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What's Left In Speaker Design To Reduce Distortion/Increase Detail Retrieval?

fineMen

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My assertion is that it can, in theory... but the recording techniques and equipment needed to do it are rare or nonexistent today.
Nope, neither nor. Not even with hearing aids it is possible. You're a romantic dreamer?
 

Blumlein 88

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I guess I should revise what I said there.


I mean a system that reliably, consistently, and somewhat independently of room reflections or any such considerations, (assuming a perfect recording with as many channels and 3D information as you might want) produces a truly illusory experience.

What I'm talking about: You are blindfolded... a jazz band plays a song in the room. They take the jazz band out of the room and play a recording of the same jazz band. You literally can't tell the difference. For the system to be considered ideal, IMO 90% or more of listeners need to fail an ABX between the recording and the real thing. This has never happened or even been attempted as far as I know, except maybe in very specific lab conditions.


AFAIK there is no such system today mostly because stereo recordings can't really do that, except in rare circumstances where the room happens to spatialize the recording in just the right way.

A truly ideal listening setup could do that every time, reliably, as long as the recording was of sufficient quality.

I think today drivers that are capable of making up such a system exist, but systems like that have been built rarely / never, nor have recordings like that been made very often, if ever.

I guess what I'm saying is I see plenty of room for audio tech to improve, but IMO voice coils are too big and power-hungry to build a consumer-friendly system that could do it.
Of course it wasn't exactly lab grade research. Some demo's have reportedly fooled audiences. Recordings of musicians were made in remote quiet hilltops so there were no reflections. Then on an outdoor amphitheater the recording of each musician was played thru a speaker at the same location on a stage. It apparently was pretty convincing even with the 1960's tech in use.

I've made up close recordings of individual musicians in a group. Played back with one speaker per musician and speakers placed where they were live it is very high quality they are here result. Yes, people say directionality of a speaker and instruments are very different etc etc. Yes true. But a single real audio source with no phantoms for each musician with speakers arranged similarly is enough to get you a big step toward sounding real. Sounding like they are right here in whatever room you set up this kind of playback.
 

kemmler3D

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Nope, neither nor. Not even with hearing aids it is possible. You're a romantic dreamer?
I'm saying with UNLIMITED TECHNOLOGY it's possible. Of course it is, that's almost a tautology. You think somehow your ear can tell the difference between real and fake if the sound waves reaching your ear are literally identical? Again, no, that's a tautology, it cannot.

The assumption that's debatable is whether such technology is actually possible. But that's a technical argument that I may not be fully qualified to participate in.
 

IPunchCholla

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Travis

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This question popped in to my mind simply from some recent experience listening to some speakers at another audiophile's place.

I currently listen to some smaller floor standing speakers with good quality drivers (Joseph Audio Perspective 2 Graphene) and I find there to be a gob-smacking sense of clarity
and detail in to recordings. Along the lines of "how could it get better than this?" (And I've heard lots of other speakers).

Then I go over to my Pal's place and listen to a pair of big ol' Estelon speakers, one of the newer "it" brands in high end audio circles. I forget which new model, but they retail for something like $65K. Now, most of us have had plenty of experiences showing us that money doesn't necessarily buy you any better sound in high end audio. But I have to say, even though the presentation ultimately wasn't to my liking as much as my own system, they just seemed to obviously dig out more sonic information in the recordings. So for instance drums on a track on my system would be well placed in spatial terms, and I can hear if the drums were placed in a reverb. But the Estelon speakers just seem to effortlessly carve out precisely where the drums are in the soundstage and the precise acoustics or added reverb around the drums...and exactly where that reverb "ends" is more vivid and obvious. Basically there is this constant sense of more sonic information, presenting more precision about what is in the recording.

Which had me wondering what accounted for these differences. Better drivers? The more heroic efforts that went in to removing the influence of the Estelon cabinets? The whole design?

Now, that's just accounting for why this question was on my mind. Anyone can simply ignore the above example (it's just my subjective impressions after all) but still get to the issue I'm wondering about:

What is left in terms of speaker design to achieve, in terms of lowering audible distortion and hence retrieving more neutral sonic information from recordings?

(I add "neutral" because of course one can always hype a speaker's high frequency response to increase perceived detail...that's not what I'm talking about).

