Bessel vs Critically Damped Enclosure
QTC under 0.707 already give very big enclosures... If you have the room to house such large size unit, well that's fine !
Unfortunately, I dont... Then I solved the combined problem of place, tightness and infra-bass response by the Ripole solution subwoofer, in a 2x12" configuration, with a 400x400x440mm format and integrated Sub module :
But as usual, if you want to reproduce earthquake sounds at huge volume, you will need to go bigger...
T
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Wow, I missed my my presentation of basic time frequency analysis.... Pls post it Id love to read it!
I posted John Watkinsons analysis which I believe is correct, you can challenge it all day long but I doubt he reads this forum so good luck on getting a reply.
I did post that I believe the time domain is the be all and end all in transducer and loudspeaker design (we have been able to perfect tune frequency response with DSP /Eq 20 years) and I stand by that... You clearly disagree... So what is left?
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Oh really, one plus one = two.... Yea baby!
Yes indeed my "overly emphatic" (interesting ... How can you ascertain to what extent I wish to emphasise my points... Surely your talents do not extend to mind reading as well as supreme audio guru?
) conjecture is based on JW's papers (and many others which are available on Google and the AES site) on time domain in acoustics .You appear to believe that the time and frequency domains are not equivalent. That is your implicit presentation of time and frequency analysis. It is confused, as my first response highlights.
It is not correct. The paper has not been refereed and post #80 highlights some of its errant conjecture. Importantly, your quoting of the paper also neglects the specific provisos that Watkinson has included.
My first response outlined the areas of research where the answers are likely to be found. Certainly your application of inter-aural thresholds is of no relevance to this thread at all.
Is there another reason for using bold text other than emphasis?
Again, I refer you to my first response where my limits in understanding are clearly stated. I do not contribute here because I am a "supreme audio guru" and nor do I care particularly what others think of me. Instead I contribute here to share knowledge and learn from others who know better.
I am not aware of any published AES papers that support either your hypothesis or that of John Watkinson (or rather your interpretation of his paper, which as I remarked in a previous post, is not the same as my own). There are, however, many AES papers that do not support your thesis, not to mention the methods of long established physics.
As a starting point on linear system theory and the equivalence of the time and frequency domain, I thoroughly recommend reading "Time-Frequency Distributions of Loudspeakers: The Application of the Wigner Distribution" by Janse and Kaizer, JAES Vol.31 No. 4, April 1983. It starts with a summary of the theory, which some readers might find somewhat off-putting, but ends with some good practical examples.
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Well you do prove my point that "It is hard to fill a glass that is already full" ...
Again you missed my point and replaced it with an incorrect assumption, so for the benefit of others:
Your words "You appear to believe that the time and frequency domains are not equivalent. That is your implicit presentation of time and frequency analysis. It is confused, as my first response highlights."
Wrong... I stated and will re state here: All that counts is getting the time domain correct, (1) By designing a time domain accurate transducer, (2) Installing it in a time domain accurate cabinet/loading... That is 95% of the job done. Frequency response is easy as PyeSince around 2015 high quality DSP / Eq is so cheap that almost any price point of driver/loudspeaker can be tuned to whatever curve one desires. ie once you have built your time domain accurate loudspeaker you can easily Eq the frequency response.
Again as stated, all electro/mechanical transducers are at least an order of magnitude too slow to approach the required performance standard.... We need a transducer which operates at electrical speeds, NOT mechanical speeds.
The bad news is (1) Currently they dont exist. (2) Currently the vast majority of transducer/loudspeaker manufacturers are not even thinking about investing in the R&D to develop them.
The good news is this will change within a few years...
Again you missed my point and replaced it with an incorrect assumption, so for the benefit of others:
Your words "You appear to believe that the time and frequency domains are not equivalent. That is your implicit presentation of time and frequency analysis. It is confused, as my first response highlights."
Wrong... I stated and will re state here: All that counts is getting the time domain correct, (1) By designing a time domain accurate transducer, (2) Installing it in a time domain accurate cabinet/loading... That is 95% of the job done. Frequency response is easy as Pye
Since around 2015 high quality DSP / Eq is so cheap that almost any price point of driver/loudspeaker can be tuned to whatever curve one desires. ie once you have built your time domain accurate loudspeaker you can easily Eq the frequency response.Again as stated, all electro/mechanical transducers are at least an order of magnitude too slow to approach the required performance standard.... We need a transducer which operates at electrical speeds, NOT mechanical speeds.
The bad news is (1) Currently they dont exist. (2) Currently the vast majority of transducer/loudspeaker manufacturers are not even thinking about investing in the R&D to develop them.
