Hypercube Loudspeakers

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Quote xrk971..

"For measurement I use an old laptop PC (actually a small netbook), REW software, a Dayton UMM-6 USB mic. I use a small class D amp (TPA3116D2) between the netbook and the speaker. If you can get the Dayton DATS v2 - it can be very useful for impedance measurement of speaker and TS parameter measurement.

I assume you have a PC and an amp already. You just need the $70 mic and you are set. Other amps such as a $5 TDA7297 work well too. Just needs to have sufficiently flat output to drive the load from your speaker from say 30Hz to 20kHz.

Let me know if you need more tips. That REW software is fantastic. Can't believe it's free.

Good luck"


This information is some of the simplest, low tech, easiest to understand directions that I've seen on here! Sure takes some fears away from a newbee that doesn't know anything and yearns to run with the big dogs..

A sincere Thank You, Sir..
 
You are welcome. You will also need cables to connect the laptop to the amp (3.5 mm stereo to RCA etc). Follow directions on installing the driver for the mic before plugging it in for the first time. USB can be finicky at times. It helps to have a camera tripod to mount your mic and position it precisely where you want. Also elevate your speaker on a stand or stool, aim it 45 deg relative to room walls, and keep it away from walls for measurment to reduce effects of wall reflections. If you measure at 1m you usually get a floor bounce cancellation dip at 150Hz. Measure at 0.5m instead. Put pillows on floor between the mic and the speaker. Set your amp level so you don't clip the mic. Lots of things to try but not hard to do.
 
Very interesting link....Its great to 21st century R&D and computing power used to debunk so much of the 19th century (!) and early 20th century experiments that are still raised aloft by the vast majority of the loudspeaker industry and DIY community....

Perhaps I'm missing something, but computer animation aside the link appears to contain nothing that would surprise the likes of, say, Rayleigh, Scott Russell, Olney, Olson, Beranek, and the vast number of highly competent RF engineers from the late 19th & first half of the 20th Century. You know -the ones who were paid 6-figure salaries (in contemporary terms, not adjusted to modern equivalents) in the middle of the Great Depression, and were given essentially unlimited resources to build & measure everything they could. It's pretty standard & nothing you can't find in a half-decent textbook, the hyperphysics site and its equivalents.
 
Time domain Vs freqency domain

Hi Scott,

I don’t want to drift off the main Hypercube topic but the point that you ask for clarification on is such a fundamental point, it effects all loudspeakers / cabinets / drivers....Its well worth repeating.

The key point the paper explains graphically & technically is that sound travels in a series of Time Constrained transients (compression & rarefactions in the air pressure) and not as frequency (amplitude) constrained sine waves.

The repercussions of this fact are dramatic...100 years of barking up the wrong tree....Blindly following the pioneering but limited research and flawed conclusions of many of the early scientists you site in your last post....They had very limited resources....No spectral analysis / real time FFT / computers / 3D imaging / understanding of radar / microwaves / and absolutely no idea of the subtle workings of the ear brain biology and psychoacoustics....

Put another way, Scott, would you expect your doctor to diagnose a medical condition using research from the 1920's....?

This fundamental fact has eluded so many for so long that the entire loudspeaker industry has "grown up" based on frequency domain based criteria....No one ever dares to challenge the old school thinking....This is wrong!
More detailed information on this subject is provided in John Watkinson's latest two papers that I have attached a few posts ago....If you take the time to read them I am sure all will become clear.

You will not like the truth as it exposes the inherent failings in conventional "push pull" drivers and in transmission line, ported, back loaded horns and all delayed resonance systems.
The question is can you handle the truth?😉

All the best
Derek.
 
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This might be ok?
D.

Overkill,
Thanks for the article. Interesting read and I don't disagree that getting time transients correct is critical to a good speaker. However, the author seems to dismiss all speakers that have acoustical time delay - this is not necessarily bad for example in the case of a tapped or FLH horn sub (which can be designed to have a clean impulse bass response but delayed in time) that can be used with tops that have time delay applied via DSP so that the total signal is integrated and time coherent. One of the few speakers that can achieve loud SPL and reproduce square waves is Danley's Synergy. Given that it also acts as a point source that is phase and time accurate probably is one reason why it sounds so good.