Are we done? Or is there more to achieve in terms of materials and design (drivers, cabinets etc)? Is a very flat frequency response all there is (since resonances will purportedly show up in frequency response)? Or could we take a speaker that measures very even, yet some upgrade in driver material/design or even more reduction in cabinet resonances may yield even higher sonic performance, retrieving some subtle details that were obscured before?

Where can we go from here?
I think we’re gonna need a bigger boat.

My first thought led me to a kind of backwards approach to your question. What would the leader in speaker measurement technology utilized by speaker and transducer manufacturers say about this. Fortunately, we have some insight about this from Klipple’s paper in connection with his Heyser lecture ten years ago


Quoting the intro to his presentation:

The Richard C. Heyser Memorial Lecturer at the 134th AES Convention is Wolfgang Klippel. The title of his presentation plays on the challenge of engineering small loudspeakers that can also be loud, have high efficiency and low distortion. While the basic construction and principle of the moving coil loudspeaker has remained unchanged for a century, our understanding of its operation and limitations has increased very considerably. Modern signal processing techniques, coupled with careful design of the magnet, voice coil and suspension, can be employed to create small drive units that have an optimum trade-off between the crucial design factors. An understanding of the perceptual effects that arise from these design trade-offs helps to bring about loudspeakers that not only measure well but sound good. This is particularly important as compact devices such as mobile phones demand more and more quality from tiny loudspeakers in minimal enclosures.”

Have we reached the pinnacle some 10 years later where “Modern signal processing techniques, coupled with careful design of the magnet, voice coil and suspension, can be employed to create small drive units that have an optimum trade-off between the crucial design factors”?

I don’t think we are even close to that.

More importantly, is there a clear and universal understanding of the “perceptual effects that arise from these design trade-offs.” Because, according to Dr. Klipple, this will “bring about loudspeakers that not only measure well but sound good.” I take that to mean that just because a particular speaker measures well doesn’t mean that it will sound good. He agrees with Heyser that you still have to listen to them to see if the trade offs left you mud.

Travis

Here is the full abstract:

Full Abstract​

The loudspeaker is the weakest part in the audio chain. This statement might be true because the electro-acoustical transducer is one of the remains of the analogue era still using a moving coil and a diaphragm, much as it was a century ago. A more important argument is the low efficiency of the loudspeaker generating more heat than sound power output while adding undesired distortion to the output signal. The transducer, enclosure and other acoustical elements increasingly determine the size, weight and cost of the audio system because electronic parts become smaller or are replaced by digital signal processing.

At the moment there is no alternative transduction principle which is mature enough to compete with the conventional techniques. Therefore the moving coil as the best practice is preserved and interlaced with new ideas provided by research, design and manufacturing. This evolutionary process also results in significant progress in the absence of a revolutionary change. The development of smaller loudspeakers for personal audio equipment, automotive and professional applications is a visible example of this process. The generation of sufficient acoustical output at acceptable quality requires higher amplitudes in the mechanical system and pushes the working range to the physical limits.

The cultivation of large signal transducer performance is the central topic in this lecture. In the eighties the nonlinear and thermal modeling of the transducer attracted the interest of more and more researchers eventually resulting in a reliable theory. The theory has been evaluated by fitting the models to real transducers and measuring the large signal parameters dynamically. This identification technique requires only the electrical signal at loudspeaker terminals and uses the back EMF for sensing the velocity of the voice coil. Adaptive measurement techniques using the loudspeaker itself as sensor have been developed to monitor the behavior of the loudspeaker in normal applications (e.g. car) while reproducing music or other audio signals. Recorded parameter variation and state variables (e.g. voice coil displacement) reveal the ageing of the suspension and the impact of the climate.

The nonlinear and thermal parameters open a new way for loudspeaker diagnostics because these parameters describe the properties of the loudspeaker itself without the interaction with a stimulus (e.g. test signal, music). Numerical simulation tools (e.g. FEM) applied to magnetic and suspension systems show the relationship between material, geometry and the nonlinear parameters. However, only a measurement can reveal an offset in the rest position of the voice coil caused by suspension parts made out of fabric, rubber, foam and other visco-elastic material.