The good news is this will change within a few years...
I repeat these two domains are entirely equivalent. In a linear sense, the notion that the time domain is all important makes no more sense than saying the frequency domain is all important. You further conflated the time domain with inter-aural time differences, which is just wrong.
You are presumably unaware that sufficiently high quality DSP augmented loudspeakers have been produced since the 1990s, with prototypes having been produced in the 1980s. The overwhelming conclusion from professionals in the field then as now is that DSP is not some "cure for all ills": If you have an under-performing loudspeaker in an analogue set-up, it will typically still be an under-performing loudspeaker with DSP added.
No. As far as the linear response goes, it is possible to generate perfect band-limited impulses that cover the audible range, albeit compromised by having more than one loudspeaker (unless you listen in true mono), and more than one driver per loudspeaker for sufficient SPLs. Whilst DSP can ameliorate some non-linearities, it cannot compensate for comprised SPLs due to limited displacement volumes.
Again, no. There is no evidence to support this statement at all for the reason I stated above. Unless you are hinting at digital loudspeakers?
The bad news is actually that digital loudspeakers have been tried too. 1 Limited, for example, produced a spiral like device filled with aerogel in the 1990s, but it lacked sufficient displacement volume to be successful, even in large arrays. I am also aware of an experimental plasma loudspeaker from that era that was unsuccessful because air turns out not to be sufficiently linear at such high frequencies (not to mention the ozone it generated). Others in that era exploited small conventional driver arrays or diffuse mode drivers, even utilising whole walls to increase the radiating area, but with limited success too. Very few such technologies have survived in the high fidelity arena.
Unless you can envisage a change in the laws of physics, wherein you can produce large displacement volumes from small drivers with low distortion, your optimism is folly. I would love to be proved wrong, but musical instruments are the size they are for sound reasons.
I think I need an emoji of a glass of water overflowing.... You clearly have neither the capacity or attitude capable of absorbing new ideas.
I have explained clearly my points and am content most people will understand and maybe investigate further.
PS I have been using DSP /Eq for over 25 years, as a "sticking plaster" to treat the "ruptured artery" of so called state of the art" transducers and I was the first exhibitor in CES 2004 to showcase a DEQX DSP with linear phase crossovers, driver correction and room Eq... It sounded phenomenal but like every other "cost no object" Hi Fi then and since.... IT SUCKED! 100% of people could tell that it was a HiFi and not a real piano, never mind a real test ie Big Band or Orchestra.
Why? because ALL transducers are fundamentally flawed.... See my previews posts and JW Etc articles.... Ground Hog Day!
So my document holder is happy to remain stationary... I rest my case
I have explained clearly my points and am content most people will understand and maybe investigate further.
PS I have been using DSP /Eq for over 25 years, as a "sticking plaster" to treat the "ruptured artery" of so called state of the art" transducers and I was the first exhibitor in CES 2004 to showcase a DEQX DSP with linear phase crossovers, driver correction and room Eq... It sounded phenomenal but like every other "cost no object" Hi Fi then and since.... IT SUCKED! 100% of people could tell that it was a HiFi and not a real piano, never mind a real test ie Big Band or Orchestra.
Why? because ALL transducers are fundamentally flawed.... See my previews posts and JW Etc articles.... Ground Hog Day!
So my document holder is happy to remain stationary... I rest my case
I don't recall you defining a "required performance standard", but if your standard is 100% of people believing a HiFi is a real piano, big band or orchestra, it is simply unachievable other than in an artificially controlled environment.
That said, Powersoft and B&C speakers already invested the R&D to develop the IPAL (Integrated Powered Adaptive Loudspeaker) system prior to 2011, and presented an AES paper explaining it.
https://www.audioclub.ro/cs-content/cs-docs/9844-1598281048.pdf
To keep it short, the IPAL creates a virtual transducer capable of self correcting the "fundamentally flawed" mechanical timing problems inherent with real transducers capable of large displacement volumes.
Although the IPAL system can largely eliminate the time domain problems associated both of transducers and cabinet/loading, it has not achieved widespread use, as users are not willing to pay the cost for those improvements, even though they demonstrably are far better .
Art
Thats very interesting, thanks for posting Art, I will have a good read and follow up the links.
Back in the 1980's Peter Walker of Quad described the "Perfect" transducer as " A mass-less pulsating sphere able to cover 20 Hz to 20 KHz" or something close to that?
Something along these lines is coming down the track...
Thanks again
A.
Back in the 1980's Peter Walker of Quad described the "Perfect" transducer as " A mass-less pulsating sphere able to cover 20 Hz to 20 KHz" or something close to that?