Btw, back on topic but related to your quest for time accurate speakers with good transient response, check out the latest measurents of the HyperFAST. Applying a Linkwitz transform to a sealed cabinet keeps the nice low group delay of a sealed cabinet while giving some needed bass extension. The impulse, step, and phase response are all pretty good too. Nothing to do with Hypercube as the reason it works, but Hypercube does provide a relatively stiff and internally reflection free cabinet:
http://www.diyaudio.com/forums/full-range/265915-hyperfast-hypercube-based-2-way-4.html#post4147908

Phase:
453117d1418188662-hyperfast-hypercube-based-2-way-hyperfast-linkwitz-phase.png


Impulse:
453118d1418188662-hyperfast-hypercube-based-2-way-hyperfast-linkwitz-ir.png


Step:
453119d1418188662-hyperfast-hypercube-based-2-way-hyperfast-linkwitz-step.png


Group delay:
453120d1418188662-hyperfast-hypercube-based-2-way-hyperfast-linkwitz-gd.png


XO used:
453116d1418188662-hyperfast-hypercube-based-2-way-hyperfast-linkwitz-xo.png
 
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Overkill,
Thanks for the article. Interesting read and I don't disagree that getting time transients correct is critical to a good speaker. However, the author seems to dismiss all speakers that have acoustical time delay....

Precisely...This in fact correct!

I am sorry to be the bearer of bad news (please don’t shoot the messenger!) but you have hit the nail on the head.
The only way we hear air borne sounds (apart from bone conduction) any sound is this:
One fleeting transient (compression in the air) followed almost instantly by a smooth decay back to ambient air pressure (rarefaction).
This holds true at all frequencies.

Any delayed resonance from ports/ horns / transmission lines or from oscillating cones on rubber surrounds are detected as gross distortions by the ear / brain.
They are detected as unnatural or “ghost echoes”, not natural echoes of the original transient like room / wall reflections or echoes in a cave or echoes bouncing of trees or rocks.

All delayed energy transients are by definition time domain distortions.
The human ear / brain has evolved to place the highest priority on identifying where sounds originate from….Not what sounds are, that is a lower priority.
The only way we can locate where sounds originate from in 3 D space is by comparing the difference in arrival times of the transient ( air compression) between our left and right ears.
This process involves miniscule movements in the ear drum and the brain must accurately process tiny time differences in real time….Astonishing biology!
Our ear / brains have evolved over millions of years to process this information, in all that time and in all the natural world there has never been any other way to generate sounds ( fleeting transient followed by almost instant decay) so in order to survive, we evolved to decode the air pressure transients by focusing on location not identification….This was to avoid being eaten by a predator….Where is that snapping twig sound coming from….
Sorry if I am ranting on about the biology, but there is no point in addressing any other of your points as everything is based on the one simple fact.

All natural sounds are produced using one method….One clean near instantaneous transient with rapid decay back to ambient, this is the only way we hear sound….This is also the only way to reproduce natural sounds.

Hope this helps
All the best
Derek.
 
Derek, please cease patronising me (and everybody else who does not happen to automatically accept your statements). Can I handle the truth indeed... yes. Very well. The basic physics involved for loudspeaker enclosures has been a well understood science for decades, whether you like it or not. To claim otherwise is frankly twaddle. Likewise to imply that the majority of people here (presumably with the exception of yourself) are labouring under the illusion that there is no compromise involved in xyz, be it a drive unit, an enclosure or anything else. Guess what: we do. Really quite well in many cases, you will no doubt be delighted to learn.

Transient perfect? A laudable goal. Qtc = 0.5 sealed box, or a very low tuned EBS with effective control over the diffraction signature is as good as you're likely to get in practical terms as far as the enclosure is concerned, since it is tracking the signal as accurately as is possible for the unit in question & is neither under nor overdamped (which are both inaccurate), with a single let-go point.