Operating loudspeakers at high amplitudes increases the risk of damaging the loudspeaker. The large signal model provides all state information (e.g displacement and temperature of the voice coil) indicating a mechanical and thermal overload of the transducer. In woofers the motor and suspension nonlinearity is used to limit the maximum peak displacement and to prevent the voice coil from bottoming. New measurement techniques have been developed to detect a rubbing voice coil, loose particles and other loudspeaker defects which produce impulsive distortion unacceptable to the human ear. The measurement of harmonic and intermodulation distortion cannot provide a comprehensive description of the large signal behavior as it reveals only symptoms of the nonlinearities depending on the choice of the stimulus. In contrast, the extended large signal model can be used to predict all state variables and the output signal for an arbitrary input signal. The effect of each nonlinearity and cooling mechanisms can be analyzed and design choices can be evaluated before the first prototype is finished. Novel auralization techniques make it possible to enhance or attenuate different kinds of signal distortion by a user defined scaling factor and to assess the audibility and impact on sound quality by listening tests or perceptual models. The closer link between physical and perceptual assessment opens new ways for defining the target performance of an audio product more accurately in marketing, developing products at optimal performance-cost ratio and for ensuring sufficient quality in manufacturing. This optimization process leads to more or less nonlinear transducers having a clearly defined (regular) nonlinear characteristic. Distortion may become audible for a critical stimulus but is accepted in a trade-off between maximum output, efficiency, size, weight and cost. The main objective in taming the transducer nonlinearities is the motor stability, to keep the voice coil in the gap and to gain maximum peak to peak displacement. Asymmetrical curve shapes of the nonlinearities are reduced to avoid a DC-displacement of the coil generated by a nonlinear rectification process which reduces efficiency and may cause bottoming of the voice coil.

Portable applications where power consumption and battery capacity is an issue will require transducers with the highest efficiency and minimal use of resources (e.g. neodymium-magnet). Such a “green” speaker is a highly nonlinear speaker using a motor topology where the voice coil exploits the magnet field in the gap and gives the highest force factor value Bl(x) at the rest position x=0. Unfortunately, the varying force factor Bl(x) generates significant intermodulation distortion throughout the audio band if the voice coil is displaced and windings leave the gap. This undesired side-effect of an efficient motor structure can be compensated by an inverse nonlinear preprocessing of the electrical input signal. The control algorithms use the large signal transducer model and identify the parameters adaptively. By merging electro-acoustics and signal processing the loudspeaker becomes a self-learning system providing optimum performance over the lifetime of the audio product. Certainly, this is not the last step in the evolution of the loudspeaker ...
 

Wseaton

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Going back to my hifi selling days.
You guys ever notice that the really, really high end speakers all tend to be effing HUGE? Either these massive horns you could rent as an apartment in tokyo, or huge arrays with stupid amounts of piston area or ribbon / planar hybrids.
Our ears are most sensitive to mid range....not the nuances of ribbon vs dome tweeters above 9k.
Detail retrieval in my book is all about mid range accuracy and transient response. Also, you cannot pressurize an average room with a pair of 6" voice coils mounted in box no matter how good the exotic bracing. This is why high end ear buds sound so good. Little piston throw, and they only have to pressurize a tiny space.
Stopped over at one of my audio buddies a few weeks ago and listened to his old martin logan ESLs. Having been stuck in cone driver hell for the past 10 years the ESLs were like moving from VHS to bluray in terms of midrange transparency. They may not be perfect and have other drawbacks, but it explains why I'm seeing all these crazy designs in the uber high end space that try to emulate planar radiation without the drawbacks.
A conventional speaker voice coil is a magnet flopping around in a magnetic field suspended by some stiff 'something' trying to force it back like a rubber band. Hence why big amps usually sound better than smaller ones to fight through the floor of magnetic sludge trying to reproduce tiny nuances in sonic clarity. Modern speaker coils have made marginal improvements in terms of reduced distortion and more controlled radiation patterns, but the biggest improvement is in terms of production profit margin.
So, you're stuck on cones for price reasons,,,fine. You guys have seen the Wilson Chronosonics, right? You could get a better performing design for a fraction the price by using vanilla paper and soft domes and by actively driving each module with an inexpensive amp loaded in each module and controlled by a central processor. Let the DSP in each amp control the time alignment...not pushing the drivers around. Gee, computers are good at that. It's like the joke about changing a lightbulb by turning the ladder. The point here is high end designs are meant to find a use for exotic amplifiers and gear and NOT be optimized for sonic retrieval. Engineers know how to reproduce a sound wave accurately. Marketing doesn't. Consumers want soundbars that cost $200 and reproduce a dozen discrete channels with 2" cones, then whine they can't hear dialogue if they turn it down. Want better sonic retrieval? Start with dumping classic cone designs in a monolithic wood box, and admit active designs are superior.
 