Something along these lines is coming down the track...
Thanks again
A.
Yes indeed Mr Klinky that is a very succinct question.
It is indeed a time domain phenomenon and critical to how humans detect, decode and locate sound (the JW article explains a bit about how the Human Auditory System - HAS works).
If simplified in extreme, everything JW suggests is that if the: (1) transducer and/or (2) loudspeaker loading, introduces time domain errors above this 10 microsecond threshold then the HAS detects this and "gets confused" ie "the computer says no!"
Part of this is evolution ie the HAS evolved as a survival mechanism as detecting where sounds were located in 3D space was the top priority (life or death). Identifying what a sound was, is a low priority ie When we were primates up a tree and we heard a noise, we instantly bolted 10 meters away from that sound...Was that sound a snapping twig caused by a predator stalking me, or a falling coconut? It did not matter! The point was after I had bolted to a safe distance and a safe location I was alive and had the luxury of time to identify what the sound was, real or false alarm I am still alive!
Important point, please read this! Why, biologically, time domain is the be all and end all of accurate sound detection/decoding
Since mammals/primates humans evolved our HAS, every single sound ever detected has been REAL with perfect time domain / step response ie NOT a sine wave (they do not exist in nature) or a sound produced by a man made transducer / loudspeaker with gross time domain errors measured in the millisecond scale.
Our HAS is built around and relies upon the time domain and is astonishingly accurate (microsecond scale) and is ultra sensitive to errors in time... This is why our HAS instantly detects and rejects any and all sounds as "fake" when they are reproduced with time domain errors in the millisecond scale...
I hope I am explaining this well enough to encourage others to investigate this further.
Thanks for reading this far if you have!
Cheers
A.
It is indeed a time domain phenomenon and critical to how humans detect, decode and locate sound (the JW article explains a bit about how the Human Auditory System - HAS works).
If simplified in extreme, everything JW suggests is that if the: (1) transducer and/or (2) loudspeaker loading, introduces time domain errors above this 10 microsecond threshold then the HAS detects this and "gets confused" ie "the computer says no!"
Part of this is evolution ie the HAS evolved as a survival mechanism as detecting where sounds were located in 3D space was the top priority (life or death). Identifying what a sound was, is a low priority ie When we were primates up a tree and we heard a noise, we instantly bolted 10 meters away from that sound...Was that sound a snapping twig caused by a predator stalking me, or a falling coconut? It did not matter! The point was after I had bolted to a safe distance and a safe location I was alive and had the luxury of time to identify what the sound was, real or false alarm I am still alive!
Important point, please read this! Why, biologically, time domain is the be all and end all of accurate sound detection/decoding
Since mammals/primates humans evolved our HAS, every single sound ever detected has been REAL with perfect time domain / step response ie NOT a sine wave (they do not exist in nature) or a sound produced by a man made transducer / loudspeaker with gross time domain errors measured in the millisecond scale.
Our HAS is built around and relies upon the time domain and is astonishingly accurate (microsecond scale) and is ultra sensitive to errors in time... This is why our HAS instantly detects and rejects any and all sounds as "fake" when they are reproduced with time domain errors in the millisecond scale...
I hope I am explaining this well enough to encourage others to investigate this further.
Thanks for reading this far if you have!
Cheers
A.
It's a servo system with memory - motional feedback wrapped up in glossy language, and technology that has also been explored before. I have posted on this forum previously about David Birt's self-balancing bridge that is capable of extending the HF performance of conventional acoustic or optic motional feedback systems considerably: Sadly it has never achieved commercial success either.
A standard is implicit in anything concerning "high fidelity loudspeakers" - otherwise they are not worthy of the term high fidelity. The reproduction of just a real piano does indeed present a formidable target to even the best loudspeaker systems available, and few achieve such high performance. Nevertheless, distortion, peak and rms outputs remain valuable measures in assessing the fidelity of a loudspeaker, and displacement volume will likely always represent a formidable obstacle in extending performance. Are you suggesting a "moderate fidelity" label for people to consider instead?
ESL's are fundamentally limited by their displacement volume. Many Quad ESL listeners will still advocate (with good reason) that the ESL-57 audibly outperforms the later ESL-63, the ESL-63 being the first to attempt spherical wavefront construction. Later Quad versions may have improved and be more reliable, but their fundamental limitation remains the same.
We might consider an array of AMT elements mounted on some spherical surface as a technological successor to Peter Walker's invention*, but even then it would likely be displacement or low frequency limited and it's scale would have to be enormous to generate sufficient SPLs to be considered a high fidelity loudspeaker. Notably, Peter Walker's own corner mounted ESL never succeeded either.