Any delayed resonance from ports/ horns / transmission lines or from oscillating cones on rubber surrounds are detected as gross distortions by the ear / brain.

...is such a grotesque oversimplification that without a vast amount of qualification it falls into the 'nonsense' category. 'Gross distortions by the ear / brain'? And your detailed peer reviewed research with its parameters, representative sample sizes etc. are? How is this 'gross distortion' objectively quantified? In percentage terms? To whom? At what frequencies? Does it vary with these (a rhetorical point since perception of GD in humans manifestly does, a well known, unremarkable and proven fact which rather contradicts your claim above)? What other factors come into play, and at what point does it become the dominant factor?

Now, I doubt anybody would argue that transient perfect, or as near as can be achieved, is a laudable goal. Of course it is. Likewise the minimising of diffraction artefacts etc., which are related given that their time-delayed smearing are closely related to the aforementioned. Ditto for reductions in other forms of distortion. But you are damaging your own case above by reducing it to such over-simplistic levels that it loses touch with practical engineering (and biological) realities. That's the truth. How you proceed is up to you, although no doubt the university department you are working with will assist on that score. As a university lecturer myself, I am familiar with how the departments proceed, and will look forward to reading my colleagues research findings. You have every right to be proud to work with an illustrious acoustics / engineering department. But if you wish to raise an effective case, you will need to properly adopt their methodology. That is how science (and the arts) progress.
 
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"

"Can I handle the truth indeed... yes. Very well....."

Your above reaction suggests not...

" The basic physics involved for loudspeaker enclosures....."

Scott, Please read my post and the referenced papers, my post and the papers are primarily about the loudspeaker drivers, not how they are loaded or what form of cabinet they are installed in.

Out of the 30 pages of papers, only two pages are on driver loading ( debunking the transmission line and porting theory) I understand why that is painful for you to read as you are in the transmission line speaker business.

I would still suggest it is worth your while reading both papers and checking out the references. If you google John Watkinson you will see his work is controversial but accurate.

All the best
Derek.
 
Psychoacoustic perception of time transients is above well above 100Hz, more like 300Hz and up to 6kHz or so. That range is where we really need phase accurate sound from speakers. Horn based subs below 100Hz that are properly time aligned with mid/high freq tops will not be perceived as gross distortions but as if the sound was integrated from the same source.
 
Hi,

Welcome to the Twilight Zone of rational argument.

John Watkinson is intentionally "constroversial",
but sadly full of very oversimplistic twaddle, that
he portensiously describes as "Slaying Dragons".

Here is classic guruesque quote :

"It is now well established that in order to circumvent
Heisenberg’s uncertainty theorem, the ear changes
mode during the perception of a sound."
*

No it's not. Heisenberg's uncertainty principle, not
theorem, has absolutely nothing to do with hearing,
nevermind the principle is it cannot be circumvented.

The rational don't stand a chance arguing with
such a bared faced liar spouting utter nonsense.

Anyone who is an acolyte of J.W. needs an education.

rgds, sreten.

* From : http://www.resolutionmag.com/pdfs/DRAGONS/GETBAS~1.PDF

The magazine won't be sacking him anytime soon, he's doing a great job.
 
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To paraphrase an academic book review I did for a journal a couple of months back, 'If JW's pdfs are your source of information, it is safe to say any conclusions or statements derived from such "will be suspect."'

No dragons slain. Nor anything 'debunked'. Nor any 'pain' here, I am sure you will be pleased to hear. Those pdfs you attach such great import to Derek have been doing the rounds for some little time, and are as noted above propounding bald, unreferenced and over-simplistic statements without any form of qualification whatsoever. They are not peer reviewed texts from reasonable academic sources on physics, electrical or mechanical engineering -and they wouldn't get past the first vetting stage either.
 
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THX reference

Psychoacoustic perception of time transients is above well above 100Hz, more like 300Hz and up to 6kHz or so. That range is where we really need phase accurate sound from speakers. Horn based subs below 100Hz that are properly time aligned with mid/high freq tops will not be perceived as gross distortions but as if the sound was integrated from the same source.

