. I can't imagine how low cost MEMs have gotten to be, but it can't be much based on the volumes and the forecasts.

I used to argue that MEMs mics could never reach the performance of a larger electret (which is clearly true.) But then I was thinking only of a single one-v-one application. I did not envision the low cost that they could reach in very high volumes such as today and allowing the use of an array of them.
 
And in the bass areas, its not only harder to hear the distortion, but most people actually prefer it.

I disagree with the first part. My main experience with sound is live audio. When mixing on low distortion subs the difference is immediately apparent, if you know what to listen to. First off, the bass notes are more readily noticeable one to the next. Each note the bass player plays is distinct. But more importantly are the harmonics. Since a subwoofer has higher sensitivity in it's higher range, this means the harmonics are in the higher sensitivity area. Subwoofer harmonics can thus gum up the vocal range. Low distortion cones, like BMS with shorting rings will lead to a cleaner mid range. Also band pass lows will be do the same, as the harmonics will get *passed* out, since the box will not reproduce them.

To the second part, this is unfortunately true in live audio, to the uninitiated. We did sound for a country rock band one time and the sound man did not like the sound of the low distortion subs. We added a small front loaded powered speaker to the sub output, with only hundreds of watts, compared to the thousands of watts in the main sub horns. The powered speaker was set to go into clipping every time the drummer hit the kick drum. This gave the sound man the *guts* he was looking for, and he was happy.
 
Back to the question at hand: Why is THD wrong?
A loudspeaker will typically have large amounts of 2nd harmonic, but this harmonic is almost always completely masked. Hence we do not detect a problem with a loudspeaker that has 15% 2nd order nonlinearity.

On the other hand an amplifier with crossover distortion generates orders as high as ten or twenty in the spectrum. These very high harmonics are not masked at all, which is why we can hear as low as .1% THD (or less, depending) of crossover distortion in an amp. How can one thus use a measure like this that has no correlation to what we hear to "rate" the audible nonlinearity of any audio component? We can't!!

I think this right here is key. Making, and the fact that it is not linear, but a slope. As long as the distortion is under the slope, it is masked, or un-noticed. The distortion in the second example does not fall under the slope, but rises above it, and is noticeable.
 
So a woofer with five or more percent distortion might not be audible, but a midrange with 5% might be awful, and a tweeter with 5% could be lethal.
Peace,
Tom E

I would quibble with this based on crossover points. Yes, distortion would be more noticeable the higher in frequency one goes, but, the harmonics from the midrange might fall into the highly noticeable range, and the harmonics from the tweeter might fall into the above human hearing range, and thus, not lethal.
 
scott wurcer seems clearly suggest that they can't compete with large membranes microphones ?
Why ?

For a single MEMS element it's just physics of scale the mechanical impedances i.e. loss mechanisms scale just like any loss (resistive for instance) while the generator capacitance falls (as area). The net result is that given real materials that you can actually fabricate something out of there is roughly a square root relationship between noise and size (1/2 the area 3dB worse noise).

Distortion is more complicated, early MEMS mics used a MOSFET in deep subthreshold and biased the input with back to back diodes. This means there is a hard max limit on input voltage from the generator, we got out of the business before completing some experiments on true charge amplifiers at the input. IMO the MEMS guys only had to target the cheap Panasonic type capsules and when they equaled the performance they were done and many of the conventional capsules have gone by the wayside.

I think we are mixing in the enormous market for voice quality mics with recording mics. There is little value added by increasing the performance in this market which is absurdly huge, every device there is has a mic these days. The same is true of the cameras in your car vs a top DSLR. The car-cams probably are well under $1 and produced by the 100's of millions.

EDIT - I'm retired from ADI, and we used to make MEMS mics but no more. I was never in that group just consulted internally on occasion.
 
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I understand why folk eroneously think this way. They see the cone as the source of the sound and since the cone moves relative to the observer, there should be a Doppler shift. But there is a subtle issue here that is missed.

The Doppler shift occurs when there is movement of the source relative to the observer because the peaks of the sound waves are no longer separated in time by the same amount as when the source is at rest. When the source moves closer to the observer the pressure waves bunch up, when the source recedes the pressure waves stretch.

In the case of a pistonic speaker cone the pressure waves are generated when the cone moves back and forth relative to the fixed magnet in lock-step with the applied electrical signal. Multiple frequencies can be present in the signal, each one creates movement of the cone relative to the magnet. These frequencies do not mix in this linear system. It doesn't matter if the cone is slowly oscillating to a low frequency electrical signal at the same time as it wiggles to a higher frequency electrical signal. The pressure waves generated at the higher frequency will be regularly spaced in time and not modulated by the lower frequency.

The reference point for the speaker as a sound source is the permanent magnet. If the whole assembly were to move relative to the observer, then you would have a Doppler effect.

Put the speaker inside a black box, nobody knows how it generates it's sound except that it is a linear device. If the black box is stationary, there is no Doppler effect. If the whole box moves, then the frequencies it emits will all be subject to the Doppler effect, depending on the relative motion to the observer.

I'm afraid Rod didn't see clearly that the sound pressure waves are generated from the motion of a cone relative to the fixed magnet, regardless of the excursion of the cone caused by other frequency components providing that it operates in a linear range (Xmax anyone...). When the cone has a large positive excursion it wiggles at high frequency in lock-step with the applied high frequency signal just as well as at large negative excursions - the frequency of these wiggles dictated by the applied signal acting against a stationary fixed magnet. There is no modulation or change in the frequency of these high frequency signals. No Doppler effect.

THIS!

I think one thing that confuses people is that they are used to thinking of everything in sound as separate, when everything is combined. The ear only hears one signal. The brain separates it into different components.

If a piccolo and a bass drum are being played on a stage, there is not two separate sounds that meet the ear, there is one complex sound, and the brain separates them, and identifies for itself what portion is the piccolo, and what input the bass drum has.

These instruments also have the advantage of zero distortion, in the fact that they are the source. If you cut a slash in the bass drum, and it rattled, it would still not be distortion, but source.

The speaker system is the exact opposite, in that it has to get everything exactly the same, or it is distortion. The sound quality, by definition, can only degrade, it cannot improve. Any *improvement* is itself distortion.

I think there are lots of non-linearities that can be measured, that are chalked up to doppler, but are in fact other forms of distortion.

If you were to look at the instance of the piccolo and the bass drum, there would be huge wiggles in the signal when the bass drum was pounded, and the piccolo notes would look to change, when in fact if the piccolo pitch were measured, it would not change, it would just *ride* on the bass drum poundings. The same happens when amplified. Same sound, only louder. The measured distortions are not doppler, but other factors.
 
The reference point for the speaker as a sound source is the permanent magnet.

There was a reference early in this discussion to alnico, and it's timbre. This is an underdiscussed portion of this thread. There is a big difference in magnets. Not only in flux density, but in variations in flux throughout the power band. Alnico is known for having very stable flux density.

Also the main attraction for field coils is the fact that they have very stable flux density. There is a discussion going on about microphones, and I am surprised that this has not been compared to speakers. It is well known that generally speaking, dynamic mics do not have as much precision in movement, (not frequency response), as much definition, as condenser mics. This is the same factor. The condenser mic has a 48V (typically) rail, that is applies to a coil in the microphone. This is a very steady field for the diaphragm of the mic to work against.

The best sounding speakers I have ever heard were field coil speakers, and they were not the flattest frequency, nor the widest frequency response speakers I have heard. The biggest downside to field coil speakers is heat dissipation.