Axo1989

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Vanilla paper? I prefer rich chocolatey paper.
 

fineMen

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The assumption that's debatable is whether such technology is actually possible. But that's a technical argument that I may not be fully qualified to participate in.

Conclusion? Let's assume that no such tech/ is available today. You're a recording engineer, a musician to preserve some music for coming generations, what to do?!

Of course--I use the same analogy again which I'm serious about, a painting is something very different from what it depicts. It is comprised of clues for the eye to read.

Is a painting supposed to trick or fool the vision, is it to present optical illusions?

Please let's stop this; it would be hilarious to expect virtual reality from a recording, from mono to stereo to atmos, no way!

The painting is not an analogy, it relies on the very same principle: information transmission by abstraction. (To not understand this qualifies to be unqualified ;-)

***

In regard to the original question, what information is missing from a recording? What is the detail the question is referring to?

***

Anecdote; I have a recording of a male singer and a piano, some 'modern' stuff from Scandinavia. It was recorded using just and only one microphone in XY configuraton (ha ha, look that up). The stereo effect is to run away from. But since the post-processing was logically very limited the sound is as natural as it can be, wonderfull. The piano without the common shiny polish, a bit dull even, great, marvelous! The singer takes his place in the center, but naturally diffuse taking the reverberation with him when moving slightly. As subtle effect but remarkable even without exaggerated treble. The tonality in total is just right as if it were a live performance.

***

First get tonality right not with the speakers, but with the eff**ing recording / post-processing, tone it down! Withstand the urge to reveal "detail" (sic!) by emphasis of XYZ. Reduce stereo effects to a minimum!

The problem with bad sound starts literally with the recording targeting people who want fine "detail" and big "stereo". It makes the engineer bow and bend here and there until his "painting" is a sheer mess.

That's why I like pedestrian old recordings so much. A natural, pleasing and satisfying tonality--if it wasn't "remastered".
 
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Wesayso

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Of course it wasn't exactly lab grade research. Some demo's have reportedly fooled audiences. Recordings of musicians were made in remote quiet hilltops so there were no reflections. Then on an outdoor amphitheater the recording of each musician was played thru a speaker at the same location on a stage. It apparently was pretty convincing even with the 1960's tech in use.

I've made up close recordings of individual musicians in a group. Played back with one speaker per musician and speakers placed where they were live it is very high quality they are here result. Yes, people say directionality of a speaker and instruments are very different etc etc. Yes true. But a single real audio source with no phantoms for each musician with speakers arranged similarly is enough to get you a big step toward sounding real. Sounding like they are right here in whatever room you set up this kind of playback.

John Dunlavy did such tests (live performers vs speakers) to test the accuracy of his speakers, but he did use an Anechoic chamber for that (so, Lab conditions) to get rid of the problem of reflections. He stated what's needed for accurate speakers in this interview and in his short paper: Loudspeaker Accuracy. I'll upload that document to this post.
 

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fineMen

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John Dunlavy did such tests (live performers vs speakers) ...
You may discard his findings: "The live string-quartet was recorded in one of DAL's two, large anechoic chambers (24'L x 20'W x 16'H), using a matched-pair of instrumentation-quality, 1/2" omni-directional mics and a professional quality DAT recorder."

No hint on the stereo technique being used, there are many. As if the tonality of the two xyz mikes was remotely comparable to the human ear's perception--anechoic may help with that. From my personal perspective a quite rushing, naive attempt. that would benefit from refinement a lot.
 
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FrankW

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"If we want to be more sure of any such conclusions, the ideal way to evaluate would be blind listening, for instance similar to the type cited by Floyd Toole."
Yes, you retroactively tried to cover by posting this long after I pointed out the folly of your original post AND the thread title itself, spelling out clearly the audiophile concoction of "detail retrieval" IS a speaker issue, not a:
No mistake. 100% dismissed all science to account for differences.
1) You.
2) Zero controls for biases, like price, materials, cabinet's, etc, etc, etc.
3) Zero cognizance of volume/spl affecting everything, including "detail retrieval" and other audiophile contrivances
4) Zero cognizance of vastly different rooms/subsequent modal exactment, possible +/-20db variation
The latter 2 having zero to do with the speakers per se.
The Klippel paper I linked (unlike all others) made clear controlled listening was mandatory for any "investigation" into speculations.
Your fellow Canadian gets it eh? Summarized your OP nicely here :)
It's ok Matt, some people will just never get that "Subjective" isn't what they think it means.
 