Hi xrk,

For sure, front loaded horns can be physically time aligned so that the acoustic centres are all in the same vertical plane. Rear loaded horns / transmission lines / ports and all delayed resonance systems can never be time accurate regardless of positioning.

I understand and agree with you on bass & frequencies up to approx 150Hz that our ear / brain takes longer and is less accurate when trying to locate the point or origin.
The reason the THX standard for high performance audio is 80Hz crossover is that their research with the original 1996 "Thomas Homlinson eXperiment" (hence THX) found that over 90% of listeners could locate the sub-woofer crossed over at 160Hz, but only 10% of listeners could do so at 80Hz.

But please note my main point in my previous posts and the JW papers has nothing to do with this subject. Really its all about the source, the driver.
Thats the part we need to ensure is time domain accurate.

The cabinet, the loading, the crossovers, the positioning....They are all of much lower importance.

Hope this helps and all the best
Derek.
 
Who said what, when & where....?

To paraphrase an academic book review I did for a journal a couple of months back, 'If JW's pdfs are your source of information, it is safe to say any conclusions or statements derived from such "will be suspect."' QUOTE.


Interesting, you reviewed a book?
Please do post the link, I am sure its a great read.

Calling the work of such a high profile author as John "suspect" is rather dangerous and not to be taken lightly.
Where is this review / book /statement? Please post the link regardless of its relevance to our audio debate.

It is not clear to me from your post who said Johns work is "suspect"
Was this you? If not who?
Thanks in advance of your reply.

PS
When you say:
" No dragons slain. Nor anything 'debunked'.
I take it you disagree with what John is stating?

All the best
Derek.
 
The book in question had nothing whatsoever to do with audio, nor JW. That's why I said 'to paraphrase'. The book review will be out in either the 1st or 2nd quarterly edition of the journal in question next year. I'll happily send you a copy when it's out, although I doubt you would find it of much interest for obvious reasons.

Be that as it may. I call it as I find it. I couldn't care less how high profile a writer is, and I've given works by extremely well known authors in my own specialist academic field poor reviews if the book deserves it. The two pdf files you posted above are not peer reviewed academic papers on physics or EM engineering published in accredited journals, they are completely unreferenced, and frequently over-simplify to the point that generalised claims derived from such become misleading. So yes: any conclusion that is derived from such source material is 'suspect'.
 
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A vote for good transients.

All natural sounds are produced using one method….One clean near instantaneous transient with rapid decay back to ambient, this is the only way we hear sound….This is also the only way to reproduce natural sounds.


Thank you Overkill. You have made a very important point. The articles you gave links to are the best explanantions of the workings of the ear I have ever come across.

Our ears and brain are evolved to hear the snapping of a twig when a tiger is creeping up behind your back.

I have had an opportunity to visit Joseph Manger in his living room. His speakers did something for me that I have never heard from other speakers not only on his stereo speakers did the orchestra show breadth, but there was also a clear hint of depth, where there was the effect of players sitting in front and behind each other. Not only that, his system had a very large "sweet spot".

If these speaker hypercube enclosures could be shown to improve the phase response of the speaker system and a spatial effect better than conventional rectangular boxes, we could be making some real progress.

I wish I had some time to explore this notions in greater detail.
 
sretin blocked so I cant see his posts.

Hi Slothrop,

I have blocked sretin so I dont have to read his posts.
So I am shooting in the dark, but John talks a bit about the "Heisenberg inequality" in a few paragraphs of one of the papers I posted.
Maybe it will make more sense if read in this context?

Quote :
" The ear works in two distinct ways, which it moves between in order to obtain the best outcome
from the fundamental limits due to the Heisenberg inequality. The Heisenberg inequality states
that as frequency resolution goes up, time resolution goes down and vice versa. Real sounds are
not continuous, but contain starting transients. During such transients, the ear works in the time
domain. Before the listener is conscious of a sound, the time domain analysis has compared the
time of arrival of the transient at the two ears and established the direction. Following the
production of a transient pressure step by a real sound source, the sound pressure must equalise
back to ambient.
The rate at which this happens is a function of the physical size of the source. The ear, again
acting in the time domain, can measure the relaxation time and assess the size of the source.
Thus before any sound is perceived, the mental model has been told of the location and size of a
sound source.
In fact this was the first use of hearing, as a means of perceiving a threat in order to survive.