As Peter Walker states "a mass-less pulsating sphere able to cover 20 Hz to 20 KHz" is band-limited. That is, it would produce an impulse response dispersed in time exactly commensurate with its frequency response. That is just basic maths/physics. Furthermore, band-limited "mass-less" operation has been effectively achieved already using the aforementioned motional feedback to control driving impedance.
* [I append this to avoid anybody insinuating a criticism of Peter Walker from my contribution here. I knew him and respected him immensely. I had many treasured conversations with Peter and Peter Baxandall, both of whom, faced with "youngsters" impressing with superlative claims for DSP and new-fangled miracle inventions, would simply take out a notebook and a pencil, and highlight the obvious errors in the claims. They were both accomplished engineers thoroughly understanding the fundamentals of audio engineering.]
Time will surely tell. But the fundamentals of physics and likely loudspeaker performance limitations will remain steadfast too.
You continue to confuse two different subjects, and Watkinson's article does not explain the human auditory system either. Linear system analysis is not sufficent to explain our inherent auditory capabilities. As I hinted earlier, for transient sound identification, look at the bispectrum, and for interaural time differences look at the cross-bispectrum. They are not the same thing.
Why the need to get personal? it does not help your case whatsoever. Your points remain invalid and likely will remain so forever. John Watkinson's article was not refereed and is flawed - as was the loudspeaker system based upon his thinking. Please stick to relevant factual matters when you reply.Hey soundbloke, I bet you get invited to a lot of parties...
Regardless of your overwhelming enthusiasm I believe my points and JW articles are valid.
My last post is not trying to help my case, you are a lost cause so why should I waste any more time on you?
For the benefit of others...
I am pointing out that your totally negative and closed minded attitude is the exact opposite of what the DIY community needs.
In addition you refuse to even read the JW paper but still dismiss it as flawed.
This is evidenced by the fact that there are over 4 pages explaining and 12 references to, the HAS in the article which you stated does not explain the HAS.
JW has had multiple articles on acoustics and related subjects published by a wide variety of sources and is a well respected author, highly qualified and internationally recognised academic... Vs you " soundblock" I wonder who will help further the art of sound reproduction?
For the benefit of others...
I am pointing out that your totally negative and closed minded attitude is the exact opposite of what the DIY community needs.
In addition you refuse to even read the JW paper but still dismiss it as flawed.
This is evidenced by the fact that there are over 4 pages explaining and 12 references to, the HAS in the article which you stated does not explain the HAS.
JW has had multiple articles on acoustics and related subjects published by a wide variety of sources and is a well respected author, highly qualified and internationally recognised academic... Vs you " soundblock"
I wonder who will help further the art of sound reproduction?
It looks like the Powersoft module is the very expensive part, the drivers are mid to expensive but not too far above mainstream driver pricing. A lot of driver manufacturers have now developed compatible drivers.
This is interesting " Features of the Powersoft IpalMod include a Class D amplifier (Max 8500W @ 1ohm / 180 Vpeak) which is designed to work with a load with very low impedance and high inductance, and a Differential Pressure Sensor (DPC) located on the speaker baffle which sends a Zero Latency (10 µs) closed loop reading to the special on-board DSP to change the power output and behavior of speaker/s. At the same time, the IpalMod recycles the Back Electro-Motive Force (BEMF) from the speaker/s into the capacitor bank of the power supply to significantly increase the output efficiency.
https://audioxpress.com/news/lavoce-launches-powersoft-ipalmod-compatible-18-and-21-subwoofers
No, I am alerting those not versed in the fundamentals to errors in your statements. I do not have a closed mind, and I have mounted a well-supported challenge to the claims you have asserted without proper foundation. My mind is always open to genuine advances in sound reproduction, but your contribution simply does not provide any.
No, I have read the paper, and responded with quotes from it. I also recommended that you read the paper with appropriate diligence and recognise the provisos John Watkinson included. Instead you continue to cited it out of context. Furthermore, the paper is largely conjecture that does not fit well with cognitive studies, and even contradicts itself at times, of which I highlighted one example.
I am actually not aware of any academic articles published in journals by John Watkinson, but I am happy to be put right on that matter (as on any other). I am aware of his magazine-type articles and book chapters that summarised the contributions of others, but that is not the same thing.
Nevertheless, quoting academic studies out of context, regardless of who wrote them, does not validate your claims.
Furthermore, as I have now stated twice, John Watkinson's venture into loudspeaker manufacturing based on the ideas expressed in his article was not successful. It did not help the science of sound reproduction, and I have provided one good reason why.