FrankW

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I mean a system that reliably, consistently, and somewhat independently of room reflections or any such considerations, (assuming a perfect recording with as many channels and 3D information as you might want) produces a truly illusory experience.
What I'm talking about: You are blindfolded... a jazz band plays a song in the room. They take the jazz band out of the room and play a recording of the same jazz band. You literally can't tell the difference. For the system to be considered ideal, IMO 90% or more of listeners need to fail an ABX between the recording and the real thing. This has never happened or even been attempted as far as I know, except maybe in very specific lab conditions.
No "blindfolds" are needed in 99.9% of "blind" testing. A scrim in front of both band and speakers simultaneously would suffice. But you've missed the point entirely.
It was possible to create something (JJs demo, which BTW, there is a new one that can be visited) that sounded very very real to 2 audiophiles (no lack of "details retrieval", in fact "details out the wazoo") who would (via Stereophile positions) have heard every magic material "$65k" "detail retrieving" type speakers on earth at the time, far more "real" than any of that nonsense, with rather pedestrian speakers. Using what would be (to audiophiles), vastly inferior cabinets, drivers, materials, etc, with highly likely more "distortions" than todays magic drivers, materials etc, etc.
How would one with such beliefs (99% of all audio forum posters?) reconcile this?
 

Blumlein 88

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No "blindfolds" are needed in 99.9% of "blind" testing. A scrim in front of both band and speakers simultaneously would suffice. But you've missed the point entirely.
It was possible to create something (JJs demo, which BTW, there is a new one that can be visited) that sounded very very real to 2 audiophiles (no lack of "details retrieval", in fact "details out the wazoo") who would (via Stereophile positions) have heard every magic material "$65k" "detail retrieving" type speakers on earth at the time, far more "real" than any of that nonsense, with rather pedestrian speakers. Using what would be (to audiophiles), vastly inferior cabinets, drivers, materials, etc, with highly likely more "distortions" than todays magic drivers, materials etc, etc.
How would one with such beliefs (99% of all audio forum posters?) reconcile this?
J_Js method was quite something. A shame it was not allowed to develop commercially. It also was pretty simple.
 

FrankW

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J_Js method was quite something. A shame it was not allowed to develop commercially. It also was pretty simple.
Gives me hope that my pedestrian F36s are just fine, no need for Estelons and magic "low distortion" woofers et al.
Until real subjective evidence to the contrary ;)
 

tuga

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The obstacles to fit for purpose loudspeaker blind AB testing are insurmountable. You can cut corners and do a shuffler or an anechoic recording and have listeners compare that with headphones.
 

gnarly

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I've had success in substantially increasing the detail and clarity in DIY speakers vs any commercial speakers I've yet to hear,
pursuing the following set of objectives in no particular order...

*An acoustic design that minimizes center-to-center spacing between drivers that share a common frequency range.
Goal is to stay within 1/4 WL throughout all summation ranges.
*Strive towards point source.
*A flat frequency magnitude response on-axis (quasi-anechoic). (Any house-curve preference applied afterwards.)
*Polar responses that reflect smooth pattern control of the on-axis flat response,.
*Constant directivity pattern control.
*Maintain that pattern control as low in frequency as possible. Goal is down to Schroeder.
*Linear SPL throughout the spectrum, including headroom for peaks, at desired maximum average SPL.
*Match driver sections' linear SPL capabilities, all staying easily within Xmax
*Sufficient subwoofer and low-mid sections displacements to maintain that SPL linearity, preferring cone area (Sd) over excursion (Xmax)
*Minimize modulation distortion of higher SPL by increasing the number of 'ways' employed.
*Each driver section individually amped and DSP processed.
*Flat phase response to reduce phase rotation and eliminate group delay apart from the bottom end rolloff.
*Delays between driver sections that equal, and only equal, the Z-axis distances between acoustic centers.
*Minimize lobing potential between driver sections via steep complementary linear phase crossovers.

Ok, maybe just a lot of technobabble for most I imagine.

But my hopeful point is this....if a numbnuts amatuer DIY speaker builder like me, has learned things that really do improve detail & clarity,
.....many/some of which are missing from the vast majority of commercial offerings.....
how is it possible there are not many speaker improvements still lying on the table?