Frequency analysis in hearing, consistent with the evolution of speech and music came much
later. After the analysis of the initial transient, the ear switches over to working in the frequency
domain in order to analyses timbre. In this mode, the mode that will be used on steady state
signals, phase is not very important. However, the recognition of the initial transient and the
relaxation time are critical for realism. Anything in a sound reproduction system which corrupts the
initial transient is detrimental. "

I hope this helps clarify Johns position. He is fully aware of the Heisenberg inequality. The H.I. is a peer reviewed and accepted principal. Counterintuitive, maybe misunderstood by many, but it is fact!

Hope this helps.
All the best
Derek.
 
Hi,

"
All natural sounds are produced using one method….One clean near instantaneous
transient with rapid decay back to ambient, this is the only way we hear sound….
This is also the only way to reproduce natural sounds.
"

Complete and utter nonsense, clueless, uneducated, drivel.

The reality is far more complicated and far more interesting.

As it always is comparing dogma to the brilliant light of reality.

rgds, sreten.
 
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Hi Slothrop,

I have blocked sretin so I dont have to read his posts.
So I am shooting in the dark, but John talks a bit about the "Heisenberg inequality" in a few paragraphs of one of the papers I posted.
Maybe it will make more sense if read in this context?

Quote :
" The ear works in two distinct ways, which it moves between in order to obtain the best outcome
from the fundamental limits due to the Heisenberg inequality. The Heisenberg inequality states
that as frequency resolution goes up, time resolution goes down and vice versa. Real sounds are
not continuous, but contain starting transients. During such transients, the ear works in the time
domain. Before the listener is conscious of a sound, the time domain analysis has compared the
time of arrival of the transient at the two ears and established the direction. Following the
production of a transient pressure step by a real sound source, the sound pressure must equalise
back to ambient.
The rate at which this happens is a function of the physical size of the source. The ear, again
acting in the time domain, can measure the relaxation time and assess the size of the source.
Thus before any sound is perceived, the mental model has been told of the location and size of a
sound source.
In fact this was the first use of hearing, as a means of perceiving a threat in order to survive.

Frequency analysis in hearing, consistent with the evolution of speech and music came much
later. After the analysis of the initial transient, the ear switches over to working in the frequency
domain in order to analyses timbre. In this mode, the mode that will be used on steady state
signals, phase is not very important. However, the recognition of the initial transient and the
relaxation time are critical for realism. Anything in a sound reproduction system which corrupts the
initial transient is detrimental. "

I hope this helps clarify Johns position. He is fully aware of the Heisenberg inequality. The Counterintuitive, maybe misunderstood by many, but it is fact!

Hope this helps.
All the best
Derek.

Hi,

The above is utter total BS. J.W. is taking the proverbial, its his job.

"H.I. is a peer reviewed and accepted principal."

No it isn't, its absolute nonsense, your simply being a liar.

Unless "peer" review is the clueless agreeing with each other.

rgds, sreten.

Again, how the ear actually works is far more interesting and
ellucidating, than the oversimplistic ignorant dogma above.

Autocorrelation is a good place to start, with HTF's.
 
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Agree to disagree

The book in question had nothing whatsoever to do with audio, nor JW....

Ok no need to send me a copy, I dont get round to reading important stuff like climate change, loss of biodiversity and the collapse of the oceanic ecosystem so your review may not make it to my must read list read!

I can understand why you have to deny the truth (as I read and and cross reference Johns work) and slate JW as unfounded claims ( as you read it) .....If he is correct you are out of a job!
Well not quite, but your sales of TL speakers might dip a tad!

You state your case strongly but without resorting to insults, this is as it should be.

Life is short, time is precious, or to be more poetic:

" At the end of the day all we can do is decide how to spend the time we are given"

So, on that note....I am happy to disagree with you on almost every count and wish you all the best.😉

Cheers
Derek.
 
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