Maybe a rebuttal is: "you just think you are hearing improvements because you are so invested into DIY."..
Good point...I wonder it myself at times.
But two things give me solace. First, I still have some highly regarded commercial stuff to compare to, along with listening to other folks'/stores'/shows' systems.
And second, probably 90% of things I try go nowhere...I fail at improving a LOT. Keeps the egoic investment down....

Good headphones are the only thing I've heard with greater clarity and detail than latest DIY speaker build.
But they of course can't begin to provide the sound of powerful bass and bass transients, that speakers can. The visceral experience.


Oh, a few suggestions for evaluating clarity and detail.
If your speakers and amps are up to it, take one side of the stereo outdoors and listen without the room. Bass output might be weak, but it can still be quite ear opening how much clarity usually improves. Back to the almighty importance of a room, huh :)

Another room test...put on sealed back headphones and listen to tracks playing through the speaker(s), but with a measurement mic providing the signal to the phones.
Shows the drastic amount of brain/ear processing vs what a mic measures.

Last, if you can do the setup outdoors, make a recording of a song played through the speaker. Play that recording, and while evaluating the difference vs original, record the recording. Play the 2nd recording, repeating the process. See how many recordings survive listenability. Typically only one or two.
(credit for both tests to posts by TD of DSL)
 

Timcognito

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I've had success in substantially increasing the detail and clarity in DIY speakers vs any commercial speakers I've yet to hear,
pursuing the following set of objectives in no particular order...

*An acoustic design that minimizes center-to-center spacing between drivers that share a common frequency range.
Goal is to stay within 1/4 WL throughout all summation ranges.
*Strive towards point source.
*A flat frequency magnitude response on-axis (quasi-anechoic). (Any house-curve preference applied afterwards.)
*Polar responses that reflect smooth pattern control of the on-axis flat response,.
*Constant directivity pattern control.
*Maintain that pattern control as low in frequency as possible. Goal is down to Schroeder.
*Linear SPL throughout the spectrum, including headroom for peaks, at desired maximum average SPL.
*Match driver sections' linear SPL capabilities, all staying easily within Xmax
*Sufficient subwoofer and low-mid sections displacements to maintain that SPL linearity, preferring cone area (Sd) over excursion (Xmax)
*Minimize modulation distortion of higher SPL by increasing the number of 'ways' employed.
*Each driver section individually amped and DSP processed.
*Flat phase response to reduce phase rotation and eliminate group delay apart from the bottom end rolloff.
*Delays between driver sections that equal, and only equal, the Z-axis distances between acoustic centers.
*Minimize lobing potential between driver sections via steep complementary linear phase crossovers.

Ok, maybe just a lot of technobabble for most I imagine.

But my hopeful point is this....if a numbnuts amatuer DIY speaker builder like me, has learned things that really do improve detail & clarity,
.....many/some of which are missing from the vast majority of commercial offerings.....
how is it possible there are not many speaker improvements still lying on the table?

Maybe a rebuttal is: "you just think you are hearing improvements because you are so invested into DIY."..
Good point...I wonder it myself at times.
But two things give me solace. First, I still have some highly regarded commercial stuff to compare to, along with listening to other folks'/stores'/shows' systems.
And second, probably 90% of things I try go nowhere...I fail at improving a LOT. Keeps the egoic investment down....

Good headphones are the only thing I've heard with greater clarity and detail than latest DIY speaker build.
But they of course can't begin to provide the sound of powerful bass and bass transients, that speakers can. The visceral experience.


Oh, a few suggestions for evaluating clarity and detail.
If your speakers and amps are up to it, take one side of the stereo outdoors and listen without the room. Bass output might be weak, but it can still be quite ear opening how much clarity usually improves. Back to the almighty importance of a room, huh :)

Another room test...put on sealed back headphones and listen to tracks playing through the speaker(s), but with a measurement mic providing the signal to the phones.
Shows the drastic amount of brain/ear processing vs what a mic measures.

Last, if you can do the setup outdoors, make a recording of a song played through the speaker. Play that recording, and while evaluating the difference vs original, record the recording. Play the 2nd recording, repeating the process. See how many recordings survive listenability. Typically only one or two.
(credit for both tests to posts by TD of DSL)
Send one to Amir for testing. "One test is worth a thousand expert opinions"
 

kemmler3D

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No "blindfolds" are needed in 99.9% of "blind" testing. A scrim in front of both band and speakers simultaneously would suffice. But you've missed the point entirely.
It was possible to create something (JJs demo, which BTW, there is a new one that can be visited) that sounded very very real to 2 audiophiles (no lack of "details retrieval", in fact "details out the wazoo") who would (via Stereophile positions) have heard every magic material "$65k" "detail retrieving" type speakers on earth at the time, far more "real" than any of that nonsense, with rather pedestrian speakers. Using what would be (to audiophiles), vastly inferior cabinets, drivers, materials, etc, with highly likely more "distortions" than todays magic drivers, materials etc, etc.
How would one with such beliefs (99% of all audio forum posters?) reconcile this?
I think I haven't made my point clear, sorry.

I'm approaching the overall question in this thread as as "At what point would there be no point in improving speakers further?"

That point would be when speakers can perfectly reproduce sound - no practical limit on dynamic range, distortion, frequency response, OR spatial information.

As you and others have pointed out (and which I acknowledged in my first overlong post in this thread), this is already more or less possible with controlled conditions.

However, the ideal speaker system could do all of this in an average or even below-average room, within an average consumer's budget, with the speakers in suboptimal positions.

That's what I am saying is not possible today, and may not be practical with voice coils at all. VC-based speakers are too big and will probably remain too expensive to really do this, Atmos systems are already quite a stretch and PITA for Joe 6-pack. I don't know many people that can fit 10+ speakers capable of 20hz-20khz @ 115dbSPL in their living rooms...

There is also the large, but separate question of what recording technique might be considered to perfectly preserve a live performance, perhaps even including something like a volumetric measurement of the sound field... obviously there isn't really anything like that in use today.
 

gnarly

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Send one to Amir for testing. "One test is worth a thousand expert opinions"
Lol

I'll send this latest DIY(white) , or its predecessor (blue) along with the DSP processor that runs their design, along with the needed amp rack.
Easy peasy to send along to Amir, huh? :)
syn11 ported.jpgsyn10 ports close.jpgsyn11 qsys schematic snip 12-14-22.JPGrack rear.jpg

The processor and amp rack are set up for simultaneous speaker setups for A/B testing of other DIYs and commercials.
Of course doesn't need to be so complicated. Could all be put into some plate amps. So kinda ignore them, other than they are essential for my work.


BUT, pls don't ignore the large speaker....that's what I've found it takes to satisfy the acoustic design goals i listed
And I guess this is the real rub....since when will home audio embrace such large horn speakers?
Aesthetics appear to rank higher in home audio than sound. (not at all meaning to downplay realworld room constraints)

I guess in reality, the biggest single design factor i can think of for home audio to be able to increase detail and clarity, is to accept lowering the importance of aesthetics.
Like the new Genelec kinda did...big step forward for SQ I think...
 

Leif

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Of course they exist. But even when I was in a speed metal band in the 90s, we were using a 4 track to record and weren’t playing together. The song was simulated. There was no single performance being recorded. We even put together parts out of multiple takes. Now, a lot of music isn’t even recorded, but generated.

When I am talking about the vast majority of music, I am thinking in terms of all of the music listened to. What percentage of music do you think classical makes up? Even rock is a small fraction. Apple Music has/had a feature that let you listen to the 25 favorite songs being listened to around the world. Almost all of it was rap, the last I checked. None of it was recorded as a single take of a performance where the listener was expected to feel there was just a flimsy curtain between them and the performance. In fact, I would say except for vocals, much of it wasn’t recorded at all.

The second part of it is that it isn’t being mixed to sound like a recording of a single event in time and place. The locations of the instruments aren’t being put in a precise place. Instead panning and reverb effects are being used as creative tools. Drums sometimes have different toms, cymbals, snares sounding like they are in places that make no sense in terms of where the other parts are And sometimes each sounds like it is in a different space, reverb-wise. I’ve even heard things like the attack on a guitar being spaced differently than the fundamental. DAWs are now as much a creative tool as folks playing instruments.

So, no, for most music I don’t think there is meant to be a sense for the audience of “being there”. The only ”there” is you the listener right now listening to the song.
I can’t speak for others. I listen to a broad range of music, and yes I do want to have the sensation that I am present at the performance. Yes I know that for some genres the recording can be a mix, the single version of Silver Machine by Hawlkind had Lemmy‘s vocals instead of the original ones, because his voice worked better.

Of course there is an illusion, and I tend to avoid recordings where the illusion is poor. I certainly can have the illusion of being there with jazz, classical, folk, rock, and other genres.
